[cisco-voip] Bat.xlt for CUCM 12.5.1

[ROZA, Ariel via cisco-voip]

Hi, Guys and gals!

Does anyone has a bat.xlt file for a cucm 12.5? I am working on a cluster that 
seem to have it´s file deleted and I am pinging everyone I know before 
resorting to install whole server for it.

Regards,

Ariel


Ariel Pablo Roza
Post Sales UC Engineer / Southern Cone [cid:image001.png@01D72589.C1393CA0]

t: (011) 5282-0458
m: (011) 5017-4417
ariel.r...@la.logicalis.com
Av. Belgrano 955 - Piso 20 - C1092AAJ
CABA - Buenos Aires - Argentina
www.la.logicalis.com

[cid:image002.png@01D72589.C1393CA0]
[firmacalidad-1]



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Re: [cisco-voip] MRA DR / Resilience

[ROZA, Ariel]

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

De: cisco-voip  En nombre de ROZA, Ariel
Enviado el: lunes, 18 de enero de 2021 15:53
Para: NateCCIE ; Pawlowski, Adam 
CC: 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 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] MRA DR / Resilience

[ROZA, Ariel]

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

De: cisco-voip  En nombre de NateCCIE
Enviado el: miércoles, 13 de enero de 2021 10:56
Para: Pawlowski, Adam 
CC: 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

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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Re: [cisco-voip] Problem registering Webex Room Kit to the cloud

[ROZA, Ariel]

Hi, Joe

It´s the only device using cloud registration right now.

We just found out about that with a TAC case. I sent this message while I was 
switching between engineers.
It is indeed a connection error (the proxy seems at fault), but the error 
message was really misleading and we spent hours looking at other things like 
the NTP config and Timezone.

Thanks,

Ariel.
De: Joe Martini (joemar2) 
Enviado el: viernes, 23 de octubre de 2020 17:07
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] Problem registering Webex Room Kit to the cloud

Hi Ariel,

Can you check if 
activation.webex.com<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Factivation.webex.com%2F=04%7C01%7CAriel.ROZA%40la.logicalis.com%7C450351ac8ba3476f32ea08d8778f4500%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C637390804511466244%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=2MhN2styHDT0iWVAYHi1cdgqdvlluURqzAPNxH29kgo%3D=0>
 is reachable via HTTPS from the device/network?  Do you have any other devices 
that are successfully cloud registered from the same network as the Webex Room 
Kit displaying this message?

Joe

On Oct 23, 2020, at 2:17 PM, ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:

Hi Guys and Gals,

Have you ever seen this error: activation failed. system time adjustment failed 
while registering a Room Kit to Webex?

I have already setup NTP with and internal NTP Server, and Diagnostics on the 
unit show NTP as ok. But I run the wizard to register it to Webex, I input the 
activation code, and after “thinking” for a while, I get that error.

The very odd thing is that the NTP section in the configuration menu is missing 
after my first attempt to register the device (That is already set to use 
Webex, and not CUCM)

Ariel Pablo Roza
Post Sales UC Engineer / Southern Cone 

t: (011) 5282-0458
m: (011) 5017-4417
ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>
Av. Belgrano 955 – Piso 20 - C1092AAJ
CABA – Buenos Aires - Argentina
www.la.logicalis.com<https://nam10.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.la.logicalis.com%2F=04%7C01%7CAriel.ROZA%40la.logicalis.com%7C450351ac8ba3476f32ea08d8778f4500%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C637390804511476239%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=ySvz1c%2BvrMAg2yHwpQOjsnS0ZX040JvvlEHBdGF4ck8%3D=0>




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[cisco-voip] Problem registering Webex Room Kit to the cloud

[ROZA, Ariel]

Hi Guys and Gals,

Have you ever seen this error: activation failed. system time adjustment failed 
while registering a Room Kit to Webex?

I have already setup NTP with and internal NTP Server, and Diagnostics on the 
unit show NTP as ok. But I run the wizard to register it to Webex, I input the 
activation code, and after "thinking" for a while, I get that error.

The very odd thing is that the NTP section in the configuration menu is missing 
after my first attempt to register the device (That is already set to use 
Webex, and not CUCM)

Ariel Pablo Roza
Post Sales UC Engineer / Southern Cone [cid:image005.png@01D6A94F.9838FB30]

t: (011) 5282-0458
m: (011) 5017-4417
ariel.r...@la.logicalis.com
Av. Belgrano 955 - Piso 20 - C1092AAJ
CABA - Buenos Aires - Argentina
www.la.logicalis.com

[cid:image003.png@01D6A945.47D928E0]
[firmacalidad-1]



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Description: oledata.mso
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Re: [cisco-voip] Cellphone with pararell calls

[ROZA, Ariel]

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 
Enviado el: martes, 16 de junio de 2020 21:41
Para: ROZA, Ariel ; cisco-voip 
(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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[cisco-voip] Cellphone with pararell calls

[ROZA, Ariel]

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] SX80 Camera Tracking and Face Masks

[ROZA, Ariel]

:D

De: Brian Meade 
Enviado el: martes, 9 de junio de 2020 20:04
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SX80 Camera Tracking and Face Masks

Try drawing a mouth on the mask and see if that helps.

On Tue, Jun 9, 2020 at 1:31 PM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi Guys,

Did anyone experience problems with speaker tracking while people using 
facemasks?
I received reports from people that verified lack of tracking while using a 
mask, and that it works properly when they take them off.
This is not weird, and somehow expected. But wanted to know if there´s any kind 
of solution/fix/workaround.
Tried looking for bugs in the BST, but does not show anything for SX80

Regards,


[cid:image001.png@01D63F24.2ADC9500]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
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Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
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Business and technology working as one
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[cisco-voip] SX80 Camera Tracking and Face Masks

[ROZA, Ariel]

Hi Guys,

Did anyone experience problems with speaker tracking while people using 
facemasks?
I received reports from people that verified lack of tracking while using a 
mask, and that it works properly when they take them off.
This is not weird, and somehow expected. But wanted to know if there´s any kind 
of solution/fix/workaround.
Tried looking for bugs in the BST, but does not show anything for SX80

Regards,


[cid:image001.png@01D63E69.CC34D130]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
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Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image011.jpg@01D63E6A.4DE02900][cid:image004.png@01D63E69.CC34D130][cid:image005.png@01D63E69.CC34D130][cid:image006.png@01D63E69.CC34D130][cid:image007.png@01D63E69.CC34D130][cid:image008.png@01D63E69.CC34D130][cid:image009.png@01D63E69.CC34D130]
[cid:image010.jpg@01D63E69.CC34D130]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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Re: [cisco-voip] [PLM] muliple PAKs and One license request?

[ROZA, Ariel]

*Smacks his head*

 Thanks, Anthony

De: Anthony Holloway 
Enviado el: jueves, 6 de febrero de 2020 12:45
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] [PLM] muliple PAKs and One license request?

You can register more than one PAK at a time, up to 10.  Would that work for 
you?

[image.png]



On Thu, Feb 6, 2020 at 7:25 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, guys.

Is it posible to register multiple PAKS using the same License Request from 
PLM? Or do I have to get a new License Request after I load each license.
I can´t find a clear answer so far.
I have to register 5+ CUCM PAKs for a customer, and I have to be sure 
beforhand, so I don´t screw up the licensing.

I have a couple of license requests saved a few months appart from each other, 
and I ran a diff on them and saw that there were small differences. So, I am 
not sure if the license requests are static or change overtime, or how often.

Anyone knows any of this?

Regards,


[cid:image002.png@01D5DCF6.81257980]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
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Business and technology working as one
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[cid:image010.jpg@01D5DCF6.81257980]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
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[cisco-voip] [PLM] muliple PAKs and One license request?

[ROZA, Ariel]

Hi, guys.

Is it posible to register multiple PAKS using the same License Request from 
PLM? Or do I have to get a new License Request after I load each license.
I can´t find a clear answer so far.
I have to register 5+ CUCM PAKs for a customer, and I have to be sure 
beforhand, so I don´t screw up the licensing.

I have a couple of license requests saved a few months appart from each other, 
and I ran a diff on them and saw that there were small differences. So, I am 
not sure if the license requests are static or change overtime, or how often.

Anyone knows any of this?

Regards,


[cid:image001.png@01D5DCD7.579A0A50]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image011.jpg@01D5DCD7.B706E760][cid:image004.png@01D5DCD7.579A0A50][cid:image005.png@01D5DCD7.579A0A50][cid:image006.png@01D5DCD7.579A0A50][cid:image007.png@01D5DCD7.579A0A50][cid:image008.png@01D5DCD7.579A0A50][cid:image009.png@01D5DCD7.579A0A50]
[cid:image010.jpg@01D5DCD7.579A0A50]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma

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

[ROZA, Ariel]

But without clustering, if Core1 fails, Edge1 will still be active and Jabber 
clients will still see Edge1 running and attempt to connect through it!

De: cisco-voip  En nombre de Charles 
Goldsmith
Enviado el: martes, 28 de enero de 2020 23:18
Para: Lelio Fulgenzi 
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Expressway Cluster failover for MRA...

We've built them as individual pairs (Edge/Core) and then use DNS to control 
which one goes where.  Without the cluster, we know that Edge1 will always talk 
to Core1.

I get the feeling that clustering was always meant to be in the same DC, and 
for redundancy purposes in the same DC.

If you have two DC's, either a cluster at each DC, or just a pair at each DC, 
depending on the business needs.

On Tue, Jan 28, 2020 at 8:11 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

How does no. 2 actually solve the problem of having to log back in?

Is this a supported/suggested deployment method?

It’s been a while since I first looked at things and don’t recall things 
mentioning using the cluster name in the SRV records.

I’m intrigued. And interested!


-sent from mobile device-

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

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


On Jan 28, 2020, at 9:03 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
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 
mailto:jonv...@gmail.com>> 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] Webex Meeting Server and VMWare patching

[ROZA, Ariel]

Hi, everyone.

I have been asked to patch some vulnerabilities in a couple of IRP Servers from 
a Webex Meeting Server solution that are exposed to the Internet. There are 8 
more serves in the Webex deployment besides the IRPs. It is my first time 
working with VMWare and patches. Although I have the process figured out, I 
still have one question.
Is it necessary to patch all the 10 servers, or is it enough with the servers 
that have been singled out.
I have consuted others, and know that -in theory- every ESXi host is 
independent from the rest so I am working with the idea of doing only those two 
servers.
(I know that the IDEAL solution is to patch everyting, but that system will 
soon be decomissioned and it doesen´t seem to be worth all the work)

What do you think?

Thanks,

Ariel

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Re: [cisco-voip] tools to parse CDR quickly on Windows (windows grep?)

[ROZA, Ariel]

About Windows grep:

https://sourceforge.net/projects/grepwin/


De: cisco-voip  En nombre de Lelio Fulgenzi
Enviado el: jueves, 17 de octubre de 2019 10:55
Para: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Asunto: [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] Expressway cluster certificates.

[ROZA, Ariel]

Hi Ryan,

Both Expressway servers are signed by the internal CA. I have uploaded the root 
and intermediate certificates, too.
But I am renewing the certificates on an existing cluster, and whoever 
instelled it, they manually added the ExpC certs into tomcat-trust.

So, I understand that it would be safe to remove the ExpC certs from 
tomcat-trust and everything would be working fine?
What about the use the cluster name/don´t use the cluster name contradiction?

Thanks,

Ariel.

De: Ryan Huff 
Enviado el: lunes, 14 de octubre de 2019 18:14
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] Expressway cluster certificates.

Are the expressway-C server using self-signed certificates (I doubt it because 
you said they are multi-san)?

Generally, CUCM doesn’t need to trust the identity certificate (unless it is 
self signed). In all other cases, CUCM needs to trust the certificate authority 
the signed the expressway-c certificates.

If for example, GoDaddy signed the SSL certificates for the Expressway-C, CUCM 
just needs to trust the GoDaddy certificate authority chain.
Sent from my iPhone


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[cisco-voip] Expressway cluster certificates.

[ROZA, Ariel]

Hi, Guys

I am renewing the certificates in an Expressway X8.10.1 cluster. But I am 
running into a conflict between the official documentation and how CUCM works.

I have set both Expressway-C certificates to use the Cluster name for the 
Common Name and each server´s name as a SAN, as the oficial guide states.
But when I load both signed certificates into CUCM trust stores, it shows only 
one of the certificates, instead of both, as CUCM uses the CN tu build its 
listo f certs, and both ExpC´s CN is the same (Although they are two diferente 
certificates)

So, I started to re-read all related documents I could find and I found some 
contradictions that I do not now how to solve.

On one hand, I have the official “Certificate  Creation and Deployment Guide” 
that states:

“A certificate identifies the Expressway. It contains names by which it is 
known and to which traffic is routed. If the Expressway is known by multiple 
names for these purposes, such as if it is part of a cluster, this must be 
represented in the X.509 subject data, according to the guidance of RFC5922. 
The certificate must contain the FQDN of both the Expressway itself and of the 
cluster. The following lists show what must be included in the X.509 subject, 
depending on the deployment model chosen.
If the Expressway is not clustered:
■ Subject Common Name = FQDN of Expressway
■ Subject Alternate Names = leave blank*
If the Expressway is clustered, with individual certificates per Expressway:
■ Subject Common Name = FQDN of cluster
■ Subject Alternate Name = FQDN of Expressway peer, FQDN of cluster*

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-10/Cisco-Expressway-Certificate-Creation-and-Use-Deployment-Guide-X8-10.pdf

On the other hand I have the “Configure and Troubleshoot Collaboration Edge 
(MRA) Certificates“ that says:

Cluster Certificates

It is strongly recommended that if you have a cluster of Expressway-C or 
Expressway-E servers for redundancy that you generate a separate CSR for each 
server and have it signed by a CA.  Most deployments will use the server name 
for the subject and list all peers and the cluster ID as SANs.  It is possible 
for you to use the cluster-id as the subject to use the same certificate for 
all nodes in the cluster, therefore avoiding the cost of multiple certs signed 
by a public CA.  If absolutely necessary, this can be done with the following 
process or by using OpenSSL to generate both the private key and CSR manually:

Step 1.  Generate a CSR on the master of the cluster and configure it to list 
the cluster-alias as the subject.  Add all peers in the cluster as alternative 
names, along with all other required SANs.

Step 2.  Sign this CSR and upload to the master peer.

Step 3.  Log into the master as root and download the private key located in 
/tandberg/persistent/certs.

Step 4.  Upload both the signed certificate and matching private key to each 
other peer in the cluster.

Note: This is not recommended for the following reasons:
1. It is a security risk because all peers are using the same private key.  If 
one is somehow compromised an attacker can decrypt traffic from any of the 
servers.
2.  If a change needs to be made to the certificate, this entire process must 
be followed again rather than a simple CSR generation and signing.

https://www.cisco.com/c/en/us/support/docs/unified-communications/expressway/213872-configure-and-troubleshoot-collaboration.html#anc17


So, one says to use the cluster name, the other says the opposite. And I have 
the CUCM showing me only one cert intead of two.


What should I do? Re-sign both certificates with the peer name as CN and 
cluster as SAN and be done with it? Ori s there a legitimate way to use the 
cluster name and not have issues with CUCM?

Right now, the Expressway cluster is in service, because I left the cluster´s 
main peer certificate showing in CUCM, but as far as I know, the backup peer 
won´t work.

TIA,


[cid:image001.png@01D582B8.04178020]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image003.jpg@01D582BA.3104E530][cid:image004.png@01D582B8.04178020][cid:image005.png@01D582B8.04178020][cid:image006.png@01D582B8.04178020][cid:image007.png@01D582B8.04178020][cid:image008.png@01D582B8.04178020][cid:image009.png@01D582B8.04178020]
[cid:image010.jpg@01D582B8.04178020]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.

[cisco-voip] Question about CUCM CA-Signed certificates.

[ROZA, Ariel]

Hi, people,

Out of curiosity, does anyone know what´s the specific purpose, in a CUCM 
cluster, of the "Client Authentication" extensión that we must include in the 
template used to sign CUCM certificates? Because I am renewing some expired 
tomcat certificates in a cluster, and those certs only have the "Server 
Authentication" extensión, instead of both, Server and Client.
And my customer never complained about issues in their setup.

Regards,

Ariel.
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Re: [cisco-voip] Status icons in videoconferences

[ROZA, Ariel]

My bad. The screenshot is from Skype for Business. I´ll have to find out how 
those statuses are handled there.

De: Brian Meade 
Enviado el: jueves, 13 de junio de 2019 11:09
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] Status icons in videoconferences

Ariel,

Where was that screenshot taken?  It looks different than Cisco Meeting App and 
the WebRTC App I'm used to for CMS.  I didn't see those icons in the user guide 
either.

Thanks,
Brian Meade

On Wed, Jun 12, 2019 at 12:11 PM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
I am having this issue with a multicluster videoconference in a CMS room.. Some 
of the devices show a greyed out icon in their statuses. (highligthed in the 
picture). Those three devices are registred to the same cluster.
How are those icons governed? What turns them on or off?

Regards,

Ariel

[cid:image001.png@01D522B4.9F986500]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
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Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
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vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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[cisco-voip] Status icons in videoconferences

[ROZA, Ariel]

I am having this issue with a multicluster videoconference in a CMS room.. Some 
of the devices show a greyed out icon in their statuses. (highligthed in the 
picture). Those three devices are registred to the same cluster.
How are those icons governed? What turns them on or off?

Regards,

Ariel

[cid:image001.png@01D52120.3E7F7C30]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image003.jpg@01D52120.3E7F7C30][cid:image006.png@01D298D0.103CDF50][cid:image007.png@01D47689.672FCCD0][cid:image005.png@01D47689.672FCCD0][cid:image006.png@01D47689.672FCCD0][cid:image008.png@01D47689.672FCCD0][cid:image009.png@01D47689.672FCCD0]
[cid:image010.jpg@01D47689.672FCCD0]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma

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Re: [cisco-voip] Removing certificates by hand (ROZA, Ariel)

[ROZA, Ariel]

Well, I opened the TAC case and when I showed the Certificate Management page 
to the engineer, the damned expired certificate was right there. (I have a 
screenshot I took yesterday to prove that it was missing!).
This is not the first time it happens to us. It seems we had the same issue 
with the cert (Being visible/invisible in the os admin pages) three months ago.

Still checking with the TAC guy if there´s a bug or a database inconsistency.

De: cisco-voip  En nombre de ROZA, Ariel
Enviado el: miércoles, 10 de abril de 2019 15:38
Para: Pawlowski, Adam ; cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Removing certificates by hand (ROZA, Ariel)

Thanks, Adam
Obtener Outlook para 
Android<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Faka.ms%2Fghei36=02%7C01%7Cariel.roza%40la.logicalis.com%7Ca7b864fa228c4eefa1da08d6bde3bdf2%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905183211401273=ezRyB%2B2IgOgY1zLq5cghQQ7R6c%2Bu2BnvI4IrHJPJl7M%3D=0>


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Wednesday, April 10, 2019 2:49:14 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Removing certificates by hand (ROZA, Ariel)

There are some bugs that we ran into where expired certs keep coming back after 
being deleted, or aren't seen in the cert admin page (CSR 11.5) and will alarm 
endlessly until TAC dials in and erases them.

While this is preferable to the bugs that would cause the cert page to not even 
render , its still annoying.

TAC can get these out in minutes though.

Adam

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Re: [cisco-voip] Removing certificates by hand (ROZA, Ariel)

[ROZA, Ariel]

Thanks, Adam

Obtener Outlook para Android<https://aka.ms/ghei36>


From: cisco-voip  on behalf of Pawlowski, 
Adam 
Sent: Wednesday, April 10, 2019 2:49:14 PM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Removing certificates by hand (ROZA, Ariel)

There are some bugs that we ran into where expired certs keep coming back after 
being deleted, or aren’t seen in the cert admin page (CSR 11.5) and will alarm 
endlessly until TAC dials in and erases them.

While this is preferable to the bugs that would cause the cert page to not even 
render , its still annoying.

TAC can get these out in minutes though.

Adam

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Re: [cisco-voip] Removing certificates by hand

[ROZA, Ariel]

Yes. The node matches the one in the alert

Obtener Outlook para Android<https://aka.ms/ghei36>


From: Brian Meade 
Sent: Wednesday, April 10, 2019 12:46:24 PM
To: ROZA, Ariel
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Removing certificates by hand

Are you looking on the right node that alerted?  I wouldn't expect this to be 
in the DB either if it doesn't display in OS Admin.

On Wed, Apr 10, 2019 at 11:23 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
I am receving RTMT alarms about an expired tomcat-trust certificate, but the 
certificate is not listed in the Certificate Management page. The cert was 
replaced by a new certificate that is present in the list, nonetheless.

Can I remove the old certificate by hand through the CLI? Or there is something 
to be done at the database level?

Thanks.

[cid:16a07ed4f504cff311]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Flogicalis-la.webex.com%2Fjoin%2Fariel.roza=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032382303=pHnHCderd16daeBIjGy08RjApmMvXAMiIJqpEEo0WHk%3D=0>
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
www.la.logicalis.com<https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.la.logicalis.com%2F=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032382303=YTBpZjgdSdklZtSzlFjTzC0967Bc53l5nwVYsi9XpbU%3D=0>
Business and technology working as one
[cid:16a07ed4f51201c1889][cid:image006.png@01D298D0.103CDF50][cid:image007.png@01D47689.672FCCD0]<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.instagram.com%2Flogicalislatam%2F=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032392307=WbGsKZ%2FJNVKQlK8q2yMOHYxrgiuwuORhXcUFJjGFhm8%3D=0>[cid:image005.png@01D47689.672FCCD0]<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.facebook.com%2Flogicalislatam=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032402320=uSH2izOq9bpDMnHc%2F0NlajVktZiB4SGCsDmctkLLLFM%3D=0>[cid:image006.png@01D47689.672FCCD0]<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftwitter.com%2Flogicalislatam=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032412328=CN4Y8DujlkGTUYV4yiTsUPHPwHTu%2BFcUAnOIqG1qOy0%3D=0>[cid:image008.png@01D47689.672FCCD0]<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Far.linkedin.com%2Fcompany%2Flogicalis-latam=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032412328=D9hQo66WdXezKFs%2BDjRYrfqIIj9TQxIMwfTgf8HfwLU%3D=0>[cid:image009.png@01D47689.672FCCD0]<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.youtube.com%2Flogicalislatam=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C0ff6bbc519b04c708fb008d6bdcbb99d%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636905080032422333=cVdIlNay4dXLse5eo81ChFLGmGXxZA6hwK6mc3NHGAw%3D=0>
[cid:image010.jpg@01D47689.672FCCD0]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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[cisco-voip] Removing certificates by hand

[ROZA, Ariel]

I am receving RTMT alarms about an expired tomcat-trust certificate, but the 
certificate is not listed in the Certificate Management page. The cert was 
replaced by a new certificate that is present in the list, nonetheless.

Can I remove the old certificate by hand through the CLI? Or there is something 
to be done at the database level?

Thanks.

[cid:image001.png@01D4EF97.A8D134D0]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image011.jpg@01D4EF97.F3838550][cid:image006.png@01D298D0.103CDF50][cid:image007.png@01D47689.672FCCD0][cid:image005.png@01D47689.672FCCD0][cid:image006.png@01D47689.672FCCD0][cid:image008.png@01D47689.672FCCD0][cid:image009.png@01D47689.672FCCD0]
[cid:image010.jpg@01D47689.672FCCD0]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma

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

[ROZA, Ariel]

In the end, my customer finally realized the problema was on the PBX side. The 
unupgraded PBX worked fine.

Thanks Brian for your help.

Regards,

Ariel.

De: cisco-voip  En nombre de ROZA, Ariel
Enviado el: martes, 26 de marzo de 2019 14:04
Para: Brian Meade 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

´ll check with my customer, and report back.
I saw that negative parameter on the o= line, but I wasn´t completely certain 
how to handle it.
Thanks for the help!

De: Brian Meade mailto:bmead...@vt.edu>>
Enviado el: martes, 26 de marzo de 2019 13:56
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Actually meant o= line is the origin line.

On Tue, Mar 26, 2019 at 12:39 PM Brian Meade 
mailto:bmead...@vt.edu>> wrote:
It's definitely failing at parsing the SDP on that invite and finding an 
invalid parameter:
07517620.001 |16:00:23.657 |AppInfo  |//SIP/SIPUdp/wait_UdpDataInd: Incoming 
SIP UDP message size 932 from 172.27.0.15:[5060]:
[1031135,NET]
INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>
Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15>
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: gquerol
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp

v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

07517621.007 |16:00:23.657 |AppInfo  
|//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - 
sdp_ret=SDP_INVALID_PARAMETER

You may need to use a SIP Normalization script to clean up what they are 
sending.

I think it's the o= line (organization line).  That's 2nd value (-835641967) 
should be a positive number I believe.  That session-id parameter is supposed 
to match NTP format- 
https://tools.ietf.org/html/rfc4566#section-5.2<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Frfc4566%23section-5.2=02%7C01%7Cariel.roza%40la.logicalis.com%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878451827=YILRc5UxLiA30sXHmbaGaM9AAyJcHE7P4mc2VEjM8To%3D=0>

Maybe just check their server has NTP synced okay to start?

Thanks,
Brian Meade



On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Here´s the trace file with the bad call



De: Brian Meade mailto:bmead...@vt.edu>>
Enviado el: lunes, 25 de marzo de 2019 23:39
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>; 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Can you send the trace file you pulled the bad call from?

Is MTP Required set on the SIP Trunk?

On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
that was updated (a local in-house development, called Mitrol). The system 
worked fine before the upgrade, and after that it went bonkers.

De: Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

we are seeing a similar issues to one of our nodes. we did our during 
production, Brave but totally doable. After figuring out that we needed to 
point the EM profiles to the node we were keeping up for the upgrade, we took 
down the other ucs down, all went well for upgrade. All VM on my ucs are all 
done now, but there is this huge jitter issues that has risen from the ashes of 
the upgrade. Its as if my media RTP streams are being forked and the forking is 
causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im ju

Re: [cisco-voip] Softphones

[ROZA, Ariel]

Cisco Jabber works well on laptops and on phones. If you deploy Expressway 
servers in your infrastructure, you can use Jabber on Internet without a VPN

De: cisco-voip  En nombre de Kent Roberts
Enviado el: miércoles, 20 de marzo de 2019 18:19
Para: Lisa Notarianni 
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Softphones

Works well.   VPN or expressway.


On Mar 20, 2019, at 2:32 PM, Lisa Notarianni 
mailto:lisa.notaria...@scranton.edu>> wrote:

We have a few staff who travel and would make good use of a softphone.  We have 
not implemented them or had anyone use them before.  Call Manager version 
11.5.1.13039-1

Any good or bad experiences or words of wisdom before I begin research?

Thanks,

Lisa



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

[ROZA, Ariel]

´ll check with my customer, and report back.
I saw that negative parameter on the o= line, but I wasn´t completely certain 
how to handle it.
Thanks for the help!

De: Brian Meade 
Enviado el: martes, 26 de marzo de 2019 13:56
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Actually meant o= line is the origin line.

On Tue, Mar 26, 2019 at 12:39 PM Brian Meade 
mailto:bmead...@vt.edu>> wrote:
It's definitely failing at parsing the SDP on that invite and finding an 
invalid parameter:
07517620.001 |16:00:23.657 |AppInfo  |//SIP/SIPUdp/wait_UdpDataInd: Incoming 
SIP UDP message size 932 from 172.27.0.15:[5060]:
[1031135,NET]
INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>
Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15>
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: gquerol
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp

v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

07517621.007 |16:00:23.657 |AppInfo  
|//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - 
sdp_ret=SDP_INVALID_PARAMETER

You may need to use a SIP Normalization script to clean up what they are 
sending.

I think it's the o= line (organization line).  That's 2nd value (-835641967) 
should be a positive number I believe.  That session-id parameter is supposed 
to match NTP format- 
https://tools.ietf.org/html/rfc4566#section-5.2<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Frfc4566%23section-5.2=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C3ba4f63173ea43f148f608d6b20bf629%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892161791009351=b1UUK085P%2BCGGFW9RRZcg1MFIXkSIE8FU%2BVUtV1ntUQ%3D=0>

Maybe just check their server has NTP synced okay to start?

Thanks,
Brian Meade



On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Here´s the trace file with the bad call



De: Brian Meade mailto:bmead...@vt.edu>>
Enviado el: lunes, 25 de marzo de 2019 23:39
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>; 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Can you send the trace file you pulled the bad call from?

Is MTP Required set on the SIP Trunk?

On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
that was updated (a local in-house development, called Mitrol). The system 
worked fine before the upgrade, and after that it went bonkers.

De: Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

we are seeing a similar issues to one of our nodes. we did our during 
production, Brave but totally doable. After figuring out that we needed to 
point the EM profiles to the node we were keeping up for the upgrade, we took 
down the other ucs down, all went well for upgrade. All VM on my ucs are all 
done now, but there is this huge jitter issues that has risen from the ashes of 
the upgrade. Its as if my media RTP streams are being forked and the forking is 
causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im just 
losing a ton of packets between the nodes now that they(the pub and sub) are 
load balancing the media resources, or rather seeming to load ballance.

After some dial peer and server group re pointing, all devices finally were on 
the one node and we were able to upgrade the UCS, but the other is left to do. 
all of my CUCM

On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP 

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

[ROZA, Ariel]

I did not check all those options because the trunk was working before my 
customer upgraded the PBX
I have the logs, thats where I got the SIP messages from. I´ll try to upload 
them to the list

De: UC Penguin 
Enviado el: lunes, 25 de marzo de 2019 21:40
Para: ROZA, Ariel 
CC: Jonatan Quezada ; cisco-voip 
(cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Is there a SIP normalization profile attached to the SIP trunk used for “Failed 
Call from PBX”?

Are changes required to that profile after the remote PBX was modified?

For the “Failed Call from PBX”:
This is a SIP early offer invite. Does the CUCM trunk support early offer?

This invite has advertises it supports early media. Does the CUCM SIP trunk 
support early media?

There is no ptime listed in the SIP invite. How does CUCM know what ptime to 
use?

Are MTP resources available for this trunk?

Have you pulled CallManager SDL Logs?

On Mar 25, 2019, at 18:13, ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
that was updated (a local in-house development, called Mitrol). The system 
worked fine before the upgrade, and after that it went bonkers.

De: Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

we are seeing a similar issues to one of our nodes. we did our during 
production, Brave but totally doable. After figuring out that we needed to 
point the EM profiles to the node we were keeping up for the upgrade, we took 
down the other ucs down, all went well for upgrade. All VM on my ucs are all 
done now, but there is this huge jitter issues that has risen from the ashes of 
the upgrade. Its as if my media RTP streams are being forked and the forking is 
causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im just 
losing a ton of packets between the nodes now that they(the pub and sub) are 
load balancing the media resources, or rather seeming to load ballance.

After some dial peer and server group re pointing, all devices finally were on 
the one node and we were able to upgrade the UCS, but the other is left to do. 
all of my CUCM

On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

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

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>
Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15>
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
mailto:sip%3A86329@172.27.0.15>>
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>;tag=573234994
Date: Fri

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

[ROZA, Ariel]

My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
that was updated (a local in-house development, called Mitrol). The system 
worked fine before the upgrade, and after that it went bonkers.

De: Jonatan Quezada 
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

we are seeing a similar issues to one of our nodes. we did our during 
production, Brave but totally doable. After figuring out that we needed to 
point the EM profiles to the node we were keeping up for the upgrade, we took 
down the other ucs down, all went well for upgrade. All VM on my ucs are all 
done now, but there is this huge jitter issues that has risen from the ashes of 
the upgrade. Its as if my media RTP streams are being forked and the forking is 
causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im just 
losing a ton of packets between the nodes now that they(the pub and sub) are 
load balancing the media resources, or rather seeming to load ballance.

After some dial peer and server group re pointing, all devices finally were on 
the one node and we were able to upgrade the UCS, but the other is left to do. 
all of my CUCM

On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

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

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>
Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15>
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
mailto:sip%3A86329@172.27.0.15>>
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15>
CSeq: 1 INVITE
Allow-Events: presence
Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Content-Length: 0




SUCESSFULL CALL FROM CUCM
-
INVITE 
sip:*86329@172.27.0.12:5060<https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2F86329%40172.27.0.12%3A5060=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C70e1772c8c6d42083a1308d6b1709fcd%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891494629446120=AO4MLkjYlNIvMkLH5FTGzhrftQtRKkh4XhPrzaJRoCw%3D=0>
 SIP/2.0
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "  (3307)" 
mailto:sip%3A3307@10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: mailto:86329@172.27.0.12>>
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 
6b366f80-c981b024-4f13-1b80040a@10.4.128.27<mailto:6b366f80-c981b024-4f13-1b80040a@10.4.128.27>
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-E

Re: [cisco-voip] for cisco folk ... idp services are down for cisco utilities?

[ROZA, Ariel]

  *   Unplug coffee machine
  *   Plug router back in

Works 100% of the time

De: Lelio Fulgenzi 
Enviado el: jueves, 13 de diciembre de 2018 17:09
Para: ROZA, Ariel ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Asunto: RE: for cisco folk ... idp services are down for cisco utilities?

Looks like it’s back up.

> identity.cisco.com
Server:  google-public-dns-a.google.com
Address:  8.8.8.8

Non-authoritative answer:
Name:identity.glb-ext.cisco.com
Address:  173.37.217.6
Aliases:  identity.cisco.com

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

www.uoguelph.ca/ccs<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7CAriel.ROZA%40LA.LOGICALIS.COM%7Ca3c8f830d7df4caf3d9308d66136de9f%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636803285627241868=QxCrzdThI9fA7%2BEiCYjPzsj2PDPaereSjaqOtKybyOg%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

From: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
Sent: Thursday, December 13, 2018 3:01 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: for cisco folk ... idp services are down for cisco utilities?

I had some issues with identity.cisco.com, half an hour ago, but I don´t know 
the causes. I was tring to Access it from inside a customer´s network (via 
webex) and failed, but then tried from my network and did ok.

De: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
En nombre de Lelio Fulgenzi
Enviado el: jueves, 13 de diciembre de 2018 16:24
Para: voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] for cisco folk ... idp services are down for cisco 
utilities?

Sorry, forgot to include debug. 


> identity.cisco.com
Server:  mydnsserver
Address:  a.b.c.d

Non-authoritative answer:
Name:identity.cisco.com

> server 1.1.1.1
Default Server:  one.one.one.one
Address:  1.1.1.1

> identity.cisco.com
Server:  one.one.one.one
Address:  1.1.1.1

*** one.one.one.one can't find identity.cisco.com: Server failed
> server 8.8.8.8
Default Server:  google-public-dns-a.google.com
Address:  8.8.8.8

> identity.cisco.com
Server:  google-public-dns-a.google.com
Address:  8.8.8.8

Non-authoritative answer:
Name:identity.cisco.com


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

www.uoguelph.ca/ccs<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7CAriel.ROZA%40LA.LOGICALIS.COM%7Ca3c8f830d7df4caf3d9308d66136de9f%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636803285627241868=QxCrzdThI9fA7%2BEiCYjPzsj2PDPaereSjaqOtKybyOg%3D=0>
 | @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 Lelio Fulgenzi
Sent: Thursday, December 13, 2018 2:21 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] for cisco folk ... idp services are down for cisco 
utilities?


I’ve tried multiple browsers. Can’t log in to any cisco CCO credentialed site.

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

www.uoguelph.ca/ccs<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7CAriel.ROZA%40LA.LOGICALIS.COM%7Ca3c8f830d7df4caf3d9308d66136de9f%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636803285627241868=QxCrzdThI9fA7%2BEiCYjPzsj2PDPaereSjaqOtKybyOg%3D=0>
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Re: [cisco-voip] for cisco folk ... idp services are down for cisco utilities?

[ROZA, Ariel]

I had some issues with identity.cisco.com, half an hour ago, but I don´t know 
the causes. I was tring to Access it from inside a customer´s network (via 
webex) and failed, but then tried from my network and did ok.

De: cisco-voip  En nombre de Lelio Fulgenzi
Enviado el: jueves, 13 de diciembre de 2018 16:24
Para: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Asunto: Re: [cisco-voip] for cisco folk ... idp services are down for cisco 
utilities?

Sorry, forgot to include debug. 


> identity.cisco.com
Server:  mydnsserver
Address:  a.b.c.d

Non-authoritative answer:
Name:identity.cisco.com

> server 1.1.1.1
Default Server:  one.one.one.one
Address:  1.1.1.1

> identity.cisco.com
Server:  one.one.one.one
Address:  1.1.1.1

*** one.one.one.one can't find identity.cisco.com: Server failed
> server 8.8.8.8
Default Server:  google-public-dns-a.google.com
Address:  8.8.8.8

> identity.cisco.com
Server:  google-public-dns-a.google.com
Address:  8.8.8.8

Non-authoritative answer:
Name:identity.cisco.com


---
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 Lelio Fulgenzi
Sent: Thursday, December 13, 2018 2:21 PM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] for cisco folk ... idp services are down for cisco 
utilities?


I’ve tried multiple browsers. Can’t log in to any cisco CCO credentialed site.

---
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] No SDI traces while collecting trace files

[ROZA, Ariel]

Found this myself:


https://supportforums.cisco.com/document/135851/interleaved-traces-callmanager-9x


I am so used to see cmx.txt files.
It shows how much I try to avoid doing debugs these days


De: cisco-voip  En nombre de ROZA, Ariel
Enviado el: miércoles, 5 de diciembre de 2018 10:09
Para: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: [cisco-voip] No SDI traces while collecting trace files

Hi guys,

What can be the reason that makes RTMT not download any SDI trace along SDLs?
I am more used t osee only SDI traces.

I already checked that I have traces on, set to detailed for all nodes and 
trace is set to on.

thanks

[cid:image011.png@01D48C83.E481C020]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Flogicalis-la.webex.com%2Fjoin%2Fariel.roza=02%7C01%7Cariel.roza%40la.logicalis.com%7Ca69c27ae0ea04d05418e08d65ab2e284%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636796121697105390=p3WP4d6a54ITm83JdHW6nQ2K1K8sc26mBEYmV07bqTY%3D=0>
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.la.logicalis.com%2F=02%7C01%7Cariel.roza%40la.logicalis.com%7Ca69c27ae0ea04d05418e08d65ab2e284%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636796121697115399=uQNfet5PvluDPYl6OQ%2Bz3KBSzN89ZySZDGVQ36rszo8%3D=0>
Business and technology working as one
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Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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[cisco-voip] No SDI traces while collecting trace files

[ROZA, Ariel]

Hi guys,

What can be the reason that makes RTMT not download any SDI trace along SDLs?
I am more used t osee only SDI traces.

I already checked that I have traces on, set to detailed for all nodes and 
trace is set to on.

thanks

[cid:image001.png@01D48C82.88BE0A60]Ariel Roza
Support & Maintenance Engineer | Latam
t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex: 
https://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
Business and technology working as one
[cid:image003.jpg@01D48C82.88BE0A60][cid:image006.png@01D298D0.103CDF50][cid:image007.png@01D47689.672FCCD0][cid:image005.png@01D47689.672FCCD0][cid:image006.png@01D47689.672FCCD0][cid:image008.png@01D47689.672FCCD0][cid:image009.png@01D47689.672FCCD0]
[cid:image010.jpg@01D47689.672FCCD0]
Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma

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

[ROZA, Ariel]

Daniel,

I never thanked you for this info.
Yes, SERVER2 is an actual CUCM, so I presume it was put there by mistake. One 
thing I miss from the documentation is a reference default configuration of all 
the certs in a cluster.

Thanks by the bucketload!

Ariel.

De: Daniel Pagan [mailto:dpa...@fidelus.com]
Enviado el: jueves, 25 de octubre de 2018 10:09 a.m.
Para: ROZA, Ariel ; James Andrewartha 
; cisco-voip@puck.nether.net
Asunto: RE: [EXT] Re: [cisco-voip] How to handle expired Phone-VPN-trust, 
phone-SAST-trust, other certificates

In your example, the SERVER2 certificate in phone-vpn-trust is there because 
someone would have placed it there for some reason. Some additional info... 
certificates uploaded to the phone-vpn-trust store can be associated with a VPN 
gateway in /ccmadmin. When assigned to a VPN-enabled phone through a common 
phone profile, a hash of the certificate is provided to the phone in its .cnf 
file. This certificate would/should be the same SSL cert assigned to the VPN 
gateway(s) configured. During the TLS handshake between the phone and the ASA, 
the phone compares the SHA1 hash of the identity certificate it receives with 
the hash contained in its previously downloaded config file.

With that said -
Why is there SERVER2.DER in the phone-vpn-trust store?
DP: Likely someone placed it there.

Is this expected?
DP: Not by default.

Does a phone contact SERVER2 while using the Phone VPN?
DP: Only if SERVER2 is the VPN gateway. The phone uses the VPN gateway URL to 
determine where to connect, then compares the certificate hash during TLS 
negotiation.

Is there by default, or someone added, even by mistake?
DP: Added and (if SERVER2 is a UC server) likely by mistake.

Hope this helps.

- Dan


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of ROZA, Ariel
Sent: Tuesday, October 23, 2018 11:52 AM
To: James Andrewartha 
mailto:jandrewar...@ccgs.wa.edu.au>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [EXT] Re: [cisco-voip] How to handle expired Phone-VPN-trust, 
phone-SAST-trust, other certificates

My main issue is not about the deletion process, but about the purpose and 
usefulness of each of those certificates. Being able to judge if it is good to 
delete or not certain certificates (even when expired).

I have this guide:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200199-CUCM-Certificate-Regeneration-Renewal-Pr.htm<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Fsupport%2Fdocs%2Funified-communications%2Funified-communications-manager-callmanager%2F200199-CUCM-Certificate-Regeneration-Renewal-Pr.htm=02%7C01%7CAriel.ROZA%40LA.LOGICALIS.COM%7C19f5c667111d4a9a6d6608d63a7b11a8%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636760697594134119=%2F6ScEcs1JeCxB%2B5%2FsVDHMxcQJN%2FfNPJ8vmzC3ClizXM%3D=0>

that gives a description of the purpose of each store, but it does not give 
specifics on why is there a particular  certificate in a store. Ie. Why is 
there SERVER2.DER in the phone-vpn-trust store? Is this expected? Does a phone 
contact SERVER2 while using the Phone VPN? Is there by default, or someone 
added, even by mistake?

And the expired certs that I have are not some that are renewable. All of them 
are in -trust stores.

So I am quite puzzled about them.

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de James 
Andrewartha
Enviado el: martes, 23 de octubre de 2018 12:39 a.m.
Para: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] How to handle expired Phone-VPN-trust, 
phone-SAST-trust, other certificates

And if you have any problems deleting them (I had one that just would not go 
away and gave me alarms for years), just call TAC and they'll take you through 
the SQL to kill them permanently.

On 23/10/18 03:08, NateCCIE wrote:
The expired certs will throw alarms even if they have been superseded by newer 
certs.

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

-Nate

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

Hi, guys!

I have a customer that is receiving alarms over some expired certificates, and 
I would like to know which is the best way to handle them.
The certs are loaded in SERVER1 and all named SERVER2.der, except the CAPF ones.
.der in phone-vpn-trust.
 .der in phone-trust
.der in phone-SAST-trust
.der in phone-CTL-trust
And several CAPF-xx.der in Call

Re: [cisco-voip] Expressway version X8.11.x

[ROZA, Ariel]

From the 8.11 release notes:

Factory Reset of Peer Leaving Cluster
From X8.11 we have modified the behavior of cluster peers when they are removed 
from the cluster or when the
cluster is disbanded. This change is part of the unique root of trust 
improvement in X8.11. To remove a peer from a
cluster you clear all peer address fields on that peer. When you do this, from 
X8.11, the Expressway prepares itself
to factory reset on the next restart (and displays a banner to remind you that 
it is in this state).
If you need to avoid the factory reset, restore the clustering peer address 
fields as they were. Replace the original
peer addresses in the same order, and then save the configuration to clear the 
banner.
The factory reset is automatically triggered when the peer restarts, to remove 
sensitive data and clustering
configuration. The reset clears all configuration except the following items, 
which are preserved so that you can still
access the Cisco Expressway:
■ IP addresses preserved
■ Server certificate, associated private key, and CA trust store preserved
■ Admin and root accounts and passwords preserved
■ SSH keys preserved
■ Option keys preserved
■ HTTPS access enabled
■ SSH access enabled
CAUTION: You MUST follow the published clustering guidance when forming, 
changing, or upgrading
Expressway clusters. Your cluster may be unrecoverable and you may lose data if 
you do not follow the correct
sequence. See the Cisco Expressway Cluster Creation and Maintenance Deployment 
Guide, for your version, on the
Cisco Expressway Series configuration guides page

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Lelio 
Fulgenzi
Enviado el: miércoles, 24 de octubre de 2018 10:59 a.m.
Para: Dana Tong ; voip puck 

Asunto: Re: [cisco-voip] Expressway version X8.11.x

As a new Expressway user, I’m really interested in what this is about. There 
isn’t even a “factory default” option in any of the menus I checked. I thought 
maybe you could accidentally check this and it required a restart.

Very odd. Very concerning.

---
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 Dana Tong
Sent: Wednesday, October 24, 2018 2:50 AM
To: voip puck mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Expressway version X8.11.x

What is it with new Expressway code?

I have a cluster that was working happily and no one touched the config but all 
of a sudden it says that it is going to factory reset itself.

Is there any way to avoid this?

[cid:image002.png@01D46BA3.F4F8BBD0]
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Re: [cisco-voip] How to handle expired Phone-VPN-trust, phone-SAST-trust, other certificates

[ROZA, Ariel]

My main issue is not about the deletion process, but about the purpose and 
usefulness of each of those certificates. Being able to judge if it is good to 
delete or not certain certificates (even when expired).

I have this guide:
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200199-CUCM-Certificate-Regeneration-Renewal-Pr.htm

that gives a description of the purpose of each store, but it does not give 
specifics on why is there a particular  certificate in a store. Ie. Why is 
there SERVER2.DER in the phone-vpn-trust store? Is this expected? Does a phone 
contact SERVER2 while using the Phone VPN? Is there by default, or someone 
added, even by mistake?

And the expired certs that I have are not some that are renewable. All of them 
are in -trust stores.

So I am quite puzzled about them.

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de James 
Andrewartha
Enviado el: martes, 23 de octubre de 2018 12:39 a.m.
Para: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] How to handle expired Phone-VPN-trust, 
phone-SAST-trust, other certificates

And if you have any problems deleting them (I had one that just would not go 
away and gave me alarms for years), just call TAC and they'll take you through 
the SQL to kill them permanently.

On 23/10/18 03:08, NateCCIE wrote:
The expired certs will throw alarms even if they have been superseded by newer 
certs.

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

-Nate

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

Hi, guys!

I have a customer that is receiving alarms over some expired certificates, and 
I would like to know which is the best way to handle them.
The certs are loaded in SERVER1 and all named SERVER2.der, except the CAPF ones.
.der in phone-vpn-trust.
 .der in phone-trust
.der in phone-SAST-trust
.der in phone-CTL-trust
And several CAPF-xx.der in Callmanager-trust

So far I have dealt with renewing Callmanager, TFTP and TVS cert, but I always 
kept clear from those other certs
Shoud I delete them, shoud I keep them, even as they are expired and throwing 
alarms?


Regards.


Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458
c: +54 9 11 5017-4417 webex: 
http://logicalis-la.webex.com/join/ariel.roza<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flogicalis-la.webex.com%2Fjoin%2Fariel.roza=02%7C01%7Cariel.roza%40la.logicalis.com%7C42e5247c66914b1d315c08d638992622%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636758627765789267=cqchqbY98HGTZ4rDIEBWzaoBX2dPJkE8dCnqeu%2BmSXA%3D=0>
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.la.logicalis.com%2F=02%7C01%7Cariel.roza%40la.logicalis.com%7C42e5247c66914b1d315c08d638992622%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636758627765789267=gJhPidfXD%2BeH0mg8xm0p1NRM7RmDRZ%2BWZouhlcUEgFE%3D=0>
_
Business and technology working as one
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  [Descripción: Descripción: Descripción:Descripción: 
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Descripción: Descripción: D

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

[ROZA, Ariel]

Hi, guys!

I have a customer that is receiving alarms over some expired certificates, and 
I would like to know which is the best way to handle them.
The certs are loaded in SERVER1 and all named SERVER2.der, except the CAPF ones.
.der in phone-vpn-trust.
 .der in phone-trust
.der in phone-SAST-trust
.der in phone-CTL-trust
And several CAPF-xx.der in Callmanager-trust

So far I have dealt with renewing Callmanager, TFTP and TVS cert, but I always 
kept clear from those other certs
Shoud I delete them, shoud I keep them, even as they are expired and throwing 
alarms?


Regards.


Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458
c: +54 9 11 5017-4417 webex: http://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
_
Business and technology working as one
[cid:image003.png@01D3894B.346BF840]

[cid:image005.png@01D3894B.43930F20]

[cid:image008.jpg@01D46A16.CC3EA7B0][Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
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Descripción: tw] [Descripción: Descripción: 
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Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
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Re: [cisco-voip] what are the cool kids using to read CUCM logs nowadays?

[ROZA, Ariel]

I also found VERY useful to use grepWin 
(https://tools.stefankueng.com/grepWin.html) when tring to find something in 
multiple files. It´s several times faster than Notepad++´s multifile search.

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Lelio 
Fulgenzi
Enviado el: jueves, 17 de mayo de 2018 01:20 p.m.
Para: Florian Kroessbacher ; Anthony Holloway 
; Ryan Huff 
CC: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Asunto: Re: [cisco-voip] what are the cool kids using to read CUCM logs 
nowadays?

Ah. Interesting. Thanks Florian!

---
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: Florian Kroessbacher 
>
Sent: Thursday, May 17, 2018 12:15 PM
To: Anthony Holloway 
>; Ryan 
Huff >; Lelio Fulgenzi 
>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
>
Subject: Re: [cisco-voip] what are the cool kids using to read CUCM logs 
nowadays?

Maybe this is something

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


--

Florian Krößbacher

florian.kroessbac...@gmail.com

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 [https://s3.amazonaws.com/htmlsig-assets/grey/googleplus.png] 

  [https://s3.amazonaws.com/htmlsig-assets/grey/linkedin.png] 


Am 17. Mai 2018, 18:12 +0200 schrieb Lelio Fulgenzi 
>:
Awesome info! Thanks.

My hope is to just get an idea of what’s going on. But I agree, I usually tend 
to go to the TAC for more detailed help.


---
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: Anthony Holloway 
>
Sent: Thursday, May 17, 2018 10:56 AM
To: Ryan Huff >
Cc: Lelio Fulgenzi >; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
>
Subject: Re: [cisco-voip] what are the cool kids using to read CUCM logs 
nowadays?

I also second TranslatorX as 

Re: [cisco-voip] Cisco Jabber, UDS, High CPU

[ROZA, Ariel]

Hi, Ryan.

CUCM/Presence  is 11.5 SU2. If those messages appear in the default log level, 
I haven´t seen any.
And I am not sure the bug applies, as I havn´t been told of any general 
degradation, besides the 20-min CPU spike.
As a fix/workaround I asked my customer to add multiple SRV records for 
cisco-uds. (they have only one, pointing to the affected subscriber).
The internal Jabber clients use EDI, but all the cucm-uds calls come from the 
Expressway and MRA Jabber clients.

Regards,

Ariel.

De: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Enviado el: lunes, 16 de abril de 2018 01:26 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: cisco-voip list <cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Cisco Jabber, UDS, High CPU

What version of UCM are you on? If it’s 11.5 SU3 check informix ccm.log for 
something like this:
23:13:59 SCHAPI: last statement aus_refresh_stats(integer,integer)
23:13:59 SCHAPI: [Auto Update Statistics Refresh 43-2] Error -217 Column 
(aus_cmd_dbs_priority) not found in any table in the query (or SLV is 
undefined).

This is CSCva78144.

I ask because I’m pretty sure one of those last two numbers is the time taken 
to process the request, and neither of those are a good number.

-Ryan

On Apr 13, 2018, at 2:22 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:

Hi, guys and gals.

I am trying to troubleshoot a High CPU Alarm on a CUCM Subscriber:

I ran the “utils diagnose test” command and got warnings on the tomcat_sessions 
module. gathered all the  tomcats logs around the time of the event and found 
lots of  GET /cucm-uds/users HTTP/1.1 200 1383 1611 queries.

In the Bug Search tool I found several bugs for old jabber versions regarding 
this, one enhancement request but no fixes.

I was wondering if it is possible to load balance the UDS queries around the 
cluster using several DNS SRV records for _cisco-uds._tcp with the same 
priority /weight to load balance the queries, or if this breaks functionality, 
somehow. The only examples I found are ones with different weight/priority.

Regards,

Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458
c: +54 9 11 5017-4417 webex: 
http://logicalis-la.webex.com/join/ariel.roza<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flogicalis-la.webex.com%2Fjoin%2Fariel.roza=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C4b35caa30dc64d803bde08d5a3b6b99f%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C1%7C636594927564159330=ZQi5sFrZq917itDJw9c%2BMgeuLZzKVI0Qh%2FJhV4bZVKY%3D=0>
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
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Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
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El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
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[cisco-voip] Cisco Jabber, UDS, High CPU

[ROZA, Ariel]

Hi, guys and gals.

I am trying to troubleshoot a High CPU Alarm on a CUCM Subscriber:

I ran the "utils diagnose test" command and got warnings on the tomcat_sessions 
module. gathered all the  tomcats logs around the time of the event and found 
lots of  GET /cucm-uds/users HTTP/1.1 200 1383 1611 queries.

In the Bug Search tool I found several bugs for old jabber versions regarding 
this, one enhancement request but no fixes.

I was wondering if it is possible to load balance the UDS queries around the 
cluster using several DNS SRV records for _cisco-uds._tcp with the same 
priority /weight to load balance the queries, or if this breaks functionality, 
somehow. The only examples I found are ones with different weight/priority.

Regards,

Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458
c: +54 9 11 5017-4417 webex: http://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
_
Business and technology working as one
[cid:image003.png@01D3894B.346BF840]

[cid:image005.png@01D3894B.43930F20]

[cid:image004.jpg@01D3D33B.44B8CF90][Descripción: Descripción: Descripción: 
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  [Descripción: 
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Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: yt] 


Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma.

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Re: [cisco-voip] Renewing TVS and IPSec certificates

[ROZA, Ariel]

Thank you, both ☺

De: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Enviado el: martes, 27 de marzo de 2018 04:13 p.m.
Para: Brian Meade <bmead...@vt.edu>; ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Renewing TVS and IPSec certificates

You also don’t need to restart the TFTP service when regenerating TVS certs, in 
fact it can cause problems depending on how quickly you pull the trigger and 
how many phones you have.

-Ryan

On Mar 27, 2018, at 1:37 PM, Brian Meade 
<bmead...@vt.edu<mailto:bmead...@vt.edu>> wrote:

If just doing IPSec, you just need to restart the DRS services.  No affect on 
ITL.

As soon as you regenerate TVS on any node, all phones will reboot and ITL will 
be updated instantly.  Be very careful to do this one node at a time making 
sure all phones re-register before moving on.  Really only a huge issue if you 
are doing the CallManager.pem at the same time.

TVS certs aren't in the CTL last I checked so you shouldn't need to regenerate 
the CTL on mixed-mode clusters if just updating TVS/IPSec.

On Tue, Mar 27, 2018 at 1:03 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi guys,

I have some specific questions regarding renewing some certificates.
The situation is this. In a cluster with 4 servers, I have two of them that 
have their tvs.pem and ipsec.pem certificates paste their due date.
I have reviewed a lot of the documentation available, and designed a 
step-by-step procedure to renew them, doing so one server at a time, stopping 
the TFTP, etc.

But still I have some specific doubts:

If I just want to renew the ipsec.pem cert ¿Do I have to go through the 
complete process to renew the ITL and reset the phones to get it? Or can I just 
renew the self-signed cert and restart the DRS Services?
If I am using mixed mode ¿Do I have to regenerate the CTL file as well?

Regards,

Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458<tel:+54%2011%205282-0458>
c: +54 9 11 5017-4417<tel:+54%209%2011%205017-4417> webex: 
http://logicalis-la.webex.com/join/ariel.roza<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flogicalis-la.webex.com%2Fjoin%2Fariel.roza=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C5938325dd18c4a3dbffa08d59416c5ce%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C1%7C636577747920378175=U8nZXPD%2FWP%2B4NPv4kDLgy5fEywLuSL03sh9wpiFY%2FTs%3D=0>
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El contenido del presente correo electrónico e inclusive sus anexos contienen 
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[cisco-voip] Renewing TVS and IPSec certificates

[ROZA, Ariel]

Hi guys,

I have some specific questions regarding renewing some certificates.
The situation is this. In a cluster with 4 servers, I have two of them that 
have their tvs.pem and ipsec.pem certificates paste their due date.
I have reviewed a lot of the documentation available, and designed a 
step-by-step procedure to renew them, doing so one server at a time, stopping 
the TFTP, etc.

But still I have some specific doubts:

If I just want to renew the ipsec.pem cert ¿Do I have to go through the 
complete process to renew the ITL and reset the phones to get it? Or can I just 
renew the self-signed cert and restart the DRS Services?
If I am using mixed mode ¿Do I have to regenerate the CTL file as well?

Regards,

Ariel Roza
Collaboration Support Engineer
t: +54 11 5282-0458
c: +54 9 11 5017-4417 webex: http://logicalis-la.webex.com/join/ariel.roza
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
_
Business and technology working as one
[cid:image003.png@01D3894B.346BF840]

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[cid:image004.jpg@01D3C5D4.570377A0][Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: tw] [Descripción: Descripción: 
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Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: fb] 
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Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: yt] 


Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma.

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[cisco-voip] Info for answering a pen test report on Expressway

[ROZA, Ariel]

Guys/gals,

A customer of mine had a pen test performed on their Expressway server and 
asked me for advice on correcting the issues reported.

There are three things to fix:

a) SSL 64-bit Block Size Cipher Supported (SWEET32)
b) SSL Medium Strength Cipher Suites Supported (Should support only high 
strength)
c) Clickjacking: X-Frame-Options header missing

For a) I already checked that need an upgrade to version 8.8.3 or later
For b) I suppose I have to review their security/SSL settings,
But what about c)? Is there a way to verify, or is it documented anywhere, 
which HTTP headers are supported by the platform, and better yet, in which 
version? I have searched thorugh the site, and saw several Expressway debugs 
that show the header being used, but have no reference points like version 
numbers, or similar.

My customer Expressway version is 8.8.2

Regards,

Ariel.

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Re: [cisco-voip] Configuring intercom in DX80

[ROZA, Ariel]

Hi, Ryan.

Thanks, that seems to be it. With your info, I was able to find  a document 
explaining how to migrate from Android software to CE software and viceversa.
https://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/ce82/dx80-dx70-convert-between-CE-android-based-software.pdf

Much appreciated.


De: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Enviado el: martes, 02 de enero de 2018 12:27 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: cisco-voip list <cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Configuring intercom in DX80

If you are running CE on the DX (which I think you are) then intercom isn’t a 
supported feature. It’s only available on the Android software.


-Ryan

On Dec 28, 2017, at 9:15 AM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:

Hi!

Has anyone configured intercom in a DX80 before?

I am trying to set it up on a DX90 with 9.1.4 firmware but I´m running into an 
issue that’s driving me nuts.

The device is taking the “standard phone button template” and its copies. which 
have only has one, unmodifiable, line.
There is a second PBT defined in CUCM, called “Cisco DX80 SIP” that has more 
lines, but the phone doesn´t list it as a vaild option in the PBT drop-down 
list.

The CUCM version Is 9.1. The DX80 firmware is 9.1.4, and the date sheet says 
that the feature is supported on the device.

¿Has anyone done it before?

Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458
c: +54 9 11 5017-4417
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.la.logicalis.com%2F=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C2c8ba9101c8c43367b4b08d551f538d7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C1%7C636505036029476469=LyMTH1N4oCZkIm1WcAxyhqgFzWnO4C9V%2FWwsltXysI8%3D=0>
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[cisco-voip] Configuring intercom in DX80

[ROZA, Ariel]

Hi!

Has anyone configured intercom in a DX80 before?

I am trying to set it up on a DX90 with 9.1.4 firmware but I´m running into an 
issue that's driving me nuts.

The device is taking the "standard phone button template" and its copies. which 
have only has one, unmodifiable, line.
There is a second PBT defined in CUCM, called "Cisco DX80 SIP" that has more 
lines, but the phone doesn´t list it as a vaild option in the PBT drop-down 
list.

The CUCM version Is 9.1. The DX80 firmware is 9.1.4, and the date sheet says 
that the feature is supported on the device.

¿Has anyone done it before?

[cid:image002.jpg@01D37FCD.329FE800]Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458
c: +54 9 11 5017-4417
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
_
Business and technology working as one
[cid:image006.png@01D298D0.103CDF50]

[cid:image005.jpg@01D37FCD.329FE800][Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: tw] [Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: fb] 
  [Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: Descripción: Descripción: Descripción: 
Descripción: Descripción: Descripción: yt] 


Logicalis Argentina S.A. solo puede ser obligado por sus representantes legales 
conforme los límites establecidos en el acto constitutivo y la legislación en 
vigor.
El contenido del presente correo electrónico e inclusive sus anexos contienen 
información confidencial.
El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto al 
destinatario, ni puede ser copiado de cualquier forma.

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Re: [cisco-voip] ITL File regeneration question

[ROZA, Ariel]

I just made a side to side compare of the old itl vs. the new itl and the only 
difference in the signature field:

Old ITL:
12SIGNATURE256
42  78  1c  54  ff  6e  d3  aa
11  a2  1e  7a  92  73  fd  cd
7d  18  4e  10  c7  ee  c6  44
21  d2  99  29  79  56  3e  6
4b  74  d1  7a  ad  dd  7c  49
38  ed  33  9d  f6  a5  c1  9d
7a  b4  aa  77  e1  5f  2d  c5
32  f6  40  82  a5  6c  14  2b
ba  c8  38  4e  ba  f8  92  fe
3  d2  57  11  79  33  37  35
9c  4e  24  db  b8  17  3d  b7
84  ef  21  18  e5  2c  32  d0
e  90  fa  d7  98  6a  a6  f2
a6  3c  1a  64  85  49  30  c0
82  0  c5  4e  73  80  cf  c8
76  d4  65  b9  9d  6d  82  bc
c  b4  f9  27  8b  ad  37  3a
ec  3d  af  d2  3d  37  80  21
81  80  b9  89  f6  d  10  51
65  9d  52  1  f4  13  95  15
1  9f  b5  8f  96  19  56  27
af  b0  ba  49  95  44  b5  da
f8  85  a5  4  ed  53  d7  53
49  8a  1a  f0  19  e3  69  37
42  2a  40  94  5  cb  70  b4
77  31  60  6d  1b  94  45  b5
df  7  bb  3a  cb  c2  53  9c
d1  5  e2  33  b6  af  ce  f7
81  d2  43  84  68  19  aa  32
33  c  7f  4a  45  42  b6  a6
ed  c9  90  1a  5e  83  92  4d
1d  41  5d  10  c5  dd  3b  34

Versus

New ITL:
12SIGNATURE256
92  a0  c4  85  4f  1f  5f  11
f  41  1a  2f  b9  2b  2d  fd
a7  e0  6a  f8  32  8  bc  31
6e  40  9a  7a  c4  74  23  af
49  26  34  5  2c  8b  81  e0
82  57  8b  f7  9c  57  63  d8
2d  77  e2  eb  f6  b8  bb  84
d6  2  92  bf  fa  5c  5  71
8  e4  b2  d2  cf  32  2a  ac
f4  fa  ac  fd  74  7  67  ff
76  1b  13  9b  82  a7  39  b7
c6  76  a3  6d  54  a8  ed  3f
18  ae  e9  8e  8d  f5  5b  c4
d2  2f  e1  29  ac  24  89  40
c8  ae  db  a3  3b  83  a8  33
44  21  fb  b9  54  fe  f2  29
b1  dc  9b  aa  7e  59  98  b4
54  6b  f9  38  dd  e7  fc  a7
93  fc  86  6d  ea  c1  c6  33
8b  91  63  9a  d  1d  18  a6
f0  51  9e  96  3  f  68  13
7e  ef  ba  ba  e2  9  59  d1
31  2a  1e  71  65  a8  ea  d2
8  eb  16  5  76  8c  b1  73
d7  f7  ba  21  c6  cd  38  46
2e  9d  eb  50  18  8f  b5  dc
af  a8  57  e9  6f  13  3a  c3
96  bd  c4  58  56  6  5b  f7
f9  85  41  b1  f4  c3  34  59
d1  86  ed  82  bc  e0  fd  61
7  4b  ca  bd  47  81  2c  22
d2  a1  4a  29  c9  aa  3a  0

The rest is identical.

De: bmead...@gmail.com [mailto:bmead...@gmail.com] En nombre de Brian Meade
Enviado el: martes, 21 de noviembre de 2017 05:32 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] ITL File regeneration question

TFTP Restart will regenerate the ITL but it shouldn't actually change the 
contents if there wasn't any certificate/hostname changes.

On Tue, Nov 21, 2017 at 3:28 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, people!

A quick question for which I can´t find a clear answer (I can infer it, but I 
need to be certain):

Is the ITL file regenerated only when an action triggers it, or it is also 
regenerated when restarting the TFTP?

So far, I know that the ITL is regenerated when there´s an IP or hostname 
change, or a certificate is regenerated/updated/deleted, but… does it change 
when I restart the TFTP or should it remain static?

i.e.

I am troubleshooting an install I am not familiar with. And I ran a “show ilt” 
and it showed me this timestamp:

The ITL File was last modified on Mon Oct 30 16:55:23 PET 2017

Then, I restarted the TFTP service and see this timestamp:

The ITL File was last modified on Tue Nov 21 14:07:04 PET 2017

Does that mean that between those dates something related to the certificates 
has changed or is it just the TFTP restart that does it?
I compared the ITL file before and after and seems similar. Also all the certs 
are valid and not one looks regenerated or replaced in recent times.


Regards,

Ariel.



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[cisco-voip] ITL File regeneration question

[ROZA, Ariel]

Hi, people!

A quick question for which I can´t find a clear answer (I can infer it, but I 
need to be certain):

Is the ITL file regenerated only when an action triggers it, or it is also 
regenerated when restarting the TFTP?

So far, I know that the ITL is regenerated when there´s an IP or hostname 
change, or a certificate is regenerated/updated/deleted, but... does it change 
when I restart the TFTP or should it remain static?

i.e.

I am troubleshooting an install I am not familiar with. And I ran a "show ilt" 
and it showed me this timestamp:

The ITL File was last modified on Mon Oct 30 16:55:23 PET 2017

Then, I restarted the TFTP service and see this timestamp:

The ITL File was last modified on Tue Nov 21 14:07:04 PET 2017

Does that mean that between those dates something related to the certificates 
has changed or is it just the TFTP restart that does it?
I compared the ITL file before and after and seems similar. Also all the certs 
are valid and not one looks regenerated or replaced in recent times.


Regards,

Ariel.


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[cisco-voip] Custom TFTP server to handle Jabber config?

[ROZA, Ariel]

Hi, All!

I am trying to test something in my own Jabber Client.
I can create a Host Configuration File, but I can´t use the corporate CUCM TFTP 
server to load the file.
The documentation states that a custom TFTP server can be configured for the 
device, using the Cisco Support Field, but I can´t find info on what options or 
parameters can I put in there. The only example I have is that you can put 
"configurationfile=" to get  from the tftp, but nothing 
else. Does it accept a full URL as ?

Regards,

Ariel.

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Re: [cisco-voip] CUCM ver 11.5 off hook dial timeout

[ROZA, Ariel]

Thats the T302 parameter, found under Service Parameters ->  -> Cisco 
Callmanager (Active) -> Clusterwide Parameters

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de James 
Dust
Enviado el: viernes, 10 de noviembre de 2017 02:50 p.m.
Para: cisco-voip@puck.nether.net
Asunto: [cisco-voip] CUCM ver 11.5 off hook dial timeout

Dear all,

We have been asked to look into the possibility of changing the amount of time 
the system takes to dial an off hook call, after the final digit is dialled on 
the phone (we have timed this and it is currently around 8 seconds)

Obviously this is a system wide parameter, but can someone please confirm to me 
which exactly?

Thanks in advance.

J

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Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

[ROZA, Ariel]

Sorry, guys.

I found the explanation. I already had it in this technote:
https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-windows/200637-Avoid-Jabber-Authentication-with-the-Web.html


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de ROZA, 
Ariel
Enviado el: lunes, 06 de noviembre de 2017 02:35 p.m.
Para: Brian Meade <bmead...@vt.edu>; Ben Amick <bam...@humanarc.com>
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

Is this applicable to Jabber for Android? Or does JfA avoids Webex? I was able 
to log to Jabber avoiding Webex after the first registration, but I am still 
unable to log on from Android (All DNS Records are in Place).

Have a nice day!

Ariel.

De: bmead...@gmail.com<mailto:bmead...@gmail.com> [mailto:bmead...@gmail.com] 
En nombre de Brian Meade
Enviado el: viernes, 03 de noviembre de 2017 02:38 p.m.
Para: Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>>
CC: ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

Yea, the problem here is the clients don't download the jabber-config.xml until 
they connect to CUCM which is later.

For first login, you'll need to use provisioning URLs, install switches, or 
make a custom MSI.

On Fri, Nov 3, 2017 at 1:33 PM, Ben Amick 
<bam...@humanarc.com<mailto:bam...@humanarc.com>> wrote:
Until the first registration they don’t get the service exclusion. If the 
client has any webex accounts it will attempt to connect to those first (or has 
had in the past – you have to put in a formal support ticket to get it 
removed). There’s a command line switch you can use on the installation of 
Jabber that will disable the webex on the initial install, and that’s the fix 
if they want to maintain having webex business accounts.

Alternatively, you might be able to get a support ticket in to remove your 
domain from the webex DNS, which the company gets added to because of the cloud 
messaging built into webex, but I don’t know if they can neuter that without 
neutering the webex accounts proper.

Ben Amick
Unified Communications Analyst

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of ROZA, Ariel
Sent: Friday, November 3, 2017 1:27 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] jabber-config.xml not being downloaded or being ignored

I have a preexisting installation CUCM 10.5.2 that I am troubleshooting.

Jabber for Windows clients ask for Webex credentials for login, although the 
jabber-config.xml file exists and is configured to exclude Webex services.
I have checked every possibility and can´t see what I´m missing here.

The jabber-config contents are this:




WEBEX


UDS



I have tried Jabber clients 11.6.4 and 11.8.2

The file is loaded to both TFTP serves in the cluster
It´s accessible via URL
The name is lowercase and the format is UTF-8
I restarted the TFTP servers when needed
The Cisco Support Field in the CSF device is empty.
In the Jabber Client I used the Diagnostics Tool (CTRL-SHIFT-D) and checked 
that all setings are ok.
I reset the client, cleared all the files under %APPDATA%\Local\Cisco\ and 
%APPDATA%\Roaming\Cisco and restarted the client.
The Jabber Client can log on, if done manually, specifiing the "Callmanager 9 
"mode and the server name/IP. If I switch to Automatic, it asks for Webex 
credentials (My customer has a couple of Webex accounts, but not enough for all 
the enterprise, hence he wants avoid Webex).
I checked the jabber.log, but I didn't get any clues.

Is there anything I am missing from the process?

P.S I also posted this question here: 
https://supportforums.cisco.com/t5/jabber-clients/jabber-config-xml-not-being-downloaded-or-being-ignored/m-p/3209901#M20756<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsupportforums.cisco.com%2Ft5%2Fjabber-clients%2Fjabber-config-xml-not-being-downloaded-or-being-ignored%2Fm-p%2F3209901%23M20756=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccdb52d32fa194c59bb0308d522e1a54e%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453274894316521=RDHI4Z%2BE9TgUffC8YxJahReZbRqBWnZSykyOHwdLC7U%3D=0>


Regards,

Ariel.


[cid:image007.jpg@01D35712.5A8CA170]Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458<tel:+54%2011%205282-0458>
c: +54 9 11 5017-4417<tel:+54%209%2011%205017-4417>
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcp.mcafee.com%2Fd%2Favndz8Qd2hJ5-VEV7cTKUedTdEEILIFzxO9EVdEEILIFzDDSjqabbXPUVVZMSyyUqejhPuZQrCO6GOIl1hdJno

Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

[ROZA, Ariel]

Is this applicable to Jabber for Android? Or does JfA avoids Webex? I was able 
to log to Jabber avoiding Webex after the first registration, but I am still 
unable to log on from Android (All DNS Records are in Place).

Have a nice day!

Ariel.

De: bmead...@gmail.com [mailto:bmead...@gmail.com] En nombre de Brian Meade
Enviado el: viernes, 03 de noviembre de 2017 02:38 p.m.
Para: Ben Amick <bam...@humanarc.com>
CC: ROZA, Ariel <ariel.r...@la.logicalis.com>; cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

Yea, the problem here is the clients don't download the jabber-config.xml until 
they connect to CUCM which is later.

For first login, you'll need to use provisioning URLs, install switches, or 
make a custom MSI.

On Fri, Nov 3, 2017 at 1:33 PM, Ben Amick 
<bam...@humanarc.com<mailto:bam...@humanarc.com>> wrote:
Until the first registration they don’t get the service exclusion. If the 
client has any webex accounts it will attempt to connect to those first (or has 
had in the past – you have to put in a formal support ticket to get it 
removed). There’s a command line switch you can use on the installation of 
Jabber that will disable the webex on the initial install, and that’s the fix 
if they want to maintain having webex business accounts.

Alternatively, you might be able to get a support ticket in to remove your 
domain from the webex DNS, which the company gets added to because of the cloud 
messaging built into webex, but I don’t know if they can neuter that without 
neutering the webex accounts proper.

Ben Amick
Unified Communications Analyst

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of ROZA, Ariel
Sent: Friday, November 3, 2017 1:27 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] jabber-config.xml not being downloaded or being ignored

I have a preexisting installation CUCM 10.5.2 that I am troubleshooting.

Jabber for Windows clients ask for Webex credentials for login, although the 
jabber-config.xml file exists and is configured to exclude Webex services.
I have checked every possibility and can´t see what I´m missing here.

The jabber-config contents are this:




WEBEX


UDS



I have tried Jabber clients 11.6.4 and 11.8.2

The file is loaded to both TFTP serves in the cluster
It´s accessible via URL
The name is lowercase and the format is UTF-8
I restarted the TFTP servers when needed
The Cisco Support Field in the CSF device is empty.
In the Jabber Client I used the Diagnostics Tool (CTRL-SHIFT-D) and checked 
that all setings are ok.
I reset the client, cleared all the files under %APPDATA%\Local\Cisco\ and 
%APPDATA%\Roaming\Cisco and restarted the client.
The Jabber Client can log on, if done manually, specifiing the "Callmanager 9 
"mode and the server name/IP. If I switch to Automatic, it asks for Webex 
credentials (My customer has a couple of Webex accounts, but not enough for all 
the enterprise, hence he wants avoid Webex).
I checked the jabber.log, but I didn't get any clues.

Is there anything I am missing from the process?

P.S I also posted this question here: 
https://supportforums.cisco.com/t5/jabber-clients/jabber-config-xml-not-being-downloaded-or-being-ignored/m-p/3209901#M20756<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsupportforums.cisco.com%2Ft5%2Fjabber-clients%2Fjabber-config-xml-not-being-downloaded-or-being-ignored%2Fm-p%2F3209901%23M20756=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccdb52d32fa194c59bb0308d522e1a54e%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453274894316521=RDHI4Z%2BE9TgUffC8YxJahReZbRqBWnZSykyOHwdLC7U%3D=0>


Regards,

Ariel.


[cid:image001.jpg@01D3570B.CE6D6950]Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458<tel:+54%2011%205282-0458>
c: +54 9 11 5017-4417<tel:+54%209%2011%205017-4417>
Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcp.mcafee.com%2Fd%2Favndz8Qd2hJ5-VEV7cTKUedTdEEILIFzxO9EVdEEILIFzDDSjqabbXPUVVZMSyyUqejhPuZQrCO6GOIl1hdJnor6TbCOrqKMSdKndFEzJIWqen-LP3atTS7eLsKCOVvxP1EV78LIth5dqWqJS7-l3PWApmU6CQjrVK_nhjjpd7dQXCzATsS02fTzWApmQ-lenMTqlblbCqOmbAaJMJZ0kIToHMd9_7wqr73DD7zpFrJaBGBPdpb6BQQg2eA27NbSH1Sh7wZkYQggHcQg4fB0yq84aC87RR3Q3h1m9Ew8va14Qgi8MH7Q3rzNI2Yfibf0=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccdb52d32fa194c59bb0308d522e1a54e%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453274894316521=JGx%2F3CJQTJ4%2B3iwm2lOtLWL%2B26fi%2BNI5l7jUMThMUrY%3D=0>
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Business and technology working as one
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Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

[ROZA, Ariel]

Thanks, Ben :)

De: Ben Amick [mailto:bam...@humanarc.com]
Enviado el: viernes, 03 de noviembre de 2017 02:39 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>; cisco-voip@puck.nether.net
Asunto: RE: jabber-config.xml not being downloaded or being ignored

I haven't tested it in a while and don't have the capability do on the domain I 
have access to, but if you reset the jabber client it might remove the 
exclusion provisioning, which you would need to do to switch users. As I said, 
don't quote me on that but it's a possibility.

Ben Amick
Unified Communications Analyst

From: ROZA, Ariel [mailto:ariel.r...@la.logicalis.com]
Sent: Friday, November 3, 2017 1:37 PM
To: Ben Amick <bam...@humanarc.com<mailto:bam...@humanarc.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: RE: jabber-config.xml not being downloaded or being ignored

I know that, but after I´ll I read, I understand that if the jabber client is 
able to log in a first time, using manual settings, it should download and 
cache the jabber-config info, hence excluding Webex from them on. Right?

My customer can manually log in.

De: Ben Amick [mailto:bam...@humanarc.com]
Enviado el: viernes, 03 de noviembre de 2017 02:33 p.m.
Para: ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: RE: jabber-config.xml not being downloaded or being ignored

Until the first registration they don't get the service exclusion. If the 
client has any webex accounts it will attempt to connect to those first (or has 
had in the past - you have to put in a formal support ticket to get it 
removed). There's a command line switch you can use on the installation of 
Jabber that will disable the webex on the initial install, and that's the fix 
if they want to maintain having webex business accounts.

Alternatively, you might be able to get a support ticket in to remove your 
domain from the webex DNS, which the company gets added to because of the cloud 
messaging built into webex, but I don't know if they can neuter that without 
neutering the webex accounts proper.

Ben Amick
Unified Communications Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of ROZA, 
Ariel
Sent: Friday, November 3, 2017 1:27 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] jabber-config.xml not being downloaded or being ignored

I have a preexisting installation CUCM 10.5.2 that I am troubleshooting.

Jabber for Windows clients ask for Webex credentials for login, although the 
jabber-config.xml file exists and is configured to exclude Webex services.
I have checked every possibility and can´t see what I´m missing here.

The jabber-config contents are this:




WEBEX


UDS



I have tried Jabber clients 11.6.4 and 11.8.2

The file is loaded to both TFTP serves in the cluster
It´s accessible via URL
The name is lowercase and the format is UTF-8
I restarted the TFTP servers when needed
The Cisco Support Field in the CSF device is empty.
In the Jabber Client I used the Diagnostics Tool (CTRL-SHIFT-D) and checked 
that all setings are ok.
I reset the client, cleared all the files under %APPDATA%\Local\Cisco\ and 
%APPDATA%\Roaming\Cisco and restarted the client.
The Jabber Client can log on, if done manually, specifiing the "Callmanager 9 
"mode and the server name/IP. If I switch to Automatic, it asks for Webex 
credentials (My customer has a couple of Webex accounts, but not enough for all 
the enterprise, hence he wants avoid Webex).
I checked the jabber.log, but I didn't get any clues.

Is there anything I am missing from the process?

P.S I also posted this question here: 
https://supportforums.cisco.com/t5/jabber-clients/jabber-config-xml-not-being-downloaded-or-being-ignored/m-p/3209901#M20756<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcp.mcafee.com%2Fd%2F5fHCNAp6xEgdELTd78VCXzXxKVJ55BZBcsehd79J55BZBcsY-Orhhpvuv7ffK6Qkn3hOqerTKzsSgRmlyEa9JGX3oSVsSjrlS6NJOVIQsI6zDreCzB_HYOeh7c3AuLsKCO-PPVEVKMYUeWyaaRQRrTjVkffGhBrwqrhdLCXYDtNwse7cFL6XCOsVHkiP8rt7GqXVYE5qsJ6I_JelotlKpZqBunMAT4YUDeqR1X8wfj5NWoLhJpqIpKRySHjHiFqLbzQNurUZcn4t84fynIvbUjEfj5N7i13UBXlwX8yn3RjOYaPbAfB0yngGowvnkfgbRyn8va14L8z2Ivg7FyWFtouCbIFTd79EVKyfjp6QXzDT6ww24dDFRQNUUiKyf72gx6nDY2MM18wfoKvL7R8OJFYGsKf72jOpEVg-ZXeIS7ccuqem3hPsQsTugvjjZ6XzSzNMADXapEVLowYwNtyXXSAfvKDpxRZUwYeZAue4HzNMBsue4D6n67SkNOqra9EVp7fenq208ExIZe2MO190b31ombkd78Zcp2UZcno7FzaE34g62Mm1dge6yMnEO90icEm2M50iudydO8ZAeq208zZfypZ0WVLoOY_ER5d1h-YvkzaSDOFOVJUsMC-r1oQAq817i13UBXlwX8zMuGuq88lCq827Owhd425j43WWxW1EwH4Qg4fB0yq894olzW1JYQsCRtZquLH-Er9=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cf28562ff18034302d81708d522e1e291%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453275924907514=aJaHFnc%2BDbTxkLb%2FoByvRB3J52h8vlU%2BwAXZNW01roY%3D=0>


Regards,

Ariel.


[cid:image007.jpg@01D354B1.C88BB520]A

Re: [cisco-voip] jabber-config.xml not being downloaded or being ignored

[ROZA, Ariel]

I know that, but after I´ll I read, I understand that if the jabber client is 
able to log in a first time, using manual settings, it should download and 
cache the jabber-config info, hence excluding Webex from them on. Right?

My customer can manually log in.

De: Ben Amick [mailto:bam...@humanarc.com]
Enviado el: viernes, 03 de noviembre de 2017 02:33 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>; cisco-voip@puck.nether.net
Asunto: RE: jabber-config.xml not being downloaded or being ignored

Until the first registration they don't get the service exclusion. If the 
client has any webex accounts it will attempt to connect to those first (or has 
had in the past - you have to put in a formal support ticket to get it 
removed). There's a command line switch you can use on the installation of 
Jabber that will disable the webex on the initial install, and that's the fix 
if they want to maintain having webex business accounts.

Alternatively, you might be able to get a support ticket in to remove your 
domain from the webex DNS, which the company gets added to because of the cloud 
messaging built into webex, but I don't know if they can neuter that without 
neutering the webex accounts proper.

Ben Amick
Unified Communications Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of ROZA, 
Ariel
Sent: Friday, November 3, 2017 1:27 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] jabber-config.xml not being downloaded or being ignored

I have a preexisting installation CUCM 10.5.2 that I am troubleshooting.

Jabber for Windows clients ask for Webex credentials for login, although the 
jabber-config.xml file exists and is configured to exclude Webex services.
I have checked every possibility and can´t see what I´m missing here.

The jabber-config contents are this:




WEBEX


UDS



I have tried Jabber clients 11.6.4 and 11.8.2

The file is loaded to both TFTP serves in the cluster
It´s accessible via URL
The name is lowercase and the format is UTF-8
I restarted the TFTP servers when needed
The Cisco Support Field in the CSF device is empty.
In the Jabber Client I used the Diagnostics Tool (CTRL-SHIFT-D) and checked 
that all setings are ok.
I reset the client, cleared all the files under %APPDATA%\Local\Cisco\ and 
%APPDATA%\Roaming\Cisco and restarted the client.
The Jabber Client can log on, if done manually, specifiing the "Callmanager 9 
"mode and the server name/IP. If I switch to Automatic, it asks for Webex 
credentials (My customer has a couple of Webex accounts, but not enough for all 
the enterprise, hence he wants avoid Webex).
I checked the jabber.log, but I didn't get any clues.

Is there anything I am missing from the process?

P.S I also posted this question here: 
https://supportforums.cisco.com/t5/jabber-clients/jabber-config-xml-not-being-downloaded-or-being-ignored/m-p/3209901#M20756<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsupportforums.cisco.com%2Ft5%2Fjabber-clients%2Fjabber-config-xml-not-being-downloaded-or-being-ignored%2Fm-p%2F3209901%23M20756=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C829de00cf3c744e9490908d522e1068a%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453272231937860=PBZbPSyqys9%2BH%2F4%2BmXgXPPToYIqHs3CZBdcPKZlMj0g%3D=0>


Regards,

Ariel.


[cid:image002.jpg@01D354B1.32D6D140]Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458
c: +54 9 11 5017-4417
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcp.mcafee.com%2Fd%2Favndz8Qd2hJ5-VEV7cTKUedTdEEILIFzxO9EVdEEILIFzDDSjqabbXPUVVZMSyyUqejhPuZQrCO6GOIl1hdJnor6TbCOrqKMSdKndFEzJIWqen-LP3atTS7eLsKCOVvxP1EV78LIth5dqWqJS7-l3PWApmU6CQjrVK_nhjjpd7dQXCzATsS02fTzWApmQ-lenMTqlblbCqOmbAaJMJZ0kIToHMd9_7wqr73DD7zpFrJaBGBPdpb6BQQg2eA27NbSH1Sh7wZkYQggHcQg4fB0yq84aC87RR3Q3h1m9Ew8va14Qgi8MH7Q3rzNI2Yfibf0=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C829de00cf3c744e9490908d522e1068a%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636453272231937860=stbIJ0jQpFHUsCmldjy0DfqgF8y3kbBe33C6bFpTLLs%3D=0>
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Business and technology working as one
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[cisco-voip] jabber-config.xml not being downloaded or being ignored

[ROZA, Ariel]

I have a preexisting installation CUCM 10.5.2 that I am troubleshooting.

Jabber for Windows clients ask for Webex credentials for login, although the 
jabber-config.xml file exists and is configured to exclude Webex services.
I have checked every possibility and can´t see what I´m missing here.

The jabber-config contents are this:




WEBEX


UDS



I have tried Jabber clients 11.6.4 and 11.8.2

The file is loaded to both TFTP serves in the cluster
It´s accessible via URL
The name is lowercase and the format is UTF-8
I restarted the TFTP servers when needed
The Cisco Support Field in the CSF device is empty.
In the Jabber Client I used the Diagnostics Tool (CTRL-SHIFT-D) and checked 
that all setings are ok.
I reset the client, cleared all the files under %APPDATA%\Local\Cisco\ and 
%APPDATA%\Roaming\Cisco and restarted the client.
The Jabber Client can log on, if done manually, specifiing the "Callmanager 9 
"mode and the server name/IP. If I switch to Automatic, it asks for Webex 
credentials (My customer has a couple of Webex accounts, but not enough for all 
the enterprise, hence he wants avoid Webex).
I checked the jabber.log, but I didn't get any clues.

Is there anything I am missing from the process?

P.S I also posted this question here: 
https://supportforums.cisco.com/t5/jabber-clients/jabber-config-xml-not-being-downloaded-or-being-ignored/m-p/3209901#M20756


Regards,

Ariel.


[cid:image002.jpg@01D354AF.D6B1C1A0]Ariel Roza
Support Engineer / Southern Cone
t: +54 11 5282-0458
c: +54 9 11 5017-4417
Av. Belgrano 955 - Piso 20 - CABA - Argentina - C1092AAJ
www.la.logicalis.com
_
Business and technology working as one
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El contenido del presente correo electrónico e inclusive sus anexos contienen 
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Re: [cisco-voip] e164 dialplan conversion

[ROZA, Ariel]

This is the link to the page that contains all the PA and CVD documents 
published so far:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/PAdocs.html?dtid=osscdc000283


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Anthony 
Holloway
Enviado el: viernes, 22 de septiembre de 2017 02:15 a.m.
Para: Ben Amick 
CC: cisco-voip voyp list 
Asunto: Re: [cisco-voip] e164 dialplan conversion

You remember the big Y2K scare of the late 90s?

Storing only the last 4 digits of an E.164 number is like storing the last two 
digits of the Year.

It's not like moving to more digits prevents users from keeping their existing 
dialing habits.

I would also recommend you read the Dial Plan section of the Preferred 
Architecture (PA) for Collaboration.  It's a quick read, like less than 10 
minutes.

https://www.cisco.com/c/dam/en/us/td/docs/solutions/PA/enterprise/11x/clbpa116.pdf

If that excited you, then move on the the PA Cisco Validated Design (CVD) 
section on End Point Addressing, and Routing and Normalization.  It's pretty 
great.

https://www.cisco.com/c/en/us/td/docs/solutions/CVD/Collaboration/enterprise/11x/116/collbcvd/control.html#pgfId-1179320

On Thu, Sep 21, 2017 at 9:24 AM Ben Amick 
> wrote:
Inexperienced person here chiming in with a question:
What advantage/purpose does switching to an e164 dial plan afford you? Is it 
just more of a flexibility to mate dial plan/DNs to DIDs?
More importantly, how would an e164 dial plan mesh with a system where the 
majority of users do not have DIDs or DIDs that do not match their extension 
plan?
Is any additional overhead generated by going to an e164 dial plan?
Also, from what I understand, 4-5digit dialing is made possible only by having 
proper CSS per site with a translation pattern, not through any function of the 
DN configuration, correct?

Ben Amick
Unified Communications Analyst

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Bernhard Albler
Sent: Thursday, September 21, 2017 10:04 AM
To: Florian Kroessbacher 
>
Cc: cisco-voip voyp list 
>
Subject: Re: [cisco-voip] e164 dialplan conversion

ucm:
What Florian says is the correct way to do it.
You can also run a sql update against the numplan table which will do 2 things:
1.)I you typoed you are in for a lot of pain
2.)Ucm will load up on change notification (assuming lots of DNs) and might 
come to halt before recovering

ccx:
Agent DNs in my experience will reflect automatically. Routepoints and CTI 
ports you will need to change manually.


Personally i suggest the SQL route as the adrenaline rush is just incredible 
once you realize you made a mistake.

On Thu, Sep 21, 2017 at 3:20 PM, Florian Kroessbacher 
> wrote:
Hy out there,

throug updateLine this is working

use old pattern & partition
and set newpattern

we have done this for 12000 Lines

Am 21. Sep. 2017, 15:13 +0200 schrieb Bill Talley 
>:
I know there have been some conversations around this in the past, but I’m 
hoping there are new methods for converting from a 4-digit to full e164 
dialplan.

Is there a way to change dialplan entries on CUCM and UCCX without having to 
either individually touch each dialplan pattern or without deleting and 
reimporting/recreating? Has anyone seen or used AXL/SOAP to automate 
modification of dialplan patterns, or used PCP or some third party product to 
accomplish this?

Thanks for any feedback.

Bill

Sent from a mobile device with very tiny touchscreen input keys. Please excude 
my typtos.
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Re: [cisco-voip] Hardware Tokens/Secure Cluster

[ROZA, Ariel]

I have done some secure clusters before (with the tokens) and I can tell you 
this:

While you can secure the basic cluster (CUCM and IP Phones) with ease, you have 
to take into account several things if you need to do a more extensive 
encryption:

- Conference usage: software bridges like CUCM may not support encryption, so 
you have to use more hardware bridges on  routers if you want encrypted 
conferences (some planning and more resources)
- Non CUCM applications require some additional work (CUACA, UCCX, Unity 
Connection, PhoneProxy, etc.). This usually involves interchanging certificates 
between servers.
- If you want to encrypt traffic to H.323 gateways  you´ll have to create VPNs
- SIP Trunks with TLS will also require you to deal with certificates.
- If you have a large quantity of devices to handle certificates (<10 
servers+gateways), you better have a PKI infrastructure put in place and well 
oiled before doing anything. You will deal with certificate renew every 2 to 5 
years

You will find the info to encrypt traffic to CUACA or UCCX in the CUCM Security 
Guide (Basically, you´ll have to encrypt CTI Ports).

And, at last, take into account that encryption adds an additional layer of 
complexity when troubleshooting everything.

Hope this helps. Anything else, just ask.


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Matthew 
Loraditch
Enviado el: martes, 29 de agosto de 2017 11:03 a.m.
Para: Brian Meade 
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Hardware Tokens/Secure Cluster

Ok yes, I’m starting to understand this.

I’m also looking for guidance re CUACA and UCCX.
I’m struggling trying to find where the instructions are for either of them.

Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518
Facebook
 | 
Twitter
 | 
LinkedIn
 | 
G+

From: bmead...@gmail.com [mailto:bmead...@gmail.com] 
On Behalf Of Brian Meade
Sent: Tuesday, August 29, 2017 9:19 AM
To: Matthew Loraditch 
>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Hardware Tokens/Secure Cluster

You can use self-signed certificates now instead with the command-line tools.  
There's still some hardware tokens if you'd rather have something physical 
rather than worrying about backing up the certificates.

You can just run "utils ctl set-cluster mixed-mode" and then restart 
CallManager/TFTP on all nodes if you want to use self-signed certs.

Here's the 10.x security guide- 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/10_0_1/secugd/CUCM_BK_C68276B4_00_cucm-security-guide-100/CUCM_BK_C68276B4_00_cucm-security-guide-100_chapter_0100.html

On Mon, Aug 28, 2017 at 8:25 PM, Matthew Loraditch 
> 
wrote:
We have a client who is requesting a secure cluster. Never done it before. Do 
those hardware tokens still exist? It appears not and it’s all software based 
now?
Any fantastic blogs or step by step guides that folks have used? The 
documentation is refreshingly mind numbing.

Thanks!
-Matthew

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[cisco-voip] IPMA 7.1.5. Error during assistant configuration

[ROZA, Ariel]

Hi, Everyone!

I am troubleshooting a Customer´s IPMA Setup.
They have an old 7.1.5 cluster that has no Cisco support but I still have to 
help them.

They have multiple Managers and Assistants working, but only one has issues.
When the user tries to log in it receives the error "System Error. Please 
contact  administrator".

At first, I tried restarting the CTI Manager, IP Manager Assistant and Tomcat 
services, but the behavior remains.

I then looked at the different Managers and Assistants and found a couple of 
differences between the working ones and the failing one. It was using a 
different DN/partition scheme. I modified everything, but when I try to make 
the last change I receive an error that does not allow me to save the change. 
The change is the following.

I go to the failing assistant´s user profile and go to the "Assistant 
configuration" section.
Then, in the lower section, Manager Association to the Assistant Line, I select 
the assistant´s second line, the Manager name and the Manager´s line.

When I hit Save, a pop-up warning appears saying "Each Cisco Unified 
Communications Manager Assistant controlled line of associated managers has to 
be associated with and assistant line" and the change is not saved.
Has anyone seen something like this? I have a last resort option of creating a 
whole new user for the assistant, or replacing the existing one. But I would 
like to try an alternate solution before that extreme.

Can anyone help?

Regards,

Ariel.

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Re: [cisco-voip] Does anyone know if there is a CLI SQL command to determine the CUCM server ID (GUID type value)

[ROZA, Ariel]

Depending on your CUCM version, you can find  the corresponding “Data 
dictionary” or “Database Dictionary” document from here:
https://www.cisco.com/c/en/us/support/unified-communications/unified-communications-manager-callmanager/products-programming-reference-guides-list.html


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Brian V
Enviado el: miércoles, 16 de agosto de 2017 05:37 p.m.
Para: cisco-voip@puck.nether.net
Asunto: [cisco-voip] Does anyone know if there is a CLI SQL command to 
determine the CUCM server ID (GUID type value)


See the below example from the web browser (You can find the Server ID in CUCM 
by logging into CM Administration in the Pub > System > Server > Select Pub. 
The server ID is the same as the Server ID listed at the end of the page URL:)
[Inline image 1]
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Re: [cisco-voip] Force an outgoing call through a subscriber

[ROZA, Ariel]

Yes, I got several captures. Some that fail and some that do not.
At the moment the wireshark analisys shows failing samples show some “jitter 
drops” but not packet loss.
But all the failing ones go through the subscriber.

I asked my customer to take a capture sample of a broken audio call from the 
subscriber and from the gateway, to see if the audio breakdown is created right 
on the subscriber or further down on the network.

De: bmead...@gmail.com [mailto:bmead...@gmail.com] En nombre de Brian Meade
Enviado el: lunes, 24 de julio de 2017 06:44 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: Ryan Huff <ryanh...@outlook.com>; Anthony Holloway 
<avholloway+cisco-v...@gmail.com>; cisco-voip <cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Probably not that bug then.  Are you able to get a capture of a call using this 
as an MTP?  Is it intermittent or always happens on this sub?

I'd check the delta on the Wireshark RTP Analysis to make sure it's not jumping 
around between 16ms and 24ms and stays steady at 20ms.

On Mon, Jul 24, 2017 at 5:29 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
The cluster is on version 9.1.2

De: bmead...@gmail.com<mailto:bmead...@gmail.com> 
[mailto:bmead...@gmail.com<mailto:bmead...@gmail.com>] En nombre de Brian Meade
Enviado el: lunes, 24 de julio de 2017 04:51 p.m.
Para: ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>>
CC: Ryan Huff <ryanh...@outlook.com<mailto:ryanh...@outlook.com>>; Anthony 
Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>; 
cisco-voip <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>

Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Ariel,

What version are you on?  There were some VMWare driver issues a long time ago 
that would cause issues with IPVMS resources- 
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtz29142<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fbst.cloudapps.cisco.com%2Fbugsearch%2Fbug%2FCSCtz29142=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccfbdb5b25a9b4fe2583708d4d2cd4b55%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636365226557540072=zUx6tJy%2BAf34fNymMdO%2BbqrJkLkJCOs3rk3%2FSkjVCE8%3D=0>

Brian

On Mon, Jul 24, 2017 at 3:31 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, Ryan, Anthony, et al,

  I´m sorry by the blunt nature of my first mail. I was burned out, at my 
customer´s.

  My problem is this:
  The users are complaining of broken audio in random calls to the PSTN

  We Isolated the issue to a single subscriber that acts as MTP for calls, 
sometimes.
Wireshark captures show us that whenever the sub acts as an intermediary, audio 
gets jitter (but no drops) This is on the upstream to the PSTN. The downstream 
is always fine.
We suspect of a problem in the underlying nework, as the Publisher and the Subs 
are wired differently and only the Subscriber show these problems.
The Publisher is connected to a CAT4500 switch and the Sub is connected to a 
NEXUS 5000  that in turn is connected to the Cat4500

I am already using MRGL/MRGs to  force MTP, but sometimes the phones send the 
RTP directly to the gateway, bypassing the MTP. (calling always the same number)


De: Ryan Huff [mailto:ryanh...@outlook.com<mailto:ryanh...@outlook.com>]
Enviado el: domingo, 23 de julio de 2017 07:03 p.m.
Para: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
CC: ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>>; cisco-voip 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Anthony,

Yes, it is splitting hairs IMO :). I'm specifically talking about a typical 
scenario where RTP is sent from/to the phone for a connected, like-codec call 
that is not forcing media termination with the next device in the call leg.

As you and I have mentioned, there are a handful of ways to use different 
features and services of a CUCM server to terminate and join media streams 
(MTP, CFB ... etc) ... and I considers these as ancillary service and component 
capabilities to the server and not a core function of the server itself (hence, 
media not flowing through the server, although it maybe interacting with one or 
more of the aforementioned media services or components).

Always happy to split hairs :).

Sent from my iPhone

On Jul 23, 2017, at 5:18 PM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
This may be splitting hair here, but two things Ryan:

1) Your first sentence reads to me like a contradiction.  Could you c

Re: [cisco-voip] Force an outgoing call through a subscriber

[ROZA, Ariel]

The cluster is on version 9.1.2

De: bmead...@gmail.com [mailto:bmead...@gmail.com] En nombre de Brian Meade
Enviado el: lunes, 24 de julio de 2017 04:51 p.m.
Para: ROZA, Ariel <ariel.r...@la.logicalis.com>
CC: Ryan Huff <ryanh...@outlook.com>; Anthony Holloway 
<avholloway+cisco-v...@gmail.com>; cisco-voip <cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Ariel,

What version are you on?  There were some VMWare driver issues a long time ago 
that would cause issues with IPVMS resources- 
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtz29142<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fbst.cloudapps.cisco.com%2Fbugsearch%2Fbug%2FCSCtz29142=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Ccfbdb5b25a9b4fe2583708d4d2cd4b55%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636365226557540072=zUx6tJy%2BAf34fNymMdO%2BbqrJkLkJCOs3rk3%2FSkjVCE8%3D=0>

Brian

On Mon, Jul 24, 2017 at 3:31 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, Ryan, Anthony, et al,

  I´m sorry by the blunt nature of my first mail. I was burned out, at my 
customer´s.

  My problem is this:
  The users are complaining of broken audio in random calls to the PSTN

  We Isolated the issue to a single subscriber that acts as MTP for calls, 
sometimes.
Wireshark captures show us that whenever the sub acts as an intermediary, audio 
gets jitter (but no drops) This is on the upstream to the PSTN. The downstream 
is always fine.
We suspect of a problem in the underlying nework, as the Publisher and the Subs 
are wired differently and only the Subscriber show these problems.
The Publisher is connected to a CAT4500 switch and the Sub is connected to a 
NEXUS 5000  that in turn is connected to the Cat4500

I am already using MRGL/MRGs to  force MTP, but sometimes the phones send the 
RTP directly to the gateway, bypassing the MTP. (calling always the same number)


De: Ryan Huff [mailto:ryanh...@outlook.com<mailto:ryanh...@outlook.com>]
Enviado el: domingo, 23 de julio de 2017 07:03 p.m.
Para: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
CC: ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>>; cisco-voip 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Anthony,

Yes, it is splitting hairs IMO :). I'm specifically talking about a typical 
scenario where RTP is sent from/to the phone for a connected, like-codec call 
that is not forcing media termination with the next device in the call leg.

As you and I have mentioned, there are a handful of ways to use different 
features and services of a CUCM server to terminate and join media streams 
(MTP, CFB ... etc) ... and I considers these as ancillary service and component 
capabilities to the server and not a core function of the server itself (hence, 
media not flowing through the server, although it maybe interacting with one or 
more of the aforementioned media services or components).

Always happy to split hairs :).

Sent from my iPhone

On Jul 23, 2017, at 5:18 PM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
This may be splitting hair here, but two things Ryan:

1) Your first sentence reads to me like a contradiction.  Could you clarify 
what you're stating here?

2) Media can flow through, and even terminate on CUCM, since things like 
Conference Bridges, MTPs and now the new IVR media resource, are all doing that.

On Fri, Jul 21, 2017 at 2:53 PM Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
So the actual media (RTP) will never flow through a CUCM server; it may 
however, terminate a connected media stream on a software based MTP application 
that CUCM runs as a service (IP Voice Media Streaming Application). Signaling 
(SIP) on the other hand, will always traverse a CUCM server.

If you see a CUCM IP address in the Audio field of the SDP, then it's likely 
terminating on a CUCM based MTP resource (most often, due to some differences 
in DTMF negotiations or because the egress path in CUCM is required to use MTP).

If you are trying to test a call using a CUCM MTP resource on a particular 
cluster node; the simplest way would be to create a new MRG/MRGL that only 
specifies MTP resources from the desired cluster node and then advertise that 
MRGL to the phone and/or egress path to the pstn for the phone and then 
"require" MTP termination from the phone or egress path.

Is the problem you're troubleshooting have anything to do with one-way or 
no-way audio by chance?

Thanks,

Ryan

On Jul 21, 2017, at 3:37 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, Guys.
I need to test

Re: [cisco-voip] Force an outgoing call through a subscriber

[ROZA, Ariel]

Hi, Ryan, Anthony, et al,

  I´m sorry by the blunt nature of my first mail. I was burned out, at my 
customer´s.

  My problem is this:
  The users are complaining of broken audio in random calls to the PSTN

  We Isolated the issue to a single subscriber that acts as MTP for calls, 
sometimes.
Wireshark captures show us that whenever the sub acts as an intermediary, audio 
gets jitter (but no drops) This is on the upstream to the PSTN. The downstream 
is always fine.
We suspect of a problem in the underlying nework, as the Publisher and the Subs 
are wired differently and only the Subscriber show these problems.
The Publisher is connected to a CAT4500 switch and the Sub is connected to a 
NEXUS 5000  that in turn is connected to the Cat4500

I am already using MRGL/MRGs to  force MTP, but sometimes the phones send the 
RTP directly to the gateway, bypassing the MTP. (calling always the same number)


De: Ryan Huff [mailto:ryanh...@outlook.com]
Enviado el: domingo, 23 de julio de 2017 07:03 p.m.
Para: Anthony Holloway <avholloway+cisco-v...@gmail.com>
CC: ROZA, Ariel <ariel.r...@la.logicalis.com>; cisco-voip 
<cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Force an outgoing call through a subscriber

Anthony,

Yes, it is splitting hairs IMO :). I'm specifically talking about a typical 
scenario where RTP is sent from/to the phone for a connected, like-codec call 
that is not forcing media termination with the next device in the call leg.

As you and I have mentioned, there are a handful of ways to use different 
features and services of a CUCM server to terminate and join media streams 
(MTP, CFB ... etc) ... and I considers these as ancillary service and component 
capabilities to the server and not a core function of the server itself (hence, 
media not flowing through the server, although it maybe interacting with one or 
more of the aforementioned media services or components).

Always happy to split hairs :).

Sent from my iPhone

On Jul 23, 2017, at 5:18 PM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
This may be splitting hair here, but two things Ryan:

1) Your first sentence reads to me like a contradiction.  Could you clarify 
what you're stating here?

2) Media can flow through, and even terminate on CUCM, since things like 
Conference Bridges, MTPs and now the new IVR media resource, are all doing that.

On Fri, Jul 21, 2017 at 2:53 PM Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
So the actual media (RTP) will never flow through a CUCM server; it may 
however, terminate a connected media stream on a software based MTP application 
that CUCM runs as a service (IP Voice Media Streaming Application). Signaling 
(SIP) on the other hand, will always traverse a CUCM server.

If you see a CUCM IP address in the Audio field of the SDP, then it's likely 
terminating on a CUCM based MTP resource (most often, due to some differences 
in DTMF negotiations or because the egress path in CUCM is required to use MTP).

If you are trying to test a call using a CUCM MTP resource on a particular 
cluster node; the simplest way would be to create a new MRG/MRGL that only 
specifies MTP resources from the desired cluster node and then advertise that 
MRGL to the phone and/or egress path to the pstn for the phone and then 
"require" MTP termination from the phone or egress path.

Is the problem you're troubleshooting have anything to do with one-way or 
no-way audio by chance?

Thanks,

Ryan

On Jul 21, 2017, at 3:37 PM, ROZA, Ariel 
<ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, Guys.
I need to test problems with calls outgoing from an Ip phone to the PSTN  
through a particular subscriber (as MTP?).
How can I force them to do that.
Packet captures show me that, at times, calls go from my phone to the h323 
gateway and sometimes they go from my phone to the Sub and then to the gew.
Obtener Outlook para 
Android<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Faka.ms%2Fghei36=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C1e58758c6884429f5d0d08d4d2169e17%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636364441966497710=0AscsFwizg82xmGqihl67oG7G1Czd3U8w5dCXgStsqM%3D=0>

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Re: [cisco-voip] Random DND?

[ROZA, Ariel]

In the Jabber client you can set the DND status in two situations:

-  After XX minutes

-  When the PC user session is locked (CRTL-ALT-DEL...)
I am looking at my jabber config, and they are enabled by default.

De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Lelio 
Fulgenzi
Enviado el: jueves, 06 de julio de 2017 11:14 a.m.
Para: Ben Amick ; cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Random DND?
bber

I think you hit the nail on the head - I was surprised to see DND working on 
the phone via Jabber. Without softkeys, you need to unset it with Jabber as 
well. Is it possible for Jabber to set the user to Away after a period of time 
even with it set to DND?

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ben 
Amick
Sent: Thursday, July 06, 2017 8:39 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Random DND?

Hey Guys,
Randomly have this issue, select users will report being in a DND status, in 
some cases with users that do not even have the DND feature configured in their 
softkey templates. I believe this may be connected to users putting themselves 
in DND via Jabber, but upon checking, their Jabber statuses are set to 
available or away. Anyone know what could be triggering this?

Confidentiality Note: This message is intended for use only by the individual 
or entity to which it is addressed and may contain information that is 
privileged, confidential, and exempt from disclosure under applicable law. If 
the reader of this message is not the intended recipient or the employee or 
agent responsible for delivering the message to the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
communication is strictly prohibited. If you have received this communication 
in error, please contact the sender immediately and destroy the material in its 
entirety, whether electronic or hard copy. Thank you
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Re: [cisco-voip] UCCX issue - Finesse agents unable to login

[ROZA, Ariel]

Looks like Finesse can´t contact Active Directory to log in the Agent and the 
connection Times out. The reason the Connection Manager gives is 
ADOutOfServiceMsg


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de 
Tooze-Froggatt, Peter (ISS)
Enviado el: lunes, 03 de julio de 2017 07:43 a.m.
Para: cisco-voip@puck.nether.net
Asunto: [cisco-voip] UCCX issue - Finesse agents unable to les ogin

Hi all,

Running 11.5.1.1-61 (ES01-45)

Thought I would find out if anyone has come across the following issue:

We had an outage on our Service Desk whereby users were unable to login and 
calls couldn't be answered. Looking at the MIVR logs, calls were coming in and 
being held on a CTI port until an agent was made available. You can see agents 
('X' and 'Y') attempting to login, but shortly afterwards, the following 
exceptions are thrown:

13971851: Jun 28 08:56:59.266 BST 
%MIVR-ICD_CTI-6-WATCHDOG_INFO:Watchdogthread:: A specific description for a 
trace=Watchdog thread - socket ClientId: 
FinesseSocket[addr=127.0.0.1,port=50556,localport=12028] timed out or stuck so 
closed.  q size is :0 time before last sent msg is : 1498636615424 time after 
last sent msg 1498636615424current time is : 1498636619266timeout is : 
12stuck time is : 3842time diff is : 124866
13971852: Jun 28 08:56:59.267 BST %MIVR-ICD_CTI-3-CLIENT_SOCKET_CLOSED:Client 
connection closed: A specific description for a trace=ClientConnMgr before 
closing Socket:socket is Socket[addr=127.0.0.1,port=50556,localport=12028] for 
agentID:X
13971853: Jun 28 08:56:59.267 BST %MIVR-ICD_CTI-3-CLIENT_SOCKET_CLOSED:Client 
connection closed: A specific description for a trace=ClientConnMgr posted 
ADOutOfServiceMsg for agentID:X, agentDN: 93903
13971854: Jun 28 08:56:59.268 BST %MIVR-ICD_CTI-3-CLIENT_SOCKET_CLOSED:Client 
connection closed: A specific description for a trace=ClientConnMgr before 
closing Socket:socket is Socket[addr=127.0.0.1,port=50556,localport=12028] for 
agentID:Y
13971855: Jun 28 08:56:59.268 BST %MIVR-ICD_CTI-3-CLIENT_SOCKET_CLOSED:Client 
connection closed: A specific description for a trace=ClientConnMgr posted 
ADOutOfServiceMsg for agentID:Y, agentDN: 93825
13971856: Jun 28 08:56:59.268 BST %MIVR-SS_RM-7-UNK:Processing msg: 
ADOutOfServiceMsg (Rsrc:X)
13971857: Jun 28 08:56:59.269 BST 
%MIVR-ICD_CTI-3-SOCKET_READ_EXCEPTION:Exception while reading from a socket 
input.: A description for the nature of the exception=ClientConnMgr: 
SocketException reading msg from input stream of socket : 
Socket[addr=127.0.0.1,port=50556,localport=12028] in client thread 
MIVR_ICD_CTI_client_thread_8775-19121-,Exception=java.net.SocketException: 
Socket closed
13971858: Jun 28 08:56:59.269 BST 
%MIVR-ICD_CTI-3-EXCEPTION:java.net.SocketException: Socket closed
13971859: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
java.net.SocketInputStream.socketRead0(Native Method)
13971860: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
java.net.SocketInputStream.read(SocketInputStream.java:152)
13971861: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
java.net.SocketInputStream.read(SocketInputStream.java:122)
13971862: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
java.io.DataInputStream.readFully(DataInputStream.java:195)
13971863: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
java.io.DataInputStream.readFully(DataInputStream.java:169)
13971864: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.Ged188.io.CRACTIInputStream.readMSG(CRACTIInputStream.java:570)
13971865: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.wf.subsystems.ctiserver.ClientConn.run(ClientConn.java:71)
13971866: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.executor.Thread.run(Thread.java:539)
13971867: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.executor.impl.ExecutorStubImpl$RequestImpl.runCommand(ExecutorStubImpl.java:690)
13971868: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.executor.impl.ExecutorStubImpl$RequestImpl.run(ExecutorStubImpl.java:486)
13971869: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.executor.impl.ExecutorStubImpl$RequestImpl.run(ExecutorStubImpl.java:762)
13971870: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
EDU.oswego.cs.dl.util.concurrent.PooledExecutor$Worker.run(PooledExecutor.java:776)
13971871: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.executor.impl.PooledExecutorStubImpl$1$WorkerImpl.run(PooledExecutorStubImpl.java:99)
13971872: Jun 28 08:56:59.269 BST %MIVR-ICD_CTI-3-EXCEPTION:at 
com.cisco.util.ThreadPoolFactory$ThreadImpl.run(ThreadPoolFactory.java:853)

The user isn't able to login at all, and in fact see the same behaviour as 
described in this port:

Re: [cisco-voip] webex design or deployment guide?

[ROZA, Ariel]

For Webex Meetings Servers there´s the Planning guide

http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_8/Planning_Guide/cwms_b_planning-guide-system-requirements-cwms-2-8.html


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Lelio 
Fulgenzi
Enviado el: miércoles, 19 de abril de 2017 06:04 p.m.
Para: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Asunto: [cisco-voip] webex design or deployment guide?


For those of you using WebEx, have you found any sort of documentation with 
respect to designing/deploying WebEx?

I know there not as much to design as there is with something like CallManager, 
but I'm looking for something that describes what the features are rather than 
just how to enable or disable them like the admin guide shows.

Best practices?

Just nothing out there.

I'd rather not deploy to a thousand people only to have to change things 
because of some issues that could have been averted by decided appropriately 
whether to enabled or disable a certain feature.

Lelio


---
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Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
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www.uoguelph.ca/ccs
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[cisco-voip] Garbage in Call Logs on 2921router

[ROZA, Ariel]

Hi, guys.

While troubleshooting a call issue in a 2921 router, i found this message 
between the logs

Apr  5 12:47:39.885: //-1/E33315F498E1/CCAPI/cc_api_call_setup_ind_common:
   Sub Calling(TON=Unknown, NPI=Unknown, Data=vD=^E" 

Re: [cisco-voip] UCCX 8.5(1)SU2 Restore Woes

[ROZA, Ariel]

The warning message states: Have you tried to locate and download the 
online.log file?

WARNING: server initialization failed, or possibly timed out (if -w was used).
Check the message log, online.log, for errors.]

The warning message states: Have you tried to locate and download the 
online.log file to see what´s in it?




De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Nathan 
Reeves
Enviado el: martes, 14 de marzo de 2017 07:32 a.m.
Para: cisco-voip@puck.nether.net
Asunto: [cisco-voip] UCCX 8.5(1)SU2 Restore Woes

THrowing this out in case someone's seen anything similar.  Will be lodging a 
TAC case in the morning.

Have a UCCX HA Cluster Setup running 8.5(1)SU2.  Currently trying to restore 
the backup into an isolated network but keep hitting an issue with the 
CCXCOMPONENT restore failing at 25%.

Running the COP02 and COP03 patches on both boxes (to get round the LDAP 
Backup/Restore defect) and have a good backup from the cluster.

Restore Steps:
 - Have a restored copy of the CUCM Cluster up and running ok.
 - Built the CCX PUB node as 8.5(1)SU2 and did the initial setup using temp 
username etc for the CTI and RMCM Users.
 - Installed COP02 and rebooted, followed by COP03 and a restart.
 - Attempted to run the restore of just the PUB.  All components restore 
successfully until we get to CCXCOMPONENT.  This fails at 25% with the error 
32-Failed to start UCCX DB Service.

Log file extract is below.

Anyone seen similar?  Can't see anything in Bug DB.  Restore notes mention that 
if you're on SU2 to complete the intial setup using temp users, install COP 
files and then restore, so I think I'm doing this right.




= Log file extract 
==

2017/03/14 18:18: Calling do_restore script of DB component
2017/03/14 18:18: Executing cmd: /opt/cisco/uccx/drf/backup/db_do_restore.py 
/common/drf/restore.log /common/drf/status.txt
2017-03-14 18:18:47,888 - uccxdb.bnr - DEBUG -  Begin Unified CCX 
DB DoRestore(db_do_restore.py) 
2017-03-14 18:18:47,892 - uccxdb.bnr - DEBUG - Executing 
[/usr/local/cm/bin/controlcenter.sh list | grep "Cisco Unified CCX Database" ] 
...
2017-03-14 18:18:53,360 - uccxdb.bnr - DEBUG - Results of executing 
[/usr/local/cm/bin/controlcenter.sh list | grep "Cisco Unified CCX Database" ], 
returnCode :[0], Output :[log4j:ERROR Could not find value for key 
log4j.appender.custom
log4j:ERROR Could not instantiate appender named "custom".
Cisco Unified CCX Database[STARTING] ]
2017-03-14 18:18:53,360 - uccxdb.bnr - DEBUG - Executing [cat 
/common/drf/ccx_comps/uccx_db/uccx_version.txt] ...
2017-03-14 18:18:53,367 - uccxdb.bnr - DEBUG - Backup version of 
UCCX:8.5.1.11002-22
2017-03-14 18:18:53,476 - uccxdb.bnr - DEBUG - Restore version of 
UCCX:8.5.1.11002-22
2017-03-14 18:18:53,476 - uccxdb.bnr - DEBUG - UCCX Backup and Restore versions 
are same
2017-03-14 18:18:53,476 - uccxdb.bnr - DEBUG - Copying DB configuration files 
from backup
2017-03-14 18:18:53,477 - uccxdb.bnr - DEBUG - Copying 
/common/drf/ccx_comps/uccx_db/onconfig.uccx
2017-03-14 18:18:53,477 - uccxdb.bnr - DEBUG - Copying 
/common/drf/ccx_comps/uccx_db/sqlhosts.uccx
2017-03-14 18:18:53,478 - uccxdb.bnr - DEBUG - Copying 
/common/drf/ccx_comps/uccx_db/uccx_ids.env
2017-03-14 18:18:53,478 - uccxdb.bnr - DEBUG - Copying 
/common/drf/ccx_comps/uccx_db/services
2017-03-14 18:18:53,479 - uccxdb.bnr - DEBUG - Executing 
[/usr/local/bin/base_scripts/xmlfoo /usr/local/platform/conf/platformConfig.xml 
CcmFirstNode ParamValue] ...
2017-03-14 18:18:53,488 - uccxdb.bnr - DEBUG - Results of executing 
[/usr/local/bin/base_scripts/xmlfoo /usr/local/platform/conf/platformConfig.xml 
CcmFirstNode ParamValue], returnCode :[0], Output :[yes]
2017-03-14 18:18:53,488 - uccxdb.bnr - DEBUG - UCCX DB Ontape Backup is present 
in this node
2017-03-14 18:18:53,488 - uccxdb.bnr - DEBUG - Initiating UCCX DB Restore 
operation...
2017-03-14 18:18:53,488 - uccxdb.bnr - DEBUG - Executing [/bin/chmod 664 
/common/drf/ccx_comps/uccx_db/ontape_uccx_backup.gz] ...
2017-03-14 18:18:53,492 - uccxdb.bnr - DEBUG - Executing [source 
/opt/cisco/uccx/bin/uccx_ids.env ; onstat -] ...
2017-03-14 18:18:53,522 - uccxdb.bnr - DEBUG - Results of executing [source 
/opt/cisco/uccx/bin/uccx_ids.env ; onstat -], returnCode :[65280], Output 
:[shared memory not initialized for INFORMIXSERVER 'ccxprd01_uccx']
2017-03-14 18:18:53,522 - uccxdb.bnr - DEBUG - Executing [source 
/opt/cisco/uccx/bin/uccx_ids.env ; onmode -yuk] ...
2017-03-14 18:18:53,553 - uccxdb.bnr - DEBUG - Results of executing [source 
/opt/cisco/uccx/bin/uccx_ids.env ; onmode -yuk], returnCode :[0], Output 
:[shared memory not initialized for INFORMIXSERVER 'ccxprd01_uccx']
2017-03-14 18:18:53,554 - uccxdb.bnr - DEBUG - Unified CCX DB is not up and 
running(not On-Line). Trying to start...
2017-03-14 18:18:53,554 - uccxdb.bnr - DEBUG - Executing [source 

Re: [cisco-voip] CUBE call authorization ?

[ROZA, Ariel]

Carlos,

Maybe something like this?

Inbound Leg Headers for Outbound Dial-Peer Matching
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/inbound-hdr-for-outbound.html#concept_783172607421874995245657348632812


De: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] En nombre de Anthony 
Holloway
Enviado el: viernes, 10 de marzo de 2017 03:29 p.m.
Para: Carlos Mendioroz ; cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] CUBE call authorization ?

Fair enough.  I didn't understand what you were originally asking.  Good luck.

On Fri, Mar 10, 2017 at 9:52 AM Carlos Mendioroz 
> wrote:
AFAIK, trusted list is just to enable signalling, but I want to
differentiate what can each neighbour do.
I need to talk to each SP, but I don't want one SP to be able to call
via another...

Anthony Holloway @ 10/03/2017 10:44 -0400 dixit:
> Maybe I'm missing what you're looking for, but why not just use the
> built in ip address trusted list?
>
> https://supportforums.cisco.com/document/46566/understanding-toll-fraud-enhancements-1512t
>
> On Thu, Mar 9, 2017 at 7:16 AM Carlos Mendioroz via cisco-voip
>  
> >> wrote:
>
> Hi,
> I'm trying to migrate a CME install to a CUCM, and was thinking of doing
> it gradually. In the end, the CME should be left as a CUBE, terminating
> the ITSPs trunks.
>
> Now, I do want to have some sort of call authorization just to be on the
> safe side, and not discovering that one SP ended up making calls accross
> my GW. Thought of COR lists, but I found no easy way
> to link a dial peer to an incoming call from a given SIP trunk.
>
> I'm currently using a prefix as a enablement "secret", but there has to
> be a better way. I'm embarrased to admit I don't see it.
> Help ?
>
> --
> Carlos G Mendioroz   
> >>
> LW7 EQI  Argentina
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net 
> >
> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

--
Carlos G Mendioroz  >  LW7 EQI  
Argentina
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Re: [cisco-voip] Errors 18 and 19 on incoming call only through subscriber

[ROZA, Ariel]

I think I may be hitting this bug:

https://tools.cisco.com/bugsearch/bug/CSCtr83193

It applies to Route Lists instead of Hunt Lists, but the symptoms look similar, 
and the CUCM version matches..

What do you think?

Regards,
Ariel Roza
Service Delivery Consultant  / Argentina
Logicalis
Tel: +54 11 5282 - 0458
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com
Peru 327 – CABA – Argentina – C1067AAG
www.la.logicalis.comhttp://www.la.logicalis.com/
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From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of ROZA, 
Ariel
Sent: jueves, 05 de marzo de 2015 06:56 p.m.
To: Ryan Ratliff (rratliff); Brian Meade
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

Hi, Guys…

I restarted RIS on both nodes and did not do a thing, so I guess Ryan might be 
right. What´s the alternative? Restarting the Callmanager Service? Rebooting 
the whole sub? Both Sub and Pub?

I have h225, h245 and full ccm traces if there´s any use for them.

Regards,

Ariel.

From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: miércoles, 11 de febrero de 2015 11:23 a.m.
To: Brian Meade
Cc: ROZA, Ariel; cisco-voip voyp list
Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

RIS shouldn’t be involved in that flow, though it does read from the same 
shared memory segment that the ccm process does so unless RIS itself is screwed 
I’d expect if ccm doesn’t know a remote phone’s registration status RIS on that 
node won’t either.

-Ryan

On Feb 10, 2015, at 5:19 PM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Right, could be an SDL link down.  I've seen this caused by RIS issues where 
the node receiving the call doesn't think the phone is registered (can't find 
the LineControl process).  Restarting RISDC usually fixes it.

On Tue, Feb 10, 2015 at 4:26 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
How would this be RIS?

More like an SDL connection that’s down, or leaked a call on the sub.  Register 
a phone to the sub and see if you can call to/from another phone registered to 
the pub.

-Ryan

On Feb 10, 2015, at 10:32 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Usually this is a RISDC problem.  Check database replication is working okay 
and then restart RISDC on all nodes.

On Tue, Feb 10, 2015 at 9:06 AM, ROZA, Ariel 
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com wrote:
 Guys,

I am having and odd situation:

I have a CUCM cluster ( Pub  1 Sub) where i have set up a new branch office 
with an H323 Gateway and several phones. The config template is similar to 
other already working branches, but in this particular branch incoming calls 
are behaving strangely.
PSTN calls come through a FXO port, pass an autoattendant TCL script  on the GW 
and get routed to CUCM via two identical dialpeers ¡, one for the Sub and one 
for the Pub.
Debugs show that if calls use the Dialpeer to the Pub, the call goes fine, and 
the receiver phone answers the call. But if the call uses the Subs dialpeer, it 
fails with a Disconnect Cause code 18 (user busy) or 19 (user busy - user 
alerted).

The destination in both cases is the same: a Hunt List that is reachable via 
any internal phone.

DNA shows the logic is ok, and debugs on both, the gateway and CUCM show the 
same results, so conectivity should not be an issue.

 Does anyone have any suggestion, besides rebooting the Subs?

CUCM version is 8.6.2
The Pub and the Subs are in different subnets.

Regards,

Ariel.

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Re: [cisco-voip] Errors 18 and 19 on incoming call only through subscriber

[ROZA, Ariel]

Hi, Guys…

I restarted RIS on both nodes and did not do a thing, so I guess Ryan might be 
right. What´s the alternative? Restarting the Callmanager Service? Rebooting 
the whole sub? Both Sub and Pub?

I have h225, h245 and full ccm traces if there´s any use for them.

Regards,

Ariel.

From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: miércoles, 11 de febrero de 2015 11:23 a.m.
To: Brian Meade
Cc: ROZA, Ariel; cisco-voip voyp list
Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

RIS shouldn’t be involved in that flow, though it does read from the same 
shared memory segment that the ccm process does so unless RIS itself is screwed 
I’d expect if ccm doesn’t know a remote phone’s registration status RIS on that 
node won’t either.

-Ryan

On Feb 10, 2015, at 5:19 PM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Right, could be an SDL link down.  I've seen this caused by RIS issues where 
the node receiving the call doesn't think the phone is registered (can't find 
the LineControl process).  Restarting RISDC usually fixes it.

On Tue, Feb 10, 2015 at 4:26 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
How would this be RIS?

More like an SDL connection that’s down, or leaked a call on the sub.  Register 
a phone to the sub and see if you can call to/from another phone registered to 
the pub.

-Ryan

On Feb 10, 2015, at 10:32 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Usually this is a RISDC problem.  Check database replication is working okay 
and then restart RISDC on all nodes.

On Tue, Feb 10, 2015 at 9:06 AM, ROZA, Ariel 
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com wrote:
 Guys,

I am having and odd situation:

I have a CUCM cluster ( Pub  1 Sub) where i have set up a new branch office 
with an H323 Gateway and several phones. The config template is similar to 
other already working branches, but in this particular branch incoming calls 
are behaving strangely.
PSTN calls come through a FXO port, pass an autoattendant TCL script  on the GW 
and get routed to CUCM via two identical dialpeers ¡, one for the Sub and one 
for the Pub.
Debugs show that if calls use the Dialpeer to the Pub, the call goes fine, and 
the receiver phone answers the call. But if the call uses the Subs dialpeer, it 
fails with a Disconnect Cause code 18 (user busy) or 19 (user busy - user 
alerted).

The destination in both cases is the same: a Hunt List that is reachable via 
any internal phone.

DNA shows the logic is ok, and debugs on both, the gateway and CUCM show the 
same results, so conectivity should not be an issue.

 Does anyone have any suggestion, besides rebooting the Subs?

CUCM version is 8.6.2
The Pub and the Subs are in different subnets.

Regards,

Ariel.

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Re: [cisco-voip] Errors 18 and 19 on incoming call only through subscriber

[ROZA, Ariel]

Thanks. So, the short term workaround would be to restart the CCM Service, 
until I can apply the latest service release. Right?

From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: jueves, 05 de marzo de 2015 07:15 p.m.
To: ROZA, Ariel
Cc: Ryan Ratliff (rratliff); cisco-voip voyp list
Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

Hunt Lists use the RouteListControl process as well so definitely could be the 
issue.

On Thu, Mar 5, 2015 at 5:13 PM, ROZA, Ariel 
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com wrote:
I think I may be hitting this bug:

https://tools.cisco.com/bugsearch/bug/CSCtr83193

It applies to Route Lists instead of Hunt Lists, but the symptoms look similar, 
and the CUCM version matches..

What do you think?

Regards,
Ariel Roza
Service Delivery Consultant  / Argentina
Logicalis
Tel: +54 11 5282 - 0458tel:%2B54%2011%205282%20-%200458
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com
Peru 327 – CABA – Argentina – C1067AAG
www.la.logicalis.comhttp://www.la.logicalis.com/
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From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.netmailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of ROZA, Ariel
Sent: jueves, 05 de marzo de 2015 06:56 p.m.
To: Ryan Ratliff (rratliff); Brian Meade
Cc: cisco-voip voyp list

Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

Hi, Guys…

I restarted RIS on both nodes and did not do a thing, so I guess Ryan might be 
right. What´s the alternative? Restarting the Callmanager Service? Rebooting 
the whole sub? Both Sub and Pub?

I have h225, h245 and full ccm traces if there´s any use for them.

Regards,

Ariel.

From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: miércoles, 11 de febrero de 2015 11:23 a.m.
To: Brian Meade
Cc: ROZA, Ariel; cisco-voip voyp list
Subject: Re: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

RIS shouldn’t be involved in that flow, though it does read from the same 
shared memory segment that the ccm process does so unless RIS itself is screwed 
I’d expect if ccm doesn’t know a remote phone’s registration status RIS on that 
node won’t either.

-Ryan

On Feb 10, 2015, at 5:19 PM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Right, could be an SDL link down.  I've seen this caused by RIS issues where 
the node receiving the call doesn't think the phone is registered (can't find 
the LineControl process).  Restarting RISDC usually fixes it.

On Tue, Feb 10, 2015 at 4:26 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
How would this be RIS?

More like an SDL connection that’s down, or leaked a call on the sub.  Register 
a phone to the sub and see if you can call to/from another phone registered to 
the pub.

-Ryan

On Feb 10, 2015, at 10:32 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Usually this is a RISDC problem.  Check database replication is working okay 
and then restart RISDC on all nodes.

On Tue, Feb 10, 2015 at 9:06 AM, ROZA, Ariel 
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com wrote:
 Guys,

I am having and odd situation:

I have a CUCM cluster ( Pub  1 Sub) where i have set up a new branch office 
with an H323 Gateway and several phones. The config template is similar to 
other already working branches, but in this particular branch incoming calls 
are behaving strangely.
PSTN calls come through a FXO port, pass an autoattendant TCL script  on the GW 
and get routed to CUCM via two identical dialpeers ¡, one for the Sub and one 
for the Pub.
Debugs show that if calls use the Dialpeer to the Pub, the call goes fine, and 
the receiver phone answers the call. But if the call uses the Subs dialpeer, it 
fails with a Disconnect Cause code 18 (user busy) or 19 (user busy - user 
alerted).

The destination in both cases is the same: a Hunt List that is reachable via 
any internal phone.

DNA shows the logic is ok, and debugs on both, the gateway and CUCM show the 
same results, so conectivity should not be an issue.

 Does anyone have any suggestion, besides rebooting the Subs?

CUCM version

[cisco-voip] Errors 18 and 19 on incoming call only through subscriber

[ROZA, Ariel]

Hi, Robert

When the Gateways goes into SRST mode, IP Phones registered to it are able to 
call other phones with no issues.
The IP Phones are registered to the Subscriber
The Gateway´s Device Pool has the same CMG that every other phone in the 
company (The Subscriber is the first choice)
Database Replicaton is Ok. (Sub and Pub are at “2”). I already did a restart of 
the RIS Database service on both Sub  Pub.

I´ll gather the debugs  traces if can´t find another solution. (Don´t want to 
restart the servers…)

From: Schuknecht Robert [mailto:robert.schukne...@logicalis.de]
Sent: martes, 10 de febrero de 2015 12:58 p.m.
To: ROZA, Ariel
Subject: AW: [cisco-voip] Errors 18 and 19 on incoming call only through 
subscriber

Hi Ariel,

what happens when you call a IP-Phone directly from the Gateway? On which CUCM 
are the IP-Phones registered to? Does the gateway have an Device-Pool with an 
CUCM Group consisting of both CUCMs? Could you provide detailed CUCM Traces and 
outputs from debug h225 asn1 and debug h245 asn1? Is the DB-Replication working?

Kind Regards,

Robert



Robert Schuknecht
CCIE VOICE # 32360
System Engineer
Logicalis GmbH

Tel:  +49-2234-95418-746tel:+49-2234-95418-746
Fax:  +49-2234-95418-779
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responsibility to scan the email and attachments (if any).
Von: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] Im Auftrag von 
ROZA, Ariel
Gesendet: Dienstag, 10. Februar 2015 15:06
An: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Betreff: [cisco-voip] Errors 18 and 19 on incoming call only through subscriber


 Guys,



I am having and odd situation:



I have a CUCM cluster ( Pub  1 Sub) where i have set up a new branch office 
with an H323 Gateway and several phones. The config template is similar to 
other already working branches, but in this particular branch incoming calls 
are behaving strangely.

PSTN calls come through a FXO port, pass an autoattendant TCL script  on the GW 
and get routed to CUCM via two identical dialpeers ¡, one for the Sub and one 
for the Pub.

Debugs show that if calls use the Dialpeer to the Pub, the call goes fine, and 
the receiver phone answers the call. But if the call uses the Subs dialpeer, it 
fails with a Disconnect Cause code 18 (user busy) or 19 (user busy - user 
alerted).



The destination in both cases is the same: a Hunt List that is reachable via 
any internal phone.



DNA shows the logic is ok, and debugs on both, the gateway and CUCM show the 
same results, so conectivity should not be an issue.



 Does anyone have any suggestion, besides rebooting the Subs?



CUCM version is 8.6.2

The Pub and the Subs are in different subnets.



Regards,



Ariel.
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[cisco-voip] Errors 18 and 19 on incoming call only through subscriber

[ROZA, Ariel]

 Guys,



I am having and odd situation:



I have a CUCM cluster ( Pub  1 Sub) where i have set up a new branch office 
with an H323 Gateway and several phones. The config template is similar to 
other already working branches, but in this particular branch incoming calls 
are behaving strangely.

PSTN calls come through a FXO port, pass an autoattendant TCL script  on the GW 
and get routed to CUCM via two identical dialpeers ¡, one for the Sub and one 
for the Pub.

Debugs show that if calls use the Dialpeer to the Pub, the call goes fine, and 
the receiver phone answers the call. But if the call uses the Subs dialpeer, it 
fails with a Disconnect Cause code 18 (user busy) or 19 (user busy - user 
alerted).



The destination in both cases is the same: a Hunt List that is reachable via 
any internal phone.



DNA shows the logic is ok, and debugs on both, the gateway and CUCM show the 
same results, so conectivity should not be an issue.



 Does anyone have any suggestion, besides rebooting the Subs?



CUCM version is 8.6.2

The Pub and the Subs are in different subnets.



Regards,



Ariel.
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[cisco-voip] UCCX: Contact is inactive when getting channel

[ROZA, Ariel]

Hi, Guys.



I am trying to set up a UCCX script to receive a call and forward it to a CSQ. 
After the conversation with the Agent ends, the user must listen to a Customer 
Satisfactions survey.



I think I must have something conceptually wrong, because it does not seem to 
work as I expected.



I wrote the whole script and tested it without any agents logged into the CSQ 
and it works fine. But whenever an agent is logged in and picks up the call, 
the script ends without playing the survey.

Whenever I run a Debug on the Script, I get the attached error message.



Are the calls forwarded to an agent supposed to be final, or you CAN do extrea 
work with the call afterwards?



Im attaching a copy of the script in case it helps.

The script is just a demo I am working on, and not a full-fledged script. Also, 
it´s partially in spanish because I am too lazy to rewrite it :) The relevant 
section is after the label :ENCUESTA (:Poll)



Thanks!



ARIEL.




Script de Demo 1.aef
Description: Script de Demo 1.aef
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[cisco-voip] Escape charaters in UCCX scripts

[ROZA, Ariel]

Guys,

I am trying to work with a UCCX script, but i don´t seem to get the hang of 
using the escape characters. At least does not seem to behave like a normal 
programming language.

This is the scenario:

I am appending some info to a text file, like a common .CSV file.

I would like to achieve this:

Each line is an append from my script (a very simple test script)

2000;25427462;5;4
2000;25427462;5;4
2000;25427462;5;4
2000;25427462;5;4

But instead, I am Getting this:


2000;25427462;5;42000;25427462;5;42000;25427462;5;42000;25427462;5;4


And when I try to use the escape carácter \n, I get this:



2000;25427462;5;4\n2000;25427462;5;4\n2000;25427462;5;4\n2000;25427462;5;4


The lines I use to concatenate the file is this:

Set dFile = DOC[demo\encuesta.txt] + \n + TEXT[sCallingNumber+;+ sDNI +;+ 
sPregunta1 +; + sPregunta2]
Set sFilePath = demo\encuesta.txt
Upload Document (dFile to L[], sFilePath)

So far, I tried using \n without the quotes, but it gives me an error. Also 
'\n' seems to do the same as the double quotes...
How can I make it to insert a newline?

Regards,

Ariel Roza
Field Engineer  / Argentina
Logicalis
Tel: +54 11 4344 - 0458
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com
Peru 327 - CABA - Argentina - C1067AAG
www.la.logicalis.comhttp://www.la.logicalis.com/
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[cisco-voip] Cisco UCCX Editor on a restricted enviroment

[ROZA, Ariel]

Hi,

Has anyone tried and to and been successful in running Cisco UCCX Editor 
without administrative rights?
I know that the docs states exactly that requirement, but I am stuck inside a 
customer´s  very restrictive environment and need to get this to work.

I can´t do the following things:
No user can have local Admin Rights
Can´t connect my own laptop to the network
Can´t copy files out to a pendrive

So, I can´t copy the Editor´s install file to my own laptop to work offline.
Can anyone, in a gesture of grandness :), put a copy of the UCCX Editor 9.0(2) 
up in a Dropbox accout or similar and send me a link?
This will be MUCH appreciated

Regards, and thanks in advance
Ariel Roza
Field Engineer / Southern Cone
Logicalis
Tel: +54 11 4344-0458
ariel.r...@la.logicalis.commailto:ariel.r...@la.logicalis.com
Peru 327 - CABA - Argentina - C1067AAG
www.la.logicalis.comhttp://www.la.logicalis.com/
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Re: [cisco-voip] Digit Delay on 7 digit dialing

[ROZA, Ariel]

Matt,

The logs show no events earlier than 10:37:45… so I cannot see your “good” call 
in it…

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: martes, 13 de mayo de 2014 01:29 p.m.
To: Brian Meade
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Digit Delay on 7 digit dialing

Sure. Calls are from endpoint ip’d 1.8.36.13 to 95843432 at 10:38:02.114 
(Delayed call), No dialed number until 10:38:09.628 and Good call is to 
95320240 at 10:37:41.586, dialed number immediately at 10:37:41.587


Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: Tuesday, May 13, 2014 11:36 AM
To: Matthew Loraditch
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Digit Delay on 7 digit dialing

Can you post the traces?

Brian

On Tue, May 13, 2014 at 11:03 AM, Matthew Loraditch 
mloradi...@heliontechnologies.commailto:mloradi...@heliontechnologies.com 
wrote:
Why would I have a 7 second delay on 1 call being dialed (access code +)  7 
digits and no delay (1millisecond) on another? Same phone, same CSS, same 
route pattern, same results in DNA.
I’m reading the traces in translatorx, but having a hard time finding a reason.

Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518tel:443.541.1518
fax.  410.252.9284tel:410.252.9284

Twitterhttp://twitter.com/heliontech  |  
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Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830tel:410.252.8830



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Re: [cisco-voip] Digit Delay on 7 digit dialing

[ROZA, Ariel]

No, Matt. Sorry. My knowledge does not include understanding CUCM logs :P

From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com]
Sent: martes, 13 de mayo de 2014 01:24 p.m.
To: ROZA, Ariel; cisco-voip@puck.nether.net
Subject: RE: Digit Delay on 7 digit dialing

Yes 1 generic 7 digit pattern.
No matching translation patterns.

The logs in translatorx for the immediate call look exactly the same as the 
delayed one, except with the delay.

Reading the raw logs. I see an immediate digit analysis match. I'm trying to 
read through to find what is taking the time, but the system is very busy. Lots 
of events in the delay interval.
Anything specific to look for?


Matthew G. Loraditch - CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


From: ROZA, Ariel [mailto:ariel.r...@la.logicalis.com]
Sent: Tuesday, May 13, 2014 11:31 AM
To: Matthew Loraditch; cisco-voip@puck.nether.net
Subject: RE: Digit Delay on 7 digit dialing

If you have only one 7-digit pattern to handle all the 7-digit calls, you are 
right. If you have more, that changes things.
Any similar translation patterns?
Looks like CUCM is spending more time doing digit analisys.

Are the logs for the immediate call of the same length than those of the 
delayed call?


From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com]
Sent: martes, 13 de mayo de 2014 12:25 p.m.
To: ROZA, Ariel; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: RE: Digit Delay on 7 digit dialing

There is a 10 digit local pattern as well, but if that was the issue, shouldn't 
it happen for all 7 digit calls?
The delay is between the last Keypad Button and the Dialed Number state in the 
logs. I believe that isolates the issue to the CUCM side.


Matthew G. Loraditch - CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


From: ROZA, Ariel [mailto:ariel.r...@la.logicalis.com]
Sent: Tuesday, May 13, 2014 11:20 AM
To: Matthew Loraditch; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: RE: Digit Delay on 7 digit dialing

An existing similar pattern with more digits?
A delay at the gateway?
A delay on the receiving side (PSTN or PBX)?

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: martes, 13 de mayo de 2014 12:04 p.m.
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Digit Delay on 7 digit dialing

Why would I have a 7 second delay on 1 call being dialed (access code +)  7 
digits and no delay (1millisecond) on another? Same phone, same CSS, same 
route pattern, same results in DNA.
I'm reading the traces in translatorx, but having a hard time finding a reason.

Matthew G. Loraditch - CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


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