Re: [cisco-voip] DRF add intelligence thread

[Wes Sisk (wsisk)]

In CSCvb81026 a workaround was reconfiguring OpenSSH on the windows server

SSHD config file for windows config file has  SSHMaxConnections=16 and 
SSHTimeout=60 respectively.  Removed these parameters and CDR-CAR backup 
completed successfully.

So far I do not see any option to disable or extend timeout of intelligence 
thread

On Jun 20, 2023, at 2:35 PM, Erick Bergquist  wrote:

Hello,

Does anyone know if the added intelligence thread of backup (drf) process can 
be disabled or timeout increased?

Thanks,
Erick 

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Re: [cisco-voip] Call Manager publisher - Database Communication Error

[Wes Sisk (wsisk)]

Yep. Several instance where I see this then even creation of remote account 
fails. At that point it is likely filesystem corruption. You can try recovery 
disk. But likely rebuild and restore.

-Wes

On Jun 5, 2023, at 8:17 AM, Charles Goldsmith  wrote:

platform config xml is created during install, which is something that TAC will 
have to root in and modify / resolve for you.

On Sun, Jun 4, 2023 at 9:01 AM Terry Oakley 
mailto:terry.oak...@rdpolytech.ca>> wrote:
Cannot login to the GUI on the publisher but can on both subscribers.
cli to publisher and ran 'show tech dbstateinfo'
[Fatal Error] platformConfig.xml:187:16: Character reference "" is an 
invalid XML character.

If I go to the output file 'showtechdbstatetxt I see the same fatal error 
as above

In the  log file I have

db...@xxx.rdc.xxx: User db...@xxx.rdc.xxx's password is not correct 
format for the database server.
Password validation for user (dbims) failed!
Check for password aging/account lock-out

run utils dbreplication runtimestate
same invalid xml character error
Sync completed
All tables are in sync


When I logon to the Subscribers (GUI) and run the System report Unified CM 
Database Status I get

Source has failed due to source on xxx timing out
The publisher database cannot be reached
The local database cannot be reached

Kind of spinning my wheels as I am not sure where to start to get the publisher 
back into the happy world of the CCM cluster and remote users will be trying to 
connect via Jabber and my remote test fails.

Thanking you in advance for any assistance/direction you can provide.

Terry


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

[Wes Sisk (wsisk)]

How about a BAT/BPS export?

-w

On May 25, 2023, at 2:26 PM, Tim Reimers  wrote:

Sorry, sent before being ready...

What I was hoping for was a list of all the registered active ATAs that would 
INCLUDE the DN on them.
As it is -- if I simply use the admin website, as you said, I can get a list of 
all the devices that are registered.

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

I cannot easily also match those up with the DNs.
I suppose one could probably then write some sort of SQL command that would 
specify a long-ish list of ATA device names to query for DN.

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

I thought about asking one of our DB people to do that --
But they'd want a project set up, and require a complete list of the table 
schema, the IP/TCP port and data connection, etc
to do things as they're used to, with a direct SQL connection to query against.
They're not used to having to operate the way Cisco does with CLI only access 
to SQL in a limited fashion.
And I get why Cisco does that... not arguing that point.




On Thu, May 25, 2023 at 2:18 PM Tim Reimers 
mailto:treim...@ashevillenc.gov>> wrote:
Nate, I may have to do that.

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

T

On Thu, May 25, 2023 at 2:08 PM mailto:natec...@gmail.com>> 
wrote:
What is the goal?  I usually get what I want from device/phone, copy it all 
into the clipboard and paste without formatting into excel and continue on.  In 
older versions of CUCM you can change the rows per page, then edit the URL to 
make rows per page all of the devices on the system up to many thousands.

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

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

Do any of the other risdb CLI filters give you those devices?
Interrogating the API directly is another option.
https://developer.cisco.com/docs/sxml/#!risport70-api-reference/selectcmdevice

-Ryan

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

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

I gather that you are all expecting that 'show risdb query phone' as well as 
RTMT should be showing the ATAs registered and counted alongside the SEP devices
so long as the ATAs are running in SCCP mode and not something else...

Thanks Tim

On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk) 
mailto:ws...@cisco.com>> wrote:
Yes, registration information is in RIS not in SQL(informix). I see some other 
mentions of this, but not clear resolution.

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

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

-w

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

Hi all -

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

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

* I'm using "show risdb query phones" command, as documented here among other 
sites
https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

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

I don't see any ATA devices being returned.
Are they not in the "phone" table of the RISDB?

Thanks, Tim

* my understanding is that any variation on the "run sql select" is
simply querying the Oracle? database for _configured_ devices only, and isn't 
looking
at the memory table of the Callmanager process to see the _currently 
registered_ devices.
(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, not necessarily 
"registered", so that does not seem appropriate for the info I want).



--

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

Tim Reimers

Network Administrator

I.T S

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

[Wes Sisk (wsisk)]

Interesting. If you try device search by name?
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200578-Monitor-Cisco-IP-Phones-Using-Call-Manag.html


If it shows as "registered" in ccmadmin then ccmadmin queries risdb for device 
status. We're missing something.

-w

On May 25, 2023, at 10:58 AM, Tim Reimers  wrote:

Hi Ryan, Wes, all --

I only have two nodes, and am checking both
I wondered if they're registering as SCCP or H.323, so I looked at H.323 as 
well - no show there either, only gateway devices.

In UCM web interface under Device>Phones dialog, there are ATAs registered.

Device Protocol shows as SCCP.
Firmware on one selected at random ATA is
ATA030203SCCP051201A.zup

A view of
admin:show risdb query phoneextn

also does not find the DN of any given ATA that I can otherwise find in the 
Device

I'm not sure what you mean in terms of RTMT.
That shows the number of registered phones --
When I did a Device Search, I didn't see a way to select ATAs in specific - 
just all other models of device
(which didn't work either, so I guess I was doing something wrong somehow)

Thanks Tim


On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk) 
mailto:ws...@cisco.com>> wrote:
Yes, registration information is in RIS not in SQL(informix). I see some other 
mentions of this, but not clear resolution.

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

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

-w

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

Hi all -

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

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

* I'm using "show risdb query phones" command, as documented here among other 
sites
https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

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

I don't see any ATA devices being returned.
Are they not in the "phone" table of the RISDB?

Thanks, Tim

* my understanding is that any variation on the "run sql select" is
simply querying the Oracle? database for _configured_ devices only, and isn't 
looking
at the memory table of the Callmanager process to see the _currently 
registered_ devices.
(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, not necessarily 
"registered", so that does not seem appropriate for the info I want).



--

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

Tim Reimers

Network Administrator

I.T Services

City of Asheville

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

(desk) 828-259-5512

(cell)   828-552-1585

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

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ipsos nexus

Tim Reimers

Network Administrator

I.T Services

City of Asheville

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

(desk) 828-259-5512

(cell)   828-552-1585

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

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

[Wes Sisk (wsisk)]

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

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

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

-w

On May 25, 2023, at 9:50 AM, Tim Reimers  wrote:

Hi all -

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

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

* I'm using "show risdb query phones" command, as documented here among other 
sites
https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/

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

I don't see any ATA devices being returned.
Are they not in the "phone" table of the RISDB?

Thanks, Tim

* my understanding is that any variation on the "run sql select" is
simply querying the Oracle? database for _configured_ devices only, and isn't 
looking
at the memory table of the Callmanager process to see the _currently 
registered_ devices.
(I've seen a number of other forum posts where people suggested "run sql" 
commands to gather info, but that is statically configured, not necessarily 
"registered", so that does not seem appropriate for the info I want).



--

Quis custodiet  
ipsos nexus

Tim Reimers

Network Administrator

I.T Services

City of Asheville

treim...@ashevillenc.gov

(desk) 828-259-5512

(cell)   828-552-1585

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

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Re: [cisco-voip] Call Flow data via transfers

[Wes Sisk (wsisk)]

Generally speaking there should be adequate info in CDR to track the multiple 
call legs. There are examples of tracking call flows here:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1_SU7/cucm_b_reporting-billing-administration-guide-1151su7/cucm_b_reporting-billing-administration-guide-1151su7_chapter_01000.html

Might be interesting to hear from ISI how they interpret the data provided and 
what gaps exist.

-w

On Sep 20, 2022, at 10:27 AM, Scott Voll 
mailto:svoll.v...@gmail.com>> wrote:

So we are on CM 12.5  also using contact center express.

we use isi for our CDR's.

is there a way to track call flow from call coming into our contact center and 
being transferred to hunt groups, or other people, vm - then zero outs, and to 
other places?

seems to be a large hole in our data.  Just wondering if there is something i'm 
missing, or a better application to get this data from?

Not opposed to changing up applications if needed.  maybe a reception console 
app could provide this?  I don't think our UCCx is the correct app for our 
reception desk anyway.

Thanks

Scott
PS. does M$ Teams have this kind of issue also?


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Re: [cisco-voip] Forced Authorization codes in Webex calling

[Wes Sisk (wsisk) via cisco-voip]

I'm not sure about the webex calling implementation.

In UCM it was intentional to write CMC to CDR and to not write FAC to CDR. That 
would be kind of like writing passwords on stickies and posting them in the 
hallway.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_01.html

"The CMC writes to the CDR, so you can collect the information by using CDR 
Analysis and Reporting (CAR), which generates reports for client accounting and 
billing."
...
"The name of the authorization writes to call detail records (CDRs), so you can 
organize the information by using CDR Analysis and Reporting (CAR), which 
generates reports for accounting and billing."

-Wes

On Nov 19, 2021, at 8:49 AM, SK 
mailto:cciecollab2...@gmail.com>> wrote:

Hello all ,

Has anyone come across scenario where forced authorization codes data needs to 
be retrieved on Webex calling platform ? It appears it’s not sent along with 
the CDRs. Any other ideas on how to capture this?

Thank you .
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Re: [cisco-voip] Of Expressways and max-forwards...

[Wes Sisk (wsisk) via cisco-voip]

I think maybe you're referring to "forward maximum hop count". That parameter 
is used to guard against call routing loops caused by numbers being forwarded.

There is also a "max hop" parameter for multicast music on hold. That might be 
more related to high cpu on boundary devices or failing to receive MMoH.

The risk of increasing the forward hop loop detection is that call routing 
loops caused by forwarded numbers will consume more circuits. For IP circuits 
this may not be a big deal. For your pots/t1/e1 circuits you are more likely to 
consume all available channels and experience fully used circuits. When this 
happens you're like to see spikes in CPU usage and may experience call 
throttling like code yellow depending on the rate and extent of the forward 
loop.

Forward max hop count was put in place to address the situation where 2 users 
forward their calls to each other. Example: Market and Sales both leave for 
different appointments and forward their call to the other "for coverage". A 
call to marketing will reach that marketing line and be forwarded to sales. It 
reaches sales and is forwarded to marketing. Forwards can be internal or 
external, and can use number translations and different call routing paths. The 
forwards are end user initiated and the end user stations/lines may existing on 
separate systems where there is no single administrative, monitoring, or 
operational domain.

IP trunks do change this picture some. Call flows that involve this many 
forwards may be worth a design review.

Expected impact: In the event of a call forwarding loop you are more likely to 
consume all available traditional trunks. You may experience high cpu and 
potentially call throttling or code yellow.
Unexpected impacts: Introducing that many touch points in a call flow creates 
many opportunities for unexpected impacts. Pretty cool that that many call 
processing events can happen in close enough to real time for the call to be 
delivered end to end.

Good persistence and great find Gary! Even more kudos for taking time to share 
your experience with the community.

-Wes

On Nov 15, 2021, at 10:34 AM, Gary Parker 
mailto:g.j.par...@lboro.ac.uk>> wrote:

Eventually we narrowed it down to calls from MRA registered devices on our 
expressways (mostly Jabber but with a small number of 8845s in staff home 
offices), as all failed calls had the same source IP address when we looked at 
the corresponding CDRs; although this wasn’t visible in the SIP traces which 
made diagnosis harder (source IP address is the subscriber when looking at the 
CUBE’s SIP ). A quick look at edge and core expressways showed that “max hops” 
was to set to 15 in the relevant zones. Cisco documentation says this is the 
default, but suggests to set it higher if calls are failing with a 483 code. So 
we set the max hops to 70 and calls are now connecting as expected.



So…question: why is the max hops set so low (15) on expressway zones by default 
when it’s set to 70 on CUBEs, and is there anything this is likely to 
break/that I should look out for now I’ve made the change?

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Re: [cisco-voip] Low space on partitions in CUCM

[Wes Sisk (wsisk) via cisco-voip]

CSCvt97709 was for systems that installed using an older 80GB OVA and then 
upgraded.

Partition sizes documented in CSCvt97709:
TotalFreeUsed
Disk/active 14154228K 319736K   13689364K (98%)
Disk/inactive   14154228K 828104K   13180996K (95%)
Disk/logging49573612K   37501812K9530488K (21%)


If a system is still using those partition sizes then a rebuild is worth 
consideration. Rebuilds are easier with recent automations.

If you’re seeing alerts with larger partition sizes then more investigation may 
be warranted.

-Wes



On Sep 29, 2021, at 10:21 AM, Pete Brown 
mailto:j...@chykn.com>> wrote:

Yes?

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Wednesday, September 29, 2021 9:09 AM
To: Matthew Huff mailto:mh...@ox.com>>; Tim Smith 
mailto:tim.sm...@enject.com.au>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Low space on partitions in CUCM

Oh for the love of pete.

Severity:
4 Minor


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Matthew Huff
Sent: Wednesday, September 29, 2021 10:00 AM
To: Tim Smith mailto:tim.sm...@enject.com.au>>
Cc: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Low space on partitions in CUCM

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

Evidently this is a known bug:

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt97709

The “solution” is to do the upgrade, then do an export and a full redeployment 
with a new OVA template.

Since there are trust, certs, etc…involved in this, it is a very involved 
procedure. Ughhh…


Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | 
www.ox.com
...

From: Tim Smith mailto:tim.sm...@enject.com.au>>
Sent: Wednesday, September 29, 2021 8:05 AM
To: Matthew Huff mailto:mh...@ox.com>>
Cc: James Buchanan 
mailto:james.buchan...@gmail.com>>; Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Low space on partitions in CUCM

I've had these alerts pop up on fresh install during a PCD migration (new VM's) 
(it was 12.5 from memory)
I spent a bit of time back and forth with TAC and checking with other TAC 
engineers etc.
I think the outcome was basically that it was the standard setup of the Active 
Partition, and it isn't really an issue, except that you will get those alerts 
(and you can adjust the thresholds)

But it's only a partition inside, and basically you have other space available.

What does your show diskusage look like?



On Wed, 29 Sept 2021 at 21:56, Matthew Huff mailto:mh...@ox.com>> 
wrote:
I deleted the old firmware files and we don’t use any MOH sources other than 
the built in one. The coppfile is for cleaning up the common partition, but I 
ran it anyway. The active parttion still shows 98% used. I’m afraid TAC is 
correct, the older ova file that we used for 11.5 created too small of an 
active partition for 14, and the only solution is to do a complete rebuild.

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | 
www.ox.com
...

From: James Buchanan 
mailto:james.buchan...@gmail.com>>
Sent: Wednesday, September 29, 2021 7:37 AM
To: Matthew Huff mailto:mh...@ox.com>>
Cc: 

Re: [cisco-voip] Jabber: Wrong sequence number

[Wes Sisk (wsisk) via cisco-voip]

Hello Reto,

This looks like every RTP packet is duplicated.

This could be an artifact of how the packet capture was taken. Was a SPAN or 
similar feature used that might duplicate packets? Are the packets truly 
duplicates? Sometimes this IP ID field is useful

Otherwise, this could be a recording stream sent to a recording server.

In that situation this might be relevant:
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuw36065/?rfs=iqvred

Is this a recording stream and you are observing problems with call recording?

-Wes

On Apr 28, 2021, at 11:41 AM, Reto Gassmann mailto:v...@mrga.ch>> 
wrote:



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

[Wes Sisk (wsisk) via cisco-voip]

Andy, sounds like a good start:
https://community.cisco.com/t5/ip-telephony-and-phones/disaster-recovery-problem/td-p/2767755

I see 2 other situations that might be relevant:
1. All the same .cop files not installed
2. Attempting to restore a backup of a different version


-Wes

On Feb 19, 2021, at 6:46 PM, Andy Carse 
mailto:andy.ca...@gmail.com>> wrote:

Wes,
I select Restore Wizard then select the backup device
click next
The Ccx then spins the hour glass for 5 mins then says
“Restore request timing out. Either master agent is down or Sftp server is 
inaccessible or too slow to respond”

It’s the same location all the other UC apps backup to. Could it be that there 
are too many files in the directory?
Even though they would have different names etc?

It seems to do a couple of hundred new sessions for some reason looking at the 
backup server syslog.
It’s keeping 14 versions of cucm cluster backups is that too many, although 
I’ve not seen anything to say so.

I’m going to change the file path tomorrow and see what happens with that.

Andy

On Fri, 19 Feb 2021 at 19:16, Wes Sisk (wsisk) 
mailto:ws...@cisco.com>> wrote:
What is the exact error? What do DRS logs show?

I see one report that after re-install dbreplication is not established leading 
to "Unable to send network request to master agent. This may be due to Master 
or Local Agent being down”

Resolved by resetting dbrepliaction for all nodes.

Thanks,
Wes

On Feb 19, 2021, at 1:51 PM, Andy Carse 
mailto:andy.ca...@gmail.com>> wrote:

So I thought that if a CUCM could backup to and sftp server without issue, that 
it would be able to restore.
But that turns out to be wrong.

The cluster can backup to sftp without issue, but when I try to restore said 
backup, the restore seems to make a large amount of ssh connections and times 
out saying the DRS maybe down of the sftp server is tacking too long to respond.

this is the sftp server syslog output.
Feb 19 18:29:49 sukucmbkup systemd[1]: Started Session 226 of user support.
Feb 19 18:29:49 sukucmbkup systemd[1]: Started Session 227 of user support.
Feb 19 18:29:50 sukucmbkup systemd[1]: Started Session 228 of user support.
Feb 19 18:29:51 sukucmbkup systemd[1]: Started Session 229 of user support.
Feb 19 18:29:52 sukucmbkup systemd[1]: Started Session 230 of user support.
Feb 19 18:29:53 sukucmbkup systemd[1]: Started Session 231 of user support.

any pointers grateful

Rgds Andy

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

[Wes Sisk (wsisk) via cisco-voip]

What is the exact error? What do DRS logs show?

I see one report that after re-install dbreplication is not established leading 
to "Unable to send network request to master agent. This may be due to Master 
or Local Agent being down”

Resolved by resetting dbrepliaction for all nodes.

Thanks,
Wes

On Feb 19, 2021, at 1:51 PM, Andy Carse 
mailto:andy.ca...@gmail.com>> wrote:

So I thought that if a CUCM could backup to and sftp server without issue, that 
it would be able to restore.
But that turns out to be wrong.

The cluster can backup to sftp without issue, but when I try to restore said 
backup, the restore seems to make a large amount of ssh connections and times 
out saying the DRS maybe down of the sftp server is tacking too long to respond.

this is the sftp server syslog output.
Feb 19 18:29:49 sukucmbkup systemd[1]: Started Session 226 of user support.
Feb 19 18:29:49 sukucmbkup systemd[1]: Started Session 227 of user support.
Feb 19 18:29:50 sukucmbkup systemd[1]: Started Session 228 of user support.
Feb 19 18:29:51 sukucmbkup systemd[1]: Started Session 229 of user support.
Feb 19 18:29:52 sukucmbkup systemd[1]: Started Session 230 of user support.
Feb 19 18:29:53 sukucmbkup systemd[1]: Started Session 231 of user support.

any pointers grateful

Rgds Andy

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Re: [cisco-voip] [External] Is it possible to bulk export phones from CLI or API?

[Wes Sisk (wsisk) via cisco-voip]

Not easily. You’d need to join ~4-5 tables to get that information. Any 
especial characters would botch record delineation in CLI output.

https://developer.cisco.com/docs/axl/#!12-5-cucm-data-dictionary


-Wes

On Feb 18, 2021, at 2:11 PM, Louis Koekemoer (MEA) 
mailto:louis.koekem...@dimensiondata.com>> 
wrote:

I am mainly interested in 4 fields on the bulk phone reports

Device Name,Description,Directory Number 1,Line CSS 1

And for UDP reports
Device Profile Name,Description,Directory Number 1,Line CSS 1

From: Hunter Fuller mailto:hf0...@uah.edu>>
Sent: Thursday, 18 February 2021 21:07
To: Louis Koekemoer (MEA) 
mailto:louis.koekem...@dimensiondata.com>>
Cc: cisco-voip mailto:cisco-voip@puck.nether.net>>
Subject: Re: [External] [cisco-voip] Is it possible to bulk export phones from 
CLI or API?


What kind of data do you need to report?

I usually do stuff like this using a python tool I wrote, called axlrows. Here 
is an example:

>>> gen = get_phones(name="SEP700B4F9C44B8")
>>> phones = get_phones(name="SEP700B4F9C44B8")
>>> for phone in phones:
print(phone.name, ":", phone.description)
SEP700B4F9C44B8 : Hunter Fuller (Cisco 8851 SIP) - Home
It looks like to match "every" phone, you could write get_phones(protocol="SIP")

You can get the tool here, though it is admittedly poorly documented, so if you 
need assistance, let me know off-list.

https://github.com/uah/axlrows

--
Hunter Fuller (they)
Router Jockey
VBH Annex B-5
+1 256 824 5331

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


On Thu, Feb 18, 2021 at 12:47 PM Louis Koekemoer (MEA) 
mailto:louis.koekem...@dimensiondata.com>> 
wrote:
Hi,

I recently upgraded my CUCM clusters to CUCM 12.5su3. I hit a bit of an issue. 
2 bugs. 1 on exporting all devices or all UDP, all details and 2 generating UDP 
reports or Phone reports. I need to do these on a weekly basis and I have 3 
cluster with around 7000-1000 UDP per cluster and up to 33000 phones/devices 
per cluster. Now as per the below it breaks on around 700-800 devices. I logged 
a TAC case and this is what TAC identified. Firstly the customer is not happy 
with bleeding edge(Not released yet). I am not very clued up on API’s but is 
there possible a way I can either “export” all devices, all details via cli or 
with a API? I have a few more clusters to upgrade but all put on hold due to 
this.


1. https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvv67192

Symptom:
Export of Bulk Phones Job , stays in "Processing" State and does not "Completes"
Export of Bulk UDP Job , stays in "Processing" State and does not "Completes"

Conditions:
Export fails if it's more than 520 phone records through Phone -> Export 
Phone/UDP -> All Details

Workaround:
1. Export phones around 400 records through Phone -> Export Phone -> Specific 
Details
2. Export 800+ records through Import/Export -> Export operation.


The known fixed version is CCM.12.5(1.14900.42) but is going to be released in 
late February.

2. https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvw44625


Symptom:
When customer submits bat job for (Bulk Administration --> Phones --> Generate 
Phone Reports) failed event can be observed from Job Scheduler page.

Issue started after upgrading to CUCM 12.5 SU3

Conditions:
Generate Phone Report under Bulk Administration

Workaround:
While Generating the  Phone Report,
Bulk Administration --> Phones --> Generate Phone Reports  after selecting 
mandatory fields,
under IP Phone services fields two checkboxes are there for speed dial services 
and IP Phone services, if the user can checked those boxes and run the job 
Report will generate successfully

The known fixed version is CCM.12.5(1.14900.42) but is going to be released in 
late February.


Louis Koekemoer (MEA)
Engineer - IPT - L4
m. +27 71 6808790
t. +27 11 5754317
email. 
louis.koekem...@dimensiondata.com
dimensiondata.com





.
.

This email and all contents are subject to the following disclaimer:
https://www.dimensiondata.com/email-disclaimer

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Re: [cisco-voip] CUCM call set up issue after migration

[Wes Sisk (wsisk) via cisco-voip]

On UCM cli ‘utils diagnose test’ to ensure things look mostly okay from OS and 
low level perspective.

After that in UCMadmin ensure system->server shows the correct IP/name for each 
server.

After that ensure ccm process restarted (or whole server rebooted). Based on 
your description SDL links on tcp:8002 are not coming up because remote name/ip 
is not recognized against what is the the database and hosts file for name 
resolution.

-w

On Jan 12, 2021, at 4:05 PM, Riley, Sean 
mailto:sri...@robinsonbradshaw.com>> wrote:

This past weekend we migrated 2 CUCM servers to a new datacenter.  This 
involved changing the IP address on these 2 CUCM nodes.  These 2 nodes consist 
of the Publisher and 1 Subscriber.  We have another Sub at a remote datacenter 
that was not touched this past weekend.

Node configuration:

DC A
CM1: Pub which was re-ip’d
CM2: Sub which was re-ip’d

DC B:
CM3: Sub at remote site that was not changed

Phones are at many sites, but issue is independent of the phone type, phone 
location or subnet.  Also, Expressway phones have the same issue.

The issue is any phone that is registered to CM3 cannot call phones registered 
to CM1 or CM2 and vice versa.  The phones do not see the call coming in.  If 
SNR is configured, the call will ring to the remote destination. Phones 
registered to CM3 can make outbound PSTN calls without issue, but not receive 
inbound from PSTN (probably because the gateway is handing off to CM1 or CM2).  
While the gateways are not unique to the issue, they are running H323.

If the phones are both registered to CM3, they can call each other, but not 
phones registered to CM1 or CM2.

I have had my network team verify there is not anything they can see in the 
network causing this behavior. Database replication checks out OK and I can 
ping from/to each node.

Anyone able to point me in the right direction to figure this out?

Thanks.
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Re: [cisco-voip] [External] Re: List still active?

[Wes Sisk (wsisk) via cisco-voip]

As a person who still processes 400+emails per day for more than a few years 
now (gulp) I hear you about the abundance of information and communication 
channels. I am still working on integrating WxT in my information processing 
model. I am still decommissioning my SDLRouterThread,SDLTimerThread, and 
HighPriorty and VeryLowPriority queues.

And I personally think the abundance of information will always the case. There 
will always be more happening and communicated than I can consume.

So my task is to choose where I invest my attention. I still read this list. I 
still chime in. I still followup. I still think this important.

I enjoy this as a metaphor for stream processing and see that only increasing 
with streaming data, cloud, devops, always on. Reference, “no man ever steps in 
the same river twice”, CI/CD, Webex, Intersight, or Borg.

I appreciate the question "if this community is still active". I appreciate 
someone is watching that angle, willing to ask the question, and that the 
community is willing to take it up. Thank you. I’m glad to see the cooperation 
of this list still in place. It’s almost like ‘communication’ is important here.

We might choose to move to communities, or forums, or WxT, or something else. 
Whatever tool we might use I hope the sharing, cooperation, and communication 
are still the underpinning.


-Wes


On Dec 25, 2020, at 3:55 PM, Hunter Fuller via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

Wow. I struggle enough keeping up with the real-time chats from my colleagues 
and customers. I definitely can't handle a chat for every list I'm subscribed 
to. Not sure how those people do it.

Merry VoIPmas!

--
Hunter Fuller (they)
Router Jockey
VBH Annex B-5
+1 256 824 5331

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


On Fri, Dec 25, 2020 at 2:51 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
Chat services like Webex Teams and Discord have killed the list, IMO.

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

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

Sent from my iPhone

> On Dec 25, 2020, at 14:30, Bill Talley 
> mailto:btal...@gmail.com>> wrote:
>
> Thanks for the confirmation Ryan.  Are you also seeing a significant decline 
> in volume from the group?
>
> Hope all the usual (and even casual) participants are staying healthy and 
> employed.  Hope those aren’t reasons for the decline in forum usage.
>
> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
> Please excude my typtos.
>
>> On Dec 25, 2020, at 1:28 PM, Ryan Huff 
>> mailto:ryanh...@outlook.com>> wrote:
>>
>> I still see you.
>>
>> Sent from my iPhone
>>
 On Dec 25, 2020, at 14:28, Bill Talley 
 mailto:btal...@gmail.com>> wrote:
>>>
>>> I stopped receive list emails.   Is the list dead or was I banned? 
>>>
>>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>>> Please excude my typtos.
>>> ___
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[cisco-voip] Collaboration Solutions Analyzer (CSA) upcoming talk

[Wes Sisk (wsisk) via cisco-voip]

Upcoming Talk:
https://community.cisco.com/t5/online-tools-and-resources/collaboration-solutions-analyzer-support-talks-event/ba-p/4178251

CSA can help to identify and diagnose issues in Collaboration deployments 
including device registration and call failures.

CSA Tool:
https://cway.cisco.com/csa/

-Wes
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Re: [cisco-voip] Calls made into iPad via Jabber/WebEx teams Unified CM

[Wes Sisk (wsisk) via cisco-voip]

Possibly:
https://www.cisco.com/c/en/us/support/docs/conferencing/telepresence-server/118345-probsol-telepresence-00.html

-Wes


On Apr 2, 2020, at 12:11 PM, James Dust 
mailto:james.d...@charles-stanley.co.uk>> 
wrote:

That’s fixed it thank you Kent,














From: Kent Roberts mailto:k...@fredf.org>>
Sent: 02 April 2020 16:55
To: James Dust 
mailto:james.d...@charles-stanley.co.uk>>
Cc: cisco-voip@puck.nether.net; Adam Sturley 
mailto:adam.stur...@charles-stanley.co.uk>>
Subject: Re: [cisco-voip] Calls made into iPad via Jabber/WebEx teams Unified CM

This message originates from outside Charles Stanley. Please do not click links 
or open attachments unless you know the sender and are confident the content is 
safe.
Try turning off video on the jabber in cucm.We have a similar issue for 
some folks

Kent


On Apr 2, 2020, at 03:19, James Dust 
mailto:james.d...@charles-stanley.co.uk>> 
wrote:

Good morning all,

I have an end user working at home, who’s inbound calls to their iPad instantly 
cut off as soon as they are answered.

This happens when using both Jabber and WebEx teams (Unified CM call behaviour) 
but outbound calls are fine.

We are using expressway and CUCM 11.5

Any help appreciated.

Thanks

James

Consider the environment - Think before you print

The contents of this email are confidential to the intended recipient and may 
not be disclosed. Although it is believed that this email and any attachments 
are virus free, it is the responsibility of the recipient to confirm this.

You are advised that urgent, time-sensitive communications should not be sent 
by email. We hereby give you notice that a delivery receipt does not constitute 
acknowledgement or receipt by the intended recipient(s).

Details of Charles Stanley group companies and their regulators (where 
applicable), can be found at this URL 
http://www.charles-stanley.co.uk/contact-us/disclosure/

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Consider the environment - Think before you print

The contents of this email are confidential to the intended recipient and may 
not be disclosed. Although it is believed that this email and any attachments 
are virus free, it is the responsibility of the recipient to confirm this.

You are advised that urgent, time-sensitive communications should not be sent 
by email. We hereby give you notice that a delivery receipt does not constitute 
acknowledgement or receipt by the intended recipient(s).

Details of Charles Stanley group companies and their regulators (where 
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[cisco-voip] Temporary Licenses for UCM, Jabber, Webex Teams, Phones for COVID-19

[Wes Sisk (wsisk) via cisco-voip]

Keep on connecting those users:

https://www.cisco.com/c/m/en_us/products/unified-communications/unified-communications-manager-callmanager/obtaining-temporary-licenses.html

-Wes
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Re: [cisco-voip] 12.5 Upgrade files posted on CCO

[Wes Sisk (wsisk) via cisco-voip]

When last I looked PUT fulfillment was by electronic posting and download in 
countries where that is legal. Some geo’s require physical media and the 
creates a shipping limitation.

What inefficiency do you experience with PUT?

Thanks,
Wes

On Feb 6, 2020, at 12:17 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

I can handle the bootable issue far faster and more efficiently than the PUT 
process can, which is one of the reasons why I’ve found PUT to not be super 
useful to me.

Sent from my iPhone

On Feb 6, 2020, at 11:33, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:


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



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

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

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

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

Anyone able to confirm these are the same files?
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Re: [cisco-voip] 12.5 Upgrade files posted on CCO

[Wes Sisk (wsisk) via cisco-voip]

this.

When last I looked:
PUT does create a sales order which helps address any licensing questions, 
re-issues, etc.
‘Upgrade’ media isn’t always the same as install media. PUT gets bootable 
install media in case of reinstall or disaster.

And PUT doesn’t get the latest SU. PUT gets latest Maintenance Release.

Thanks,
Wes

On Feb 6, 2020, at 11:32 AM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:

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



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

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

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

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

Anyone able to confirm these are the same files?
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Re: [cisco-voip] DNA Multiple Analyzer 11.5+ Broken

[Wes Sisk (wsisk) via cisco-voip]

Yes, this is still open. Inquiry initiated.  Thank you for the reminder. Bonus 
points if you can open a case that we can link to the bug.

-Wes

On Oct 3, 2019, at 12:50 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

Source: https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuy89525

It's still open too, so I would assume anything new is also affected.

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Re: [cisco-voip] CPU Reservations

[Wes Sisk (wsisk) via cisco-voip]

Adam,

Yes, the reservation appears steep and outside monitors appear ripe for 
harvesting underutilized resources.


And 3 things come to mind:

* ccm.bin at its core is a distributed real time state machine. Delays have 
serious impacts. Feature are invoked across nodes. See the long history around 
CCM clustering over the WAN requirements for more background.
* mtp/moh/cfb are all essentially real time for audio.
* some processes are less predictable in their consumption of resources. 
“Resource Starvation” has many symptoms across features that are difficult to 
diagnose. This consumes more of your time as implementers, admins, and 
monitors. And it significantly degrades customer experience of the product and 
people servicing the product. There is an element of resource allocation that 
is akin to road infrastructure planning - it may appear idle many times and it 
can still be under provisioned during peak demand.

Consider for example
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtu18692

This under provision/over subscription negatively impacted several deployments.

In my experience VOIP has been a bit of a “canary in the coal mine” for IP 
applications. I see many situations where users are willing to re-send an 
email. Meanwhile the first call that does not go through, gets dropped, or 
suffers poor audio is grounds for immediate escalation.

Maybe some things to consider.

-Wes

On Jul 8, 2019, at 11:57 AM, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:

Hi all,

It’s been a bit since I’ve asked this question, if I have here before.

Do we all run our UC appliances in VMWare with the full CPU MHz and core 
reservations prescribed by Cisco, in production? Or, if you have information on 
hand regarding the actual resource usage, have any of you taken on resizing the 
VM reservations?

The various documents are very much so clear that oversubscription isn’t 
supported, but, it also talks about vCPU to cores which I’m told doesn’t really 
play out in VMWare as it’s a MHz reservation that can be scheduled in to 
available hardware.

There are various statements peppered in about running your own VM environment 
with best practices – but also the 1:1 pcore:vcore comments.

Is anyone turning these knobs? Has anyone stepped over that pcore:vcore line 
when it appears there are enough resources?

I’m looking for thoughts or unforeseen consequences that we can use to back 
somewhat of the case as to why we need to continue to fund hardware at scale 
which is largely idle.

Adam


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Re: [cisco-voip] 8851: always use first line for outbound calls with multiline

[Wes Sisk (wsisk) via cisco-voip]

Correction: it will not be visible.

On Mar 21, 2019, at 2:58 PM, Wes Sisk (wsisk) via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

Whoops, I thought it was. Thanks for the callout Lelio! It should be visible in 
24 hours.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvh30732

-Wes

On Mar 21, 2019, at 11:40 AM, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Wes - is there any way to make this big public? I’d like to understand what 
phone models and/or software this affects.

-sent from mobile device-

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

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

[University of Guelph Cornerstone with Improve Life tagline]

On Mar 20, 2019, at 1:03 PM, Wes Sisk (wsisk) via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

Please note at this point the 8851 product manager has elected not to address 
that request. If this feature is important to you then please press your 
account representative more.

Regards,
Wes

On Mar 20, 2019, at 5:56 AM, James Dust 
mailto:james.d...@charles-stanley.co.uk>> 
wrote:

Hi Reto,

We also have this issue, and it has been raised via our vendor with Cisco TAC

Cisco have raised this with their development team they have told us.

I believe the related bug id is: Bug ID CSCvh30732




Kind Regards

James Dust
ICT Network Infrastructure & Communications Engineer











55 Bishopsgate, London EC2N 3AS

Telephone: 020 7149 6314 / 07989 491136
Website<http://www.charles-stanley.co.uk/> | 
LinkedIn<https://www.linkedin.com/company/charles-stanley> | 
Twitter<https://twitter.com/_CharlesStanley>


















From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Reto 
Gassmann
Sent: 20 March 2019 09:54
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] 8851: always use first line for outbound calls with 
multiline

Hello Group

we have some IP Phones (8851) with two lines. For outgoing calls the first line 
is always used.
However, if the user receives a call on the second line (This is his secret 
privat line), the next outgoing call is initiated from the second line and 
sends the wrong (privat) number.
Is it possible to configure the phone that allways and automatically the first 
line is used for an outgoing call.
We have CUCM 10.5 and 8851 IP Phones.

Thanks a lot
Regards Reto

Consider the environment - Think before you print

The contents of this email are confidential to the intended recipient and may 
not be disclosed. Although it is believed that this email and any attachments 
are virus free, it is the responsibility of the recipient to confirm this.

You are advised that urgent, time-sensitive communications should not be sent 
by email. We hereby give you notice that a delivery receipt does not constitute 
acknowledgement or receipt by the intended recipient(s).

Details of Charles Stanley group companies and their regulators (where 
applicable), can be found at this URL 
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Re: [cisco-voip] 8851: always use first line for outbound calls with multiline

[Wes Sisk (wsisk) via cisco-voip]

Please note at this point the 8851 product manager has elected not to address 
that request. If this feature is important to you then please press your 
account representative more.

Regards,
Wes

On Mar 20, 2019, at 5:56 AM, James Dust 
mailto:james.d...@charles-stanley.co.uk>> 
wrote:

Hi Reto,

We also have this issue, and it has been raised via our vendor with Cisco TAC

Cisco have raised this with their development team they have told us.

I believe the related bug id is: Bug ID CSCvh30732




Kind Regards

James Dust
ICT Network Infrastructure & Communications Engineer











55 Bishopsgate, London EC2N 3AS

Telephone: 020 7149 6314 / 07989 491136
Website | 
LinkedIn | 
Twitter


















From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Reto 
Gassmann
Sent: 20 March 2019 09:54
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 8851: always use first line for outbound calls with 
multiline

Hello Group

we have some IP Phones (8851) with two lines. For outgoing calls the first line 
is always used.
However, if the user receives a call on the second line (This is his secret 
privat line), the next outgoing call is initiated from the second line and 
sends the wrong (privat) number.
Is it possible to configure the phone that allways and automatically the first 
line is used for an outgoing call.
We have CUCM 10.5 and 8851 IP Phones.

Thanks a lot
Regards Reto

Consider the environment - Think before you print

The contents of this email are confidential to the intended recipient and may 
not be disclosed. Although it is believed that this email and any attachments 
are virus free, it is the responsibility of the recipient to confirm this.

You are advised that urgent, time-sensitive communications should not be sent 
by email. We hereby give you notice that a delivery receipt does not constitute 
acknowledgement or receipt by the intended recipient(s).

Details of Charles Stanley group companies and their regulators (where 
applicable), can be found at this URL 
http://www.charles-stanley.co.uk/contact-us/disclosure/

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Re: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Wes Sisk (wsisk) via cisco-voip]

Nick,

The features you describe are propagated by both SDL signaling and with a 
dependence on database replication.

At casual observation it sounds like database traffic between nodes may not 
prioritized and may be delayed or dropped.

The 80 msec is especially important for near real-time convergence of the 
distributed processes. Concurrently database replication plays a critical role 
as every process reads its local database.

Very casually:
node1: "Hey node 2, RouteList5 changed”
node2: “okay, let me read the changes from my local database”
node2: I don’t see any changes….

In the mean time database replication is held up in the network….

-Wes


On Nov 6, 2018, at 3:31 PM, Nick Barnett 
mailto:nicksbarn...@gmail.com>> wrote:

We think it is happening frequently WITHOUT this command being ran. Weird stuff 
happens... like deleting a speed dial and it never goes away... or changing the 
distribution order on a route list that auotmatically reverts back after a few 
seconds... or maybe the GUI shows it never reverted back however it is clearly 
not performing the correct algo. I can duplicate the RTT issue by raising the 
packet size to 1200 and doing a repeat 100 packets. it WILL give me times over 
80ms. BUT, the SDL traffic is supposed to be QOS in a certain way and I'm sure 
that the pings I'm doing are NOT being classified and queued properly. It is 
very frustrating that I know what I'm talking (enough to discuss with them, but 
it has been 7 years since I was 100% router jockey) about and can't get them to 
pay attention to a probable network issue.

I have an IP SLA running that shows average latency in the 20ms range. IP SLA 
is a fake red herring if you ask me... it only looks at an AVERAGE every 5 
minutes and if there are no issues, of course it will look great.

Thanks,
Nick

On Tue, Nov 6, 2018 at 12:42 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
You are able to correlate the out-of-band RTT to only when the dbreplication 
stat command is ran, or are there other times the RTT is OOB that isn't related 
to querying the replication status?

Thanks,

-R

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Nick Barnett 
mailto:nicksbarn...@gmail.com>>
Sent: Tuesday, November 6, 2018 11:57 AM
To: Cisco VoIP Group
Subject: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

We all know the max latency is 80ms, but ours occasionally goes over. I'm 
trying to track down why but the network team cannot find an issue. We are able 
to reproduce the issue repeatedly by running "utils dbreplication 
runtimestate." Whether this is causing the issue (I doubt it) or that command 
just takes long enough to run that it will eventually find a time that is > 
80ms (my guess Is yes)... I'm not 100% sure.

We opened a case with TAC to find out what that command is actually doing, but 
they won't divulge the info that our network team needs.

My theory is that it's actually calling some shell script in redhat under the 
CLI appliance layer. Has anyone investigated that? Do we know what this command 
is actually doing? Specifically, i want to know where it's getting those ping 
times... is it running a generic ping with generic datagram data? Is it sending 
a 1497 packet of 0x and then 0x? Basically, I'm trying to give the 
network team something to go on because they are saying it's not them. (Of 
course they could run a packet capture and tell me (mostly) what it's doing, 
but it's hard to get their attention when they don't think it's on their end).

Thanks,
Nick

P.S.  We have frequent DB replication issues... at least a few times per 
quarter. This is so annoying and I'm pretty sure it's due to this latency, but 
I can't get anyone to pay attention.
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Re: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Wes Sisk (wsisk) via cisco-voip]

Nick,

The command is invoking database commands that Cisco does not own. They are not 
being obtuse; they genuinely do not know.

It will cause a spike in database communication between nodes.

My first guess is very much in line with yours that the burst in traffic 
exceeds certain QoS queues.

IMHO - and I emphasize the MY in that - this a rather classic discussion point 
between application teams and network teams.

What Matt suggests in a subsequent response is the the rather data intensive 
way of getting that information. Fortunately wireshark has graphs for round 
trip time.

-Wes

On Nov 6, 2018, at 11:57 AM, Nick Barnett  wrote:

We all know the max latency is 80ms, but ours occasionally goes over. I'm 
trying to track down why but the network team cannot find an issue. We are able 
to reproduce the issue repeatedly by running "utils dbreplication 
runtimestate." Whether this is causing the issue (I doubt it) or that command 
just takes long enough to run that it will eventually find a time that is > 
80ms (my guess Is yes)... I'm not 100% sure.

We opened a case with TAC to find out what that command is actually doing, but 
they won't divulge the info that our network team needs.

My theory is that it's actually calling some shell script in redhat under the 
CLI appliance layer. Has anyone investigated that? Do we know what this command 
is actually doing? Specifically, i want to know where it's getting those ping 
times... is it running a generic ping with generic datagram data? Is it sending 
a 1497 packet of 0x and then 0x? Basically, I'm trying to give the 
network team something to go on because they are saying it's not them. (Of 
course they could run a packet capture and tell me (mostly) what it's doing, 
but it's hard to get their attention when they don't think it's on their end).

Thanks,
Nick

P.S.  We have frequent DB replication issues... at least a few times per 
quarter. This is so annoying and I'm pretty sure it's due to this latency, but 
I can't get anyone to pay attention.
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Re: [cisco-voip] A Little Monday CWMS Info

[Wes Sisk (wsisk) via cisco-voip]

Working on it.

Until it is resolved, and not the greatest workaround as it contains other 
things, try looking under “Cisco Meeting Server”.

For example:
CSCve59084CWMS OVA can not be deployed on vCenter 6.5

-Wes

On Jul 30, 2018, at 2:36 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

For those of you who have manage to dodge the CWMS bullet thus far, you may not 
be aware that the product has reached God-like status, as far as software is 
concerned.

It has zero documented defects, and is self healing.



The system is designed to repair itself when necessary and rebooting can 
interrupt this process.
Source: CWMS 3.0 Planning Guide, Best Practices 
Section
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Re: [cisco-voip] Looking for insight into how Cisco 8851 deals with DNS TTL

[Wes Sisk (wsisk)]

Looks like it was known broken at one point:

The DNS Query Response TTL of the secondary DNS server is not being respected 
by the 9971/9951/8831 phones if the primary server goes down after the first 
request.
Resolution Summary
CSCut29536 for 8831 IP Phones
CSCut29519 for 99xx IP Phones
CSCut75647 for 8811/51 IP phones

So versions and deeper dive required.

-Wes


On Apr 5, 2018, at 3:14 PM, Russell Goings 
> wrote:

Hi all
   I have a client who was doing some network mucking around and part of that 
maintenance caused the two DNS servers used by a group of IP phones to go 
unavailable.  Some of these phones went unregistered with a reason code 13 - 
TCP KeepAlive Timeout.

What I (OK not me.  Just the client.  Well maybe I am interested as well) would 
like to know is how do the Cisco SIP phones, particularly the 8851 model, deal 
with the TTL from the DNS response.

Does the phone honor the DNS TTL and cache the response for 24 hours?

Does the phone have a different cache time for the DNS response?

Does it make a new DNS query every time it sends a keep-alive?

I have never looked into this before and on this Thursday afternoon my 
Google-fu is weak.

And the incident was over a week ago so not much from the phone logs to help.

Thanks

Russell Goings (AM)
IPT Engineer – Dimension Data
Direct Line 571-203-4021
russell.goi...@dimensiondata.com
Monday – Friday 8:00 – 5:00 EST

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Re: [cisco-voip] Automated PSTN ingress call regression testing?

[Wes Sisk (wsisk)]

I have seen Hammer used for inbound call verification before:

http://www.empirix.com/products/hammer/

-Wes

On Mar 8, 2018, at 3:13 PM, Nick Barnett 
> wrote:

Thanks. I am aware of the multiple carriers... but I think being able to cover 
the edge services would be a huge help in this situation. I'm just anticipating 
what they will ask for. I kind of cast a wide net with this email because I was 
not sure what kind of services were out there. If some provider offered a 
prepackaged, automated testing service that featured multiple carrier numbers, 
I'd buy it in an instant. I just need to remember baby steps :)

By "prone to issues", I just mean that it will test connected calls, but 
getting that next layer of "it connected, but is it working" would be 
difficult. Not necessarily "issues". I suppose "obstacles" would have been a 
better word.  The issues I see with the freePBX would be similar, but also 
includes perimeter security and things of that nature.

I don't have UCCX, but I'm fairly ok at cobbling together AXL and JTAPI to do 
some stuff... maybe I'll just start there since it's basically free.

On Wed, Mar 7, 2018 at 9:33 PM, Anthony Holloway 
> wrote:
Even if you do the Free IP PBX or Twilio API, you're only calling from one 
carrier.  In the scenario you described, you mentioned:

"Verizon wireless customers cannot call Sprint toll free numbers from area code 
555"

Which is very specific.  Would you imagine that you would have owned a 555 
number on Verizon to have caught that scenario faster?  What if the area code 
was 666?  Or the originating carrier was AT?  The different combinations you 
would have to account for are very high.

If you only care about your edge service and inward, and not far end carriers, 
then a Twilio API app sounds like a good plan.  Heck, you could even just write 
a UCCX script to call out and back in via tromboning off the PSTN.

I'm curious, what did you mean by "prone to issues," when referring to the API?

On Wed, Mar 7, 2018 at 1:57 PM Nick Barnett 
> wrote:
A client has a need for an off site solution that will make test calls to their 
numbers and report when there are issues. I understand that this is very vague, 
but they are interested in hearing about any and all solutions.

They have several SIP carriers and a nationwide presence, but the SIP trunking 
is centralized. They've had enough issues with one DID service failing and 
their customers having to report the issue. Ideally, the SIP providers would be 
able to automatically do "something" when they stop receiving options pings, or 
when a certain sip response is received... but it doesn't work that way with 
the behemoth phone companies.

The way it works now is that MOST issues are able to be caught successfully 
with internal monitoring... but others such certain NPA-NXX can't call another 
NPA-NXX, or carrier interconnects such as "Verizon wireless customers cannot 
call Sprint toll free numbers from area code 555"  These odd scenarios are what 
we are looking to solve. I understand this is potentially huge, but I think if 
we could automate calls to about 10 different numbers, that would cover enough 
of the ingress and carrier combinations that it would make a HUGE difference.

I've thought of spinning up an Asterisk and somehow automating the echo test 
feature. I've also thought about using the Twilio API to test if calls are 
successful. Both of these are complicated and prone to issues... so if there is 
a hosted or cloud solution that is already available, please let me know.

Any suggestions or more than welcome.

Thanks,
Nick
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Re: [cisco-voip] untraceable connection attempt?

[Wes Sisk (wsisk)]

+1. I have seen syn scan or TCP half open cause alerts with no ip, no mac.

you can get some insight if this happening using the workaround for
CSCsw73304CLI show open ports to show ports in SYN_RECV

-wes

On Dec 20, 2017, at 7:47 AM, Dave Goodwin 
> wrote:

Any chance there’s an active vulnerability scanning machine on the network? 
With SYN scanning (half-open scans), it only sends a SYN packet to each port 
and never fully opens a TCP connection. I’m wondering whether this scenario 
might cause CallManager to report this incomplete registration alarm while not 
reporting the source IP - since the TCP connection was never considered to be 
established.

I’d like to try for myself a SYN scan of port 2000 using nmap to see if I can 
produce this alarm.

On Wed, Dec 20, 2017 at 12:25 AM Lelio Fulgenzi 
> wrote:

Also, definitely not exceeded number of registered devices. Especially not on 
the node where this alarm was coming from.

Sent from my iPhone

On Dec 20, 2017, at 12:01 AM, Ryan Huff 
> wrote:

Yeah it’s tough for sure, because the error is from the device failing to 
register, before providing any identifying information about itself ... so next 
to impossible to find from the mothership point of view.

You haven’t by chance exceeded the
“Maximum Number of Registered Devices” threshold for that node have you (CM 
Service Parameter)? You’d likely have other alarms if you did though.

If it’s a small cluster scenario where you can reasonably access all the phones 
and access switches; I’d do a registration audit.

Could be as simple as a non-Cisco sip device that got plugged into a access 
port with the admin vlan and tried to use CUCM as its registrar but failed 
miserably.

I’m guessing that isn’t your scenario; my thoughts, if it were me, would be to 
clear it and see if it comes back. Very possible that it’s an innocuous event 
that just sent some packets at the wrong time :).

Thanks,

Ryan

On Dec 19, 2017, at 11:39 PM, Lelio Fulgenzi 
> wrote:


First time I think I've ever seen this. Especially with no MAC or IP addr.

Only one alert.

But we've recently started allowing Jabber connections from our data VLANS.

I'd hate for it to be the beginning of something larger.

Sent from my iPhone

On Dec 19, 2017, at 11:35 PM, Ryan Huff 
> wrote:

Could also be network connectivity among a lot of things but more often than 
not, bouncing CM service seems to fix if this is a recurring alarm. If it’s a 
one time alarm you’ve not seen before; likely legitimately referring to a 
device.

If you’ve recently added any new devices, check network connectivity / verify 
they are all registered. Could also be a bad device that is no longer working 
but still attempting a registration ... sort of.

-Ryan

On Dec 19, 2017, at 11:22 PM, Ryan Huff 
> wrote:

Sounds like you should schedule a bounce of the CM service for this node.

Have a read here for more detail: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/err_msgs/8_x/ccmalarms851.html

Thanks,

Ryan

On Dec 19, 2017, at 11:11 PM, Lelio Fulgenzi 
> wrote:

An endpoint attempted to register but did not complete registration
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Re: [cisco-voip] New install 11.5 phones won't register or the subscriber

[Wes Sisk (wsisk)]

I’ll add a hard learned lesson here:

Example:
Subscriber sends a frame >1500 bytes. IF the frame makes it through the network 
it will by dropped by the NIC on the publisher if it exceeds the MTU configured 
on the publisher. You will not see the frame with ‘utils network capture…’ on 
the publisher as that dumps frames from the processor/linux kernel and not from 
the NIC. The frame is dropped at the NIC before it reaches the kernel.

-Wes

On Oct 12, 2017, at 2:11 PM, Dave Goodwin 
> wrote:

While I haven't tried to test this specifically, my understanding is that a 
network transport that fully supports jumbo frames (e.g. 9000 bytes) will cause 
no issues with hosts configured with 1500 byte MTU trying to communicate. I 
thought the issues can happen when the configured MTU in the two hosts 
communicating exceeds the effective transport MTU anywhere in the path between 
the hosts.

On Thu, Oct 12, 2017 at 1:48 PM, Mike King 
> wrote:


On Wed, Oct 11, 2017 at 4:10 AM, Ryan Huff 
> wrote:

- Make sure you’re accounting for any non-standard MTU between the publisher 
and subscriber nodes. By default, CUCM will use 1500 if you don’t change it 
(which is the typical network standard). For example; If you have jumbo frame 
support in the network path between the two nodes and you set 1500 MTU on the 
publisher and subscriber, the subscriber installation will typically “fail node 
connectivity validation” with no visible clue as to why. This is because this 
is the first point in the subscriber installation where there is enough 
back/fourth TCP with the publisher to cause the TCP window to burst, which 
causes the TCP retransmission error. Since it happens at this point, CUCM 
assumes it’s because the publisher can’t verify the subscriber.



Hey Ryan,  Do you have links for that?  I find the behavior very odd, and not 
the way I understood standard MTU over a Jumbo enabled network to function.   
Or is this just a Cisco Caveat type deal?

Mike

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Re: [cisco-voip] Informacast SNMP error with UCM

[Wes Sisk (wsisk)]

Looks like the ultimate root cause for many was:

CSCtb70375SNMP needs to alert user of DNS connectivity issues

Symptom:
If  DNS is not configured or misconfigured the SNMP walk/get operation on that 
server  will fail with time out.

When an application queries CUCM via SNMP and DNS is not configured correctly, 
it will timeout and not respond as expected.  No administrator alert is sent to 
notify them of this timeout/issue.

Conditions:
CUCM with misconfigured DNS (or DNS server down/slow to respond)

Workaround:
Configure a working DNS server


This bug was (C)losed without a fix because it’s primarily a misconfig.

-w

On Jul 20, 2017, at 11:45 AM, Wes Sisk (wsisk) 
<ws...@cisco.com<mailto:ws...@cisco.com>> wrote:


From Syslog, we can see lot of "DBLException" with reason code 746 (on Pub and 
Sub)

From knowledge database, similar cases of "DBException with reason code 746" 
have been solved by cluster reboot or Subscriber re-build.


try ‘risdb query…’ commands from CLI on cucm. restart ccm process, or ideally 
whole cucm server.

-w

On Jul 20, 2017, at 11:32 AM, Matthew Loraditch 
<mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>> 
wrote:

We are getting a genErr all of a sudden when trying to do phone refreshes. 
String is correct and tried using SNMPv3 also.
Have restarted InformaCast and restarted SNMP Agent on UCM.

Anything else shy of a reboot followed by a possible TAC Case?

Capture shows this:


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

Facebook<https://www.facebook.com/heliontech?ref=hl> | 
Twitter<https://twitter.com/HelionTech> | 
LinkedIn<https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> 
| G+<https://plus.google.com/+Heliontechnologies/posts>

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Re: [cisco-voip] Informacast SNMP error with UCM

[Wes Sisk (wsisk)]

From Syslog, we can see lot of "DBLException" with reason code 746 (on Pub and 
Sub)

From knowledge database, similar cases of "DBException with reason code 746" 
have been solved by cluster reboot or Subscriber re-build.


try ‘risdb query…’ commands from CLI on cucm. restart ccm process, or ideally 
whole cucm server.

-w

On Jul 20, 2017, at 11:32 AM, Matthew Loraditch 
> 
wrote:

We are getting a genErr all of a sudden when trying to do phone refreshes. 
String is correct and tried using SNMPv3 also.
Have restarted InformaCast and restarted SNMP Agent on UCM.

Anything else shy of a reboot followed by a possible TAC Case?

Capture shows this:


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

Facebook | 
Twitter | 
LinkedIn 
| G+

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Re: [cisco-voip] License File

[Wes Sisk (wsisk)]

What particular features do you like about it (use)?

-w

On May 8, 2017, at 6:38 AM, Ben Amick 
> wrote:

Can confirm, reissued my CCX license myself in 2 minutes through that page just 
the other day.

Ben Amick
Telecom Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
Meade
Sent: Sunday, May 07, 2017 10:04 PM
To: James Buchanan >
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] License File

You can do most of this stuff yourself now at 
cisco.com/go/licensing.

On Sat, May 6, 2017 at 5:49 PM, James Buchanan 
> wrote:
Hello,
I just had a license file reissued in five minutes. If you need a license, 
right now might be just the right moment.
Thanks,
James

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Re: [cisco-voip] Maximum Call Duartion

[Wes Sisk (wsisk)]

That is indeed enforced globally.

There is no way to enforce this natively in CUCM.  If you have any type of 
firewall or stateful inspection between CUCM and the gateway and using H.323 
then you might look at terminating sessions. You might be able to do an 
automation with EEM on the gateway.

-Wes

On Jan 18, 2017, at 12:00 PM, Asim Mekki Basheer 
> wrote:


Hello


 we have CUCM 7.5 connected to outside With E1 PRI is there any way to set 
Timer to force disconnect outgoing Calls after 30 Minutes , the only way i 
found is :

CCM go to:

CCMAdmin--> Service --> Service Parameter --> Cisco CallManager -Max call 
duration

but this will be applied Globally .

Regards


ASIM


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Re: [cisco-voip] CSCui62452

[Wes Sisk (wsisk)]

Potentially, yes.

Feature does not get reset in situation of ‘no sip’ … ‘sip’. If 
allow-connections is not enabled then connections not allowed.

-w

On Nov 18, 2016, at 1:01 PM, Scott Voll 
> wrote:

Would this affect phones not registering?

CSCui62452

Scott

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Re: [cisco-voip] Traffic Issues with 7900 Series Phones

[Wes Sisk (wsisk)]

Adam,

Are you using dot1x? There are some interesting things in that space.

Otherwise, maybe get 9.4.2es3 to pickup the fix for
CSCuq883257965 7945 excessive core files cause phone stability problems


-w


On Nov 15, 2016, at 8:50 AM, Pawlowski, Adam 
<aj...@buffalo.edu<mailto:aj...@buffalo.edu>> wrote:

All,

We’re still looking at this with TAC, though the initial response was that the 
7941, 7961, etc done with hardware and software support. There was an 
announcement on October 20th that said software maintenance ended immediately 
(oops). Our timers and such are ubiquitous across our network, all defaults, 
and we don’t have this problem elsewhere. I went with looking for MAC change 
traps and didn’t run into anything, going down that road. The phones don’t log 
any VLAN changes either in their logs.  The phones are going out of service for 
UCM Closed TCP or UCM Reset TCP, and we see what looks like the UCM not 
responding back with the proper SCCP KeepAliveAck, which causes the phone to 
sort of do nothing for 60 seconds. By then, since both the phone is waiting 
60.0 seconds and the UCM is as well to hear from it, the connection is reset 
and closed.

Phones that are not sharing the data VLAN have been fine, but, we cannot 
implement that across this entire area due to the needed cabling, switchports, 
etc.

In another location we have these phones going what appears to be high CPU – 
the latency on the phone goes way up, with ICMP response, the response of the 
phone to buttons and actions, and the call suffers from high jitter and broken 
conversations. Oddly enough, when we cap with SPAN enabled on the phone, the 
data looks fine going through it. Power cycling the phone clears this 
temporarily.

Everyone thus far has wanted to go down the road of loss somewhere on the 
network, but, as we continue to take captures, we see the conversation complete 
at the UCM, and beyond the phone via “SPAN to PC port”, or at it with SPAN at 
the edge – the phone application itself is simply not responding in a timely 
manner, at least based on initial observation.

Given the earlier response that these devices are now done with support, this 
does not bode well, but, we are still looking.

Regards,

Adam Pawlowski
SUNYAB NCS

From: Wes Sisk (wsisk) [mailto:ws...@cisco.com]
Sent: Tuesday, November 08, 2016 12:50 PM
To: Pawlowski, Adam
Cc: Tommy Schlotterer; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Traffic Issues with 7900 Series Phones

Not much visibility into L1/L2 on those phones; drop counters on the webpage or 
phone UI is about all you get.

Are the phones randomly unregistering? This is good baseline: 
https://supportforums.cisco.com/document/52176/understanding-sccp-phone-unregistration-and-failover-networks-perspective

If some sort of frame issue, correct, not many options.

What are the nature of messages being retransmitted?
Also, anything interesting looking in the log files?

One age old odd one is CDP timers out of sync btwn phone and switch. Phone 
keeps IP but gets dumped into data vlan.  Your choice on how to approach that.

One possibility: If phones are unregistering then check the lastoutofservice 
reason on the phone, in the CM traces, or in the RTMT reports if you’re on a 
new enough version. I *think* we got these phones fixed to say “vlan change’ or 
‘cdp timeout’ or ‘ip change’ something like that if there were changes in the 
network interface.

Alternatively take a few phones stick them in a port that not trunked but in 
the voice vlan… do these exhibit the same problem?

next ‘heuristic’ guess after that is possibly arp cache refresh on the switch. 
have seen several issues where arp cache timeout was set low, switch re-arp for 
many devices concurrently, arp response dropped by input queue overflow and 
input queue drop. net result the switch ‘forgets’ which port that phone is on.

So…. where do the packets/frames EXIST and NOT EXIST in the network?

-Wes

On Nov 4, 2016, at 4:32 PM, Pawlowski, Adam 
<aj...@buffalo.edu<mailto:aj...@buffalo.edu>> wrote:

Wes,

Thanks, that's good to know about ICMP. We've seen phones that get into a state 
where they reply with response times all over the board, lossy, which, 
Reset/Restart from the UCM does not rectify. Powering the device down does 
clear the condition - the set is otherwise idle. I need to get into one of 
those via SSH and pull the CPU to see if it is up at that time, to see if 
there's an identifiable process that covers this.

We did get some captures from in front of the firewall where the UCM resides, 
and from a monitor session from the switch out at the edge where the phone is 
connected. We can see the UCM sending re-transmissions to the phone, and the 
phone eventually replying some time later. Unless there is a reason for us to 
try and get a copper tap on the segment between the switch and the phone, then, 
it would seem to be that 

Re: [cisco-voip] Traffic Issues with 7900 Series Phones

[Wes Sisk (wsisk)]

Not much visibility into L1/L2 on those phones; drop counters on the webpage or 
phone UI is about all you get.

Are the phones randomly unregistering? This is good baseline: 
https://supportforums.cisco.com/document/52176/understanding-sccp-phone-unregistration-and-failover-networks-perspective

If some sort of frame issue, correct, not many options.

What are the nature of messages being retransmitted?
Also, anything interesting looking in the log files?

One age old odd one is CDP timers out of sync btwn phone and switch. Phone 
keeps IP but gets dumped into data vlan.  Your choice on how to approach that.

One possibility: If phones are unregistering then check the lastoutofservice 
reason on the phone, in the CM traces, or in the RTMT reports if you’re on a 
new enough version. I *think* we got these phones fixed to say “vlan change’ or 
‘cdp timeout’ or ‘ip change’ something like that if there were changes in the 
network interface.

Alternatively take a few phones stick them in a port that not trunked but in 
the voice vlan… do these exhibit the same problem?

next ‘heuristic’ guess after that is possibly arp cache refresh on the switch. 
have seen several issues where arp cache timeout was set low, switch re-arp for 
many devices concurrently, arp response dropped by input queue overflow and 
input queue drop. net result the switch ‘forgets’ which port that phone is on.

So…. where do the packets/frames EXIST and NOT EXIST in the network?

-Wes

On Nov 4, 2016, at 4:32 PM, Pawlowski, Adam 
<aj...@buffalo.edu<mailto:aj...@buffalo.edu>> wrote:

Wes,

Thanks, that's good to know about ICMP. We've seen phones that get into a state 
where they reply with response times all over the board, lossy, which, 
Reset/Restart from the UCM does not rectify. Powering the device down does 
clear the condition - the set is otherwise idle. I need to get into one of 
those via SSH and pull the CPU to see if it is up at that time, to see if 
there's an identifiable process that covers this.

We did get some captures from in front of the firewall where the UCM resides, 
and from a monitor session from the switch out at the edge where the phone is 
connected. We can see the UCM sending re-transmissions to the phone, and the 
phone eventually replying some time later. Unless there is a reason for us to 
try and get a copper tap on the segment between the switch and the phone, then, 
it would seem to be that there is some reason the phone is not replying to the 
UCM. There is nothing behind the phone, or any output buffer drops. Our delay 
here in reply is in some number of seconds, so I don't believe there's any 
buffering involved that would be to that extent.

What I fear is that if we get to a point where we can determine there is some 
frame that is an issue, these devices are past the point of any patching being 
done as of a few weeks ago. But, since replacing phones is not free and 
takes a bunch of time, I still have to come up with something. I only saw a bug 
for large sized ICMPv6 with nothing particularly helpful in the wording and the 
workaround of "don't do that" so I'm not hopeful.

We have our AM and SE aware of what is going on, and they've offered to help, 
so I'm hopeful we can eventually confirm the reason we're having trouble, even 
if we can't directly fix it.


Adam

-Original Message-
From: Wes Sisk (wsisk) [mailto:ws...@cisco.com]
Sent: Friday, November 04, 2016 12:52 PM
To: Pawlowski, Adam
Cc: Tommy Schlotterer; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Traffic Issues with 7900 Series Phones

Phones process ICMP traffic with low priority and throttling. This was
implemented to stem DoS attempts. Consider looking more at Voice Quality
effects, retransmits in packet captures, or parsing CCM traces for round
trip times. As you state these phones are relatively late in life and
therefore relatively stable.

-Wes


On Nov 2, 2016, at 2:42 PM, Pawlowski, Adam 
<aj...@buffalo.edu<mailto:aj...@buffalo.edu>> wrote:

Tommy,

Sorry about that. These are a mixed bag. 41/61 both G and G-GE
phones, with the gigabit ones primarily. Some SCCP, some SIP, mostly
9.4.2SR1-1, but seen on 9.4.2SR2-2. PC attached or not, no difference, the
only difference we've been able to create that stops this, is changing the
data VLAN that runs through the phone to a different one, or something
null (with no PC).

Adam

-Original Message-
From: Tommy Schlotterer [mailto:tschlotte...@presidio.com]
Sent: Wednesday, November 02, 2016 2:37 PM
To: Pawlowski, Adam; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: RE: Traffic Issues with 7900 Series Phones

What specific Models of phones eg. 41s/61s? or 40s/60s?

Thanks

Tommy

Tommy Schlotterer | Systems Engineer
Presidio | www.presidio.com<http://www.presidio.com>
20 N. Saint Clair, 3rd Floor, Toledo, OH 43604
D: 419.214.1415 | C: 419.70

Re: [cisco-voip] Traffic Issues with 7900 Series Phones

[Wes Sisk (wsisk)]

Phones process ICMP traffic with low priority and throttling. This was 
implemented to stem DoS attempts. Consider looking more at Voice Quality 
effects, retransmits in packet captures, or parsing CCM traces for round trip 
times. As you state these phones are relatively late in life and therefore 
relatively stable.

-Wes


On Nov 2, 2016, at 2:42 PM, Pawlowski, Adam  wrote:

Tommy,

Sorry about that. These are a mixed bag. 41/61 both G and G-GE phones, 
with the gigabit ones primarily. Some SCCP, some SIP, mostly 9.4.2SR1-1, but 
seen on 9.4.2SR2-2. PC attached or not, no difference, the only difference 
we've been able to create that stops this, is changing the data VLAN that runs 
through the phone to a different one, or something null (with no PC).

Adam

> -Original Message-
> From: Tommy Schlotterer [mailto:tschlotte...@presidio.com]
> Sent: Wednesday, November 02, 2016 2:37 PM
> To: Pawlowski, Adam; cisco-voip@puck.nether.net
> Subject: RE: Traffic Issues with 7900 Series Phones
> 
> What specific Models of phones eg. 41s/61s? or 40s/60s?
> 
> Thanks
> 
> Tommy
> 
> Tommy Schlotterer | Systems Engineer
> Presidio | www.presidio.com
> 20 N. Saint Clair, 3rd Floor, Toledo, OH 43604
> D: 419.214.1415 | C: 419.706.0259 | tschlotte...@presidio.com
> 
> -Original Message-
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Pawlowski, Adam
> Sent: Wednesday, November 02, 2016 2:23 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Traffic Issues with 7900 Series Phones
> 
> After much hair pulling and frustration, I wanted to ask the group here in
> case anyone has seen this or has any thought on what we should be looking
> for.
> 
> We have a number of 7900 series phones that have been exhibiting issues
> that appear to me to be that the phone is getting hung up on something.
> Some sort of frame or packet is screwing with the network chip/board or
> the OS which is causing it trouble. I see missed traffic, missed
> responses, high ICMP echo times - and phones that eventually get stuck
> with their ICMP echo response times being all over the board - with some
> report of call trouble and CMR showing crazy jitter. If I power cycle the
> phone that clears and it works fine for a while.
> 
> I realize these items are pretty much end of useful life, pretty much all
> done with software support, and are going to drop off of the compatibility
> matrix and probably functional support in the near future. But, while we
> still have a ton of them - has anyone noted any particular type of traffic
> that causes the 7900 series phones grief?
> 
> I don't have loss on the network, there do not seem to be any transient
> broadcast storms rolling by. We do see an increased amount of mDNS, IPv6
> (phones are v4 only) etc, but nothing stands out as causing a particular
> problem. It just seems that whatever this is, is causing a memory leak or
> something, wherein it gets bad enough that things go to hell eventually.
> 
> Any thoughts?
> 
> Adam P
> SUNYAB
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Re: [cisco-voip] Applie i)S 11.7.1 stops after iOS upgrade?

[Wes Sisk (wsisk)]

https://help.webex.com/docs/DOC-10317

?

-w

On Oct 4, 2016, at 3:14 PM, Jason Aarons (AM)  
wrote:



Has anyone received any cases where Cisco Jabber version11.7.1 stops working 
after upgrading their iPhone or iPad  iOS from version 9 to 10? Or any other 
Apple related cases?


Jason Aarons, CCIEx2 No 38564
Consultant
Dimension Data
904-338-3245 mobile

Planned PTO:
Oct 7 - Oct 17






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Re: [cisco-voip] Phone Fraud H.323

[Wes Sisk (wsisk)]

depending on how far you want to go:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsgd-712-cm/fsxfer.html#wp1043824

-w

On Sep 13, 2016, at 12:54 PM, Erick Bergquist 
> wrote:

Yep, seen that scenario a few dozen times with unity.

Restriction tables and lock down CSS on CUCM side if unity does not
need to make external calls. Use a separate CSS for Unity (don't
re-use the LDCss, etc on voicemail ports/trunk).

COBRA also doesn't include restriction tables, so those need to be put
back in if you use COBRA to migrate to other server.

I wish they would add option to class of service to not allow users to
enable alternate transfer method. No good unity report to see when a
user changed transfer method either.

Erick





On Tue, Sep 13, 2016 at 6:11 AM, David Zhars 
> wrote:
The main problem, as Ryan pointed out, is more than likely weak voicemail
passwords.  Hackers are able to dial your main number and get an automated
greeting, when they press (asterisk) they get the "welcome to voicemail"
prompt.  From there, it's pretty easy to start inputting extensions
(especially if any are published on your website) and guessing passwords.
Once they have that, they can input call forwarding details when someone
receives a message, and just start calling that extension all the time.  I
have definitely seen THAT scenario before.

Dave

On Mon, Sep 12, 2016 at 10:39 PM, Lelio Fulgenzi 
> wrote:

Oh, we definitely have dial-peers. Both inbound and outbound.

I'm concerned because of the earlier comment about not all DIDs being
accounted for.

I'm pretty sure I have an "inward dial" config on each PRI. But not sure I
have a num-exp for each.

I'll double check my configs and share.

Sent from my iPhone

On Sep 12, 2016, at 10:11 PM, Nick Britt 
> wrote:

Do a

Sh run all | sec dial-p

If you don't have any DP's in the config I would imagine you are OK.

On Monday, 12 September 2016, Lelio Fulgenzi 
> wrote:


Here's a question:

We're using PRIs w/ MGCP so I'm assuming we're not affected. However, we
have SRST configured, which I believe uses H323.

Could this affect us as well?

Lelio

Sent from my iPhone

On Sep 11, 2016, at 8:46 PM, Lelio Fulgenzi 
> wrote:

+1 here. By default with (the older?) IOS if someone dialled a number
associated with the line plugged into your router, you'd get dial tone and
from there you could dial an number the dial plan allowed.



Sent from my iPhone

On Sep 11, 2016, at 11:49 AM, Nick Britt 
>
wrote:

Hi David,

Can I ask Which version of IOS you are using?

Also could you post your incoming dial peer configuration or are you just
using the default DP 0?

Ive experienced a similar issue before (luckily I didn't configure this
particular deployment)

Before IOS 15 (I believe) direct in ward dial was not applied to the
default dial peer. This allows people to call in on an unnnallocated number
with in the DID range and receive a dial tone. (Check it out quite scary)

The resolution was to apply the command direct in wars dial to all
incoming dial peers.

I will try and dig out the link from Cisco.



On Sunday, 11 September 2016, David Zhars 
> wrote:

So yesterday I was alerted by our landline company that some of our
phone numbers that come in POTS on an H323 router, we being used for phone
fraud.  I am wondering how this happens with an H323 router (I am familiar
with someone hacking Unity and setting up actions to route to Jamaica once
someone leaves a voicemail or similar).

The odd part is that these numbers are almost NEVER used for calling
out, unless the user presses a 7 for an outbound line (versus an 8 which
puts the call out on ISDN).

I found a link on how to disable OffNet calling in UCM, but should I
instead look at securing the H323 router?  Or does the call blocking rule
need to be done in UCM?

Thanks for any enlightenment you can provide.

PS- Client is in USA, call fraud to Jamaica which does not require a
country code, so harder to block.



--
- Nick

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--
- Nick


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Re: [cisco-voip] FXO port to Viking C-1000B door entry controller

[Wes Sisk (wsisk)]

IMHO this is where it gets interesting. We have to leave the ‘digital’ world 
and enter the ‘analog’ world. The “FXO” emits a signal, it travels across the 
wire, and might possibly be echoed back at the electrical level. Get the 
o-scopes ladies and gents. When the firmware of the FXO card reports this:

30278353: Jun 14 14:09:32.746: htsp_process_event: [0/1/1, FXOLS_CONNECT, 
E_DSP_SIG_0100]fxols_normal_battery
30278354: Jun 14 14:09:32.747: htsp_timer_stop2 fxols_disc_confirm

30278358: Jun 14 14:09:32.747: 
//3699561/56508068916F/VTSP:(0/1/1):-1:1:1/vtsp_process_event:
   [state:S_CONNECT, event:E_TSP_DISCONNECT_IND]

I suspect there is something *interesting* going on at the analog level. We 
start getting into pseudo-manchester-encoding territory here. Just how much 
“signal” is required to assume a 1 vs a 0?

Think of it as someone with ‘super hearing’ vs. someone that might have handled 
one too many chain saws or attended one too many metal concerts.

Given the disconnect occurs when you press any (assumption on my part) DTMF, 
then I wonder what analog signal is reflected back down the wire at the analog 
level. Reflection is a common byproduct of impedance mismatch. This is part of 
the reason high impedance interfaces are so valuable almost universally. This 
is where I buy my friend with an electrical degree a few beverages of his 
choice. 
http://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/64282-impedance-choice.html

How much knowledge or control do you have over all aspects of the physical 
implementation?  24km of 26AWG copper pair can have ‘interesting’ electrical 
properties, let alone what ‘actually’ happens over that 24km.

Impedance (AC resistance at various frequencies) may not be the same as 
Resistance (DC resistance). https://en.wikipedia.org/wiki/Electrical_impedance 
. This gets into the physical details ,and sometimes anomalies, of the 
environment.

Hat tip Ryan for brining up the i(impedance) vs r(resistance) angle.

-w

On Jul 1, 2016, at 11:52 AM, Ryan Huff 
> wrote:

Only after the local-end generates an oscillated sine wave (in this case, a 
DTMF event), the call is somehow terminated.

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Re: [cisco-voip] FXO port to Viking C-1000B door entry controller

[Wes Sisk (wsisk)]

spot on. right up until they invented analog answering machines… then we *HAD* 
to have analog disconnect supervision.

-w

On Jun 30, 2016, at 4:25 PM, Norton, Mike 
> wrote:

A phone company cannot magically reach into your house and put your POTS phone 
on-hook. Only you - the person physically holding the POTS phone - can put it 
on-hook. *This is true regardless of which side initially started the 
phonecall.*


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Re: [cisco-voip] FXO port to Viking C-1000B door entry controller

[Wes Sisk (wsisk)]

alternatively looks like the FXO port might be configured to to tone disconnect 
supervision. try disabling that all together and work forward from there. IIRC 
(dis)connect supervision is implemented in firmware on the card and is only 
sent back to IOS as an asserted event. Alternatively stated: mapping the cable 
conditions to the disconnect event is implemented in the module firmware, can 
be affected by configuration, but is only reported to IOS as a discrete event.  
Alternatively stated it’s up to the card firmware to determine 40V or 60V 
equals “1” but the card only returns “1” to IOS. The 40..60 is still 
configurable in IOS and is downloaded to (programs/configures) the card on 
initialization.

-w


On Jun 29, 2016, at 6:30 PM, Ryan Huff 
> wrote:

On the Cisco FXO port side, try "impedance 600r" under the voice port config.

Essentially, what I believe you are up against, is a sine oscillation 
disagreement between the FXO port and the Viking.

Thanks,

Ryan

On Jun 29, 2016, at 5:55 PM, Damisch, Kevin 
> wrote:

Customer has a 4431 router with an FXO card connected to a Viking C-1000B door 
entry controller.  The normal operation is to press the button on the doorbox, 
connection plar on the FXO port sends the call to an IP phone, the user 
answers, IP phone user presses ** which is the code to unlock the door.  The 
problem is with the mid-call DTMF.  As soon as *any* DTMF digit is pressed 
mid-call, we get a disconnect on the FXO port.  We don’t even have a chance to 
press the * key a second time.  The doorboxes connected to this C-1000B 
controller requires us to use an FXO port, not an FXS port.

Ran the vtsp/vpm debugs.  TAC is saying it is something with the Viking unit 
causing the disconnect, which is what they confirmed here:

30278353: Jun 14 14:09:32.746: htsp_process_event: [0/1/1, FXOLS_CONNECT, 
E_DSP_SIG_0100]fxols_normal_battery
30278354: Jun 14 14:09:32.747: htsp_timer_stop2 fxols_disc_confirm

30278358: Jun 14 14:09:32.747: 
//3699561/56508068916F/VTSP:(0/1/1):-1:1:1/vtsp_process_event:
   [state:S_CONNECT, event:E_TSP_DISCONNECT_IND]

We also did a PCM capture (will they EVER let us have the tool to decode 
it???), TAC sent me the extracted wav file from the capture and we clearly hear 
the correct DTMF * key.  We have ringing, two way audio, then at about 6.7 
seconds in the screenshot, you can see the DTMF in the waveform below:



We took it one step further and verified the 2 frequencies of * DTMF digit (942 
Hz x 1209 Hz) in the audio from the FXO port capture, which looks and sounds 
good:


It’s a 2-port card and we tried the other port, but get the same thing.  Tried 
MGCP and H323, same thing.  No, we have not tried another FXO card yet since 
the call works, we have two-way audio to the Viking doorbox.  It’s just that 
the mid-call DTMF is causing a disconnect, which appears to be coming from the 
Viking unit.  From Cisco’s perspective, the call setup is there and we are 
sending the * DTMF digit out during the call.

So, we call Viking support.  We did some tests with an analog phone/buttset.  
Sure enough, it works just fine.  You press the doorbox button, rings the 
phone, answer it, two-way audio, press **, and the door unlocks and disconnects.

Connect it back up to the FXO port, press the doorbox button, answer it, 
two-way audio, but as soon as any DTMF digit is pressed, the call disconnects, 
and the door doesn’t unlock.  The customer had the Viking C-1000A on an Avaya, 
we did a migration to Cisco UC, and now it doesn’t work.  They got a newer 
Viking C-1000B unit to see if that would help, but we have the same problem.  
TAC says it is Viking’s issue.  Viking says it is something on the Cisco side.

If anyone has setup one of these up on a FXO port and got it to work, please 
let me know what you ended up doing.  There are quite a few forum posts I found 
online, but no resolution was found or nobody followed up with what they did to 
get it to work.

Thanks!
Kevin

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Re: [cisco-voip] Error Opening Log File Telepresence Content Server 6.2.1 for Windows Server 2008 R2 SP1

[Wes Sisk (wsisk)]

Looks like "make sure that you have the secondary drive on which all the data 
will
be saved is with drive letter as E: and not something else as D: or F:. You can 
rename it
under Control Panel>Administrative Tools>Computer management.”

-Wes

On Jan 19, 2016, at 7:15 PM, Foryanto J. Wiguna  
wrote:

Dear Guys,

i have a problem when i try to install Telepresence Content Server in
Windows Server 2008 R2 SP1

Run S6_2_1.exe to install the VM Content Server software on the
appliance.  then this notification error occur...


"error opening installation log file. verify that the specified log
file location exists and is writable"

what directory should i check in my windows server 2008 R2 SP1 ?

Many Thanks
-- 
Salam,
-Foryanto J. Wiguna-
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Re: [cisco-voip] called party hears music on hold

[Wes Sisk (wsisk)]

I tracked one of these down once several years back:

a calls b
a puts call on hold
b transfers call to c
c answers and hears MoH

when a, b, and c are all phones registered on the same node/cluster this 
shouldn’t happen. However, if the call from a to b traverses a trunk 
(SIP/H.323/MGCP) then there is no “hold” state/communication/update in the 
signaling stream to prevent the transfer from b to c.

Just a thought.

-w

On Nov 5, 2015, at 10:57 AM, Reto Gassmann  wrote:

Hello Group

we have a strange problem on our UCM 10.5, that sometime happens.

If someone calls my IP Phone (7961) and I pick up the call, I hear music on 
hold. The caller then puts the call on hold and then gets back to the call. Now 
music on hold is gone and we can talk.

Has anyone had this issue before?

Regards Reto 
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Re: [cisco-voip] CUCM Web Services API

[Wes Sisk (wsisk)]

This is what Contact center tries to do via CTI.
-w

On Sep 16, 2015, at 8:46 AM, Mark Holloway  wrote:

Does anyone know if it’s possible to query CUCM over HTTP/HTTPS to see if a 
user is available to take a call? (ie. is the phone registered, is DND on or 
off, are all lines occupied and the user is busy)



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Re: [cisco-voip] 3905 dial tone problem

[Wes Sisk (wsisk)]

phones download the tone files as part of the locale. could be a bad file 
served up from the CME router. Could be the file was downloaded and corrupted 
in phone storage.

suggest changing locale, reset phone, try again, then reset locale back.

alternatively, hard reset phone.
alternatively register the phone to a “known good” CM and CME and let it 
re-download files.

-Wes

On Aug 20, 2015, at 6:41 PM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:


My thoughts ...

Maybe switch to the inactive load and test as a quick-fix.

Is this the only sip endpoint in this CME that is impacted? Are there SCCP 
endpoints and if so, do they have the same issue?

Can customer complete a dial and does the issue continue once connected?

Thanks,

Ryan


 Original Message 
From: Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com
Sent: Thursday, August 20, 2015 06:29 PM
To: cisco-voip 
(cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net) 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] 3905 dial tone problem

3905 registered to CME, when you pick up a headset, the dial tone begins.  But 
it seems to skip like an old record, it buzzes and pauses.

Not finding a bug, customer didn’t tell me load or CME version etc.

I’m thinking this is a load problem as you haven’t even dialed yet.



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Re: [cisco-voip] glibc/ghost vulnerability

[Wes Sisk (wsisk)]

The update that happened on the 20th was an internal system update. Basically a 
change happened on a case that was linked to the bug. This tickled the 
‘last-update’ date of the bug.

As far as fixed versions - I’ll look to Ryan on how/when UCCX populates 
Integrated-releases field.

-Wes

On Jul 27, 2015, at 9:39 AM, Charles Goldsmith 
wo...@justfamily.orgmailto:wo...@justfamily.org wrote:

Ryan/Wes, one last followup question, 
https://tools.cisco.com/bugsearch/bug/CSCus68524 shows that it was updated on 
the 20th, but I don't see a change, other than it may say fixed now (don't 
remember before), but it does not show what changed.

Also, of note, since it does say it's fixed, there are 0 fixed versions out.  
Can we get some clarification on it?

Thanks


On Fri, Jul 10, 2015 at 5:57 PM, Ryan LaFountain (rlafount) 
rlafo...@cisco.commailto:rlafo...@cisco.com wrote:
To add to what Wes said:

If you have other UCC products that run on VOS (Finesse, SocialMiner, 
MediaSense, CUIC) you'll see further differences between underlying VOS 
versions between them, UCCX and CUCM. This causes not only a lot of confusion 
in tracking bug fixes in the platform between products but delay in integrating 
fixes like these as Wes has described below.

We are working to address this. The first part is in better tracking of bug 
fixes and security issues in the platform and between products. The second part 
is moving to a common underlying platform version and build process for most 
UCC products. This will greatly speed up our fix inclusion and standardize the 
underlying VOS version in many of our applications leading to greater 
consistency and stability. Without exposing too much more, we should see this 
common VOS in UCC system release 11.0.

HTH.

Thank you,

Ryan LaFountain
Unified Contact Center
Cisco Services
Direct: +1 919 392 9898tel:%2B1%20919%20392%209898
Hours: M - F 9:00am - 5:00pm Eastern Time

From: cisco-voip on behalf of Charles Goldsmith
Date: Friday, July 10, 2015 at 5:21 PM
To: Wes Sisk (wsisk)
Cc: voip puck
Subject: Re: [cisco-voip] glibc/ghost vulnerability

Gotcha, thanks for the explanation Wes, that's what I was looking for and can 
explain it to the customer.  I'll let the customer know of the risks and let 
them make the decision to upgrade or wait for a minor patch.

Thanks!

On Fri, Jul 10, 2015 at 1:58 PM, Wes Sisk (wsisk) 
ws...@cisco.commailto:ws...@cisco.com wrote:
I’ll lead off with: UCCX does a fair amount of work to customize the VOS 
platform to their needs. As such they don’t pull in updates and fixes as fast 
as UCM, UC, and CUP.

I bet if you check the kernel or RHEL version you will find significant 
difference and that contributes to the complexity of the fix.
admin:show packages active kernel
Active Side Package(s): for kernel package(s)
kernel-firmware-2.6.32-431.20.3.el6.noarch
kernel-2.6.32-431.20.3.el6.x86_64
platform-kernel-tunable-1.0.0.0-1.i386
dracut-kernel-004-336.el6_5.1.noarch

RyanL may weigh in with better details.

-w

On Jul 10, 2015, at 11:41 AM, Charles Goldsmith 
wo...@justfamily.orgmailto:wo...@justfamily.org wrote:

I understand that CUCM and UCCX are both VOS, and that it's probably not the 
same version, but I don't understand why the platform team for CUCM can give us 
a minor patch but we can't get the same out of UCCX.

I'm sure most of you are like me, and steer clear of .0 releases.  There is an 
old saying, dot Oh, oh no.

I'm not comfortable advising a customer to upgrade to the 11.0 release.

Would like thoughts on this, and some explanation of the differences of the VOS 
between CUCM/CUC and UCCX.




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Re: [cisco-voip] glibc/ghost vulnerability

[Wes Sisk (wsisk)]

I’ll lead off with: UCCX does a fair amount of work to customize the VOS 
platform to their needs. As such they don’t pull in updates and fixes as fast 
as UCM, UC, and CUP.

I bet if you check the kernel or RHEL version you will find significant 
difference and that contributes to the complexity of the fix.
admin:show packages active kernel
Active Side Package(s): for kernel package(s)
kernel-firmware-2.6.32-431.20.3.el6.noarch
kernel-2.6.32-431.20.3.el6.x86_64
platform-kernel-tunable-1.0.0.0-1.i386
dracut-kernel-004-336.el6_5.1.noarch

RyanL may weigh in with better details.

-w

On Jul 10, 2015, at 11:41 AM, Charles Goldsmith 
wo...@justfamily.orgmailto:wo...@justfamily.org wrote:

I understand that CUCM and UCCX are both VOS, and that it's probably not the 
same version, but I don't understand why the platform team for CUCM can give us 
a minor patch but we can't get the same out of UCCX.

I'm sure most of you are like me, and steer clear of .0 releases.  There is an 
old saying, dot Oh, oh no.

I'm not comfortable advising a customer to upgrade to the 11.0 release.

Would like thoughts on this, and some explanation of the differences of the VOS 
between CUCM/CUC and UCCX.

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Re: [cisco-voip] Strange Number

[Wes Sisk (wsisk)]

Thanks for pointing this out! I’ll get the doc updated.

The rest of the list:

**##*1,  // analysis trace on/off
**##*2,   // terse/verbose toggle
**##*3,   // dump digit discarding instructions
**##*4,   // dump patterns
**##*5,   // dump partitions list
**##*6,   // dump css list
**##*7,   // dump feautre css content list
**##*8,   // dump device css context list
**##*9,   // dump line appearance css context list
**##*10,  // dump line css context list
**##*11,  // dump multiple partitions list
**##*12, // dump VM Profile list
**##*13, // dump VM Pilot number list
**##*14  // dump regions list  dump in memory db


Forward Manager

**##*30 // Enable Debug
**##*31 // Dump Intercept Table
**##*32 // Dump Active Call Table
**##*33 // Dump Pickup Table
**##*34 // Refresh Intercept Table
**##*35 // Clear Callforward Loop Prevention Tables

Pickup Manager

**##**30 // Enable Debug
**##**31 // Print the Pickup Group
**##**32 // Print the Pickup Member DNs
**##**33 // Print the Pickup monitored alerting calls
**##**34 // Print the active pickups
**##**35 // Print the active pickup monitorings

Many cobwebs on this one.. background:

Which Trains:
3.3 main train supports (enable code, dump intercept table, clear fwd loops)
3.3 ES/SR ... supports (enable code, dump intercept table)
3.4 supports (enable code, dump intercept table)
4.0 supports all codes

This was implemented with CSCed29942, CSCdy87797.

-Wes

On Jun 25, 2015, at 11:59 AM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

I know it’s not an exact match but it’s very similar to a dialing forest dump 
procedure off a SCCP handset:

https://supportforums.cisco.com/document/97751/how-dump-ccm-memoryimdb-diagnostic-information-traces

I personally haven’t used the “**##**32” you mentioned though.

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Pawlowski, Adam
Sent: Thursday, June 25, 2015 11:44 AM
To: 'cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net'
Subject: [cisco-voip] Strange Number

Does anyone here know what “**##**32” is supposed to be? It seems I can dial 
this but it is not actually … anything ? It just clears out and goes away.

This is from SCCP sets anyways to UCM.

Adam

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Re: [cisco-voip] Strange Number

[Wes Sisk (wsisk)]

From 10.5.2:

10.5.2.1-5

00710403.002 |13:32:04.919 |AppInfo  ||DialingForest=
{
 |DigitForest=Dialing
 {
  |Partition=
  |TreeType=TREE_TYPE_DIGIT
  |Patterns=
  {
   |(**##*01)  }
  |(**##*02) }
 |(**##*03)}
|(**##*04)}
|(**##*05)}
|(**##*06)}
|(**##*07)}
|(**##*08)}
|(**##*09)}
|(**##*10)}
|(**##*11)}
|(**##*12)}
|(**##*13)}
|(**##*14)}
|(**##*15)}
|(**##*16)}
|(**##*30) (Intercept)}
|(**##*31) (Intercept)}
|(**##*32) (Intercept)}
|(**##*35) (Intercept)}
|(**##**30) (Intercept)}
|(**##**31) (Intercept)}
|(**##**32) (Intercept)}
|(**##**33) (Intercept)}
|(**##**34) (Intercept)}
|(**##**35) (Intercept)}


Forwarding:
DEBUG_PATTERN_ENABLE[] = **##*30;
DEBUG_PATTERN_DUMP_INTERCEPT_TABLE[] = **##*31;
DEBUG_PATTERN_DUMP_ACTIVE_CALL_TABLE[] = **##*32;
DEBUG_PATTERN_CLEAR_FORWARD_LOOP_PREVENTION_TABLES[] = **##*35;

Pickup:
enableDebug = **##**30;
printPickupGroupNumberTable = **##**31;
printPickupMemberDNTable = **##**32;
printMonitoredAlertingCallTable = **##**33;
printActivePickupTable = **##**34;
printActivePickupMonitoringTable = **##**35;


On Jun 25, 2015, at 3:10 PM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:

HA! Thanks for this Wes!


From: ws...@cisco.commailto:ws...@cisco.com
To: dpa...@fidelus.commailto:dpa...@fidelus.com
Date: Thu, 25 Jun 2015 17:45:44 +
Subject: Re: [cisco-voip] Strange Number
CC: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net; 
aj...@buffalo.edumailto:aj...@buffalo.edu

Thanks for pointing this out! I’ll get the doc updated.

The rest of the list:

**##*1,  // analysis trace on/off
**##*2,   // terse/verbose toggle
**##*3,   // dump digit discarding instructions
**##*4,   // dump patterns
**##*5,   // dump partitions list
**##*6,   // dump css list
**##*7,   // dump feautre css content list
**##*8,   // dump device css context list
**##*9,   // dump line appearance css context list
**##*10,  // dump line css context list
**##*11,  // dump multiple partitions list
**##*12, // dump VM Profile list
**##*13, // dump VM Pilot number list
**##*14  // dump regions list  dump in memory db


Forward Manager

**##*30 // Enable Debug
**##*31 // Dump Intercept Table
**##*32 // Dump Active Call Table
**##*33 // Dump Pickup Table
**##*34 // Refresh Intercept Table
**##*35 // Clear Callforward Loop Prevention Tables

Pickup Manager

**##**30 // Enable Debug
**##**31 // Print the Pickup Group
**##**32 // Print the Pickup Member DNs
**##**33 // Print the Pickup monitored alerting calls
**##**34 // Print the active pickups
**##**35 // Print the active pickup monitorings

Many cobwebs on this one.. background:

Which Trains:
3.3 main train supports (enable code, dump intercept table, clear fwd loops)
3.3 ES/SR ... supports (enable code, dump intercept table)
3.4 supports (enable code, dump intercept table)
4.0 supports all codes

This was implemented with CSCed29942, CSCdy87797.

-Wes

On Jun 25, 2015, at 11:59 AM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

I know it’s not an exact match but it’s very similar to a dialing forest dump 
procedure off a SCCP handset:

https://supportforums.cisco.com/document/97751/how-dump-ccm-memoryimdb-diagnostic-information-traces

I personally haven’t used the “**##**32” you mentioned though.

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Pawlowski, Adam
Sent: Thursday, June 25, 2015 11:44 AM
To: 'cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net'
Subject: [cisco-voip] Strange Number

Does anyone here know what “**##**32” is supposed to be? It seems I can dial 
this but it is not actually … anything ? It just clears out and goes away.

This is from SCCP sets anyways to UCM.

Adam

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Re: [cisco-voip] isdn channels full message

[Wes Sisk (wsisk)]

not tested, theoretically sound, YMMV. No guarantees, warranties, etc.

use a route group with top down hunting. your real gateways are the first 
entries.
last entry is an h323 gateway. the destination IP is the local CM. This is 
essentially a loopback going out/in the IP interface.
Use the CSS for that gateway to match translation patterns that overwrite the 
called party number to be the DN that forwards to Voicemail. Also use a 
different egress CSS for the TP.
Setup your VMail system to answer based on that redirect number and deliver 
whatever message you want.

-w

On Jun 23, 2015, at 9:15 AM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:

Ah okay 

Well, an easy way would be MGCP Gateways and route groups (and just mux the 
timers to get quick failover). With H.323 however, that is a bit more tricky 
because CCM knows nothing about the gateway or PRI's state, so if you could do 
anything, it would have to be in the gateway.

You can use hunt groups in an h.323 IOS gateway to do this with multiple PRI's; 
you may be able to do this by having one of the peers in the hunt group kick 
the call back into CCM to a destination that forwards to voicemail.

Thanks,

Ryan



From: abba...@gmail.commailto:abba...@gmail.com
To: ryanh...@outlook.commailto:ryanh...@outlook.com; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] isdn channels full message
Date: Tue, 23 Jun 2015 13:55:42 +0100

Hi just to add it to.



We need it only for the outbound direction.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 23 June 2015 13:51
To: abbas wali; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] isdn channels full message



I don't believe so, at least not on the CPE side.

The presence of all channels being busy would indicate that there aren't any 
channels available for the next call (or the next call after all channels are 
busy) to ring through the PRI for you to forward to voicemail, even if there 
were a way for you to act upon channel utilization.

If you are trying to use conditional routing based on full PRI capacity, you'd 
probably have to work with your carrier and see if they can forward upon full 
channel utilization (they have the capability to do that, but will they is the 
question). However, they wouldn't be able to forward through the utilized PRI 
in that scenario, they'd have to forward to an alternate number.

Thanks,

Ryan

From: abba...@gmail.commailto:abba...@gmail.com
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Date: Tue, 23 Jun 2015 11:38:55 +0100
Subject: [cisco-voip] isdn channels full message
Hi  folks,



CUCM 9 and H323 gateways with ISDN channels to PSTN. We want to redirect to VM 
when all the channels are full.



Can CUCM do it or Dpeer?



Thanks



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Re: [cisco-voip] Interesting off hook dial delay

[Wes Sisk (wsisk)]

yeah, we’ve tried to improve DNA several times to make it more accurate. 
something about it being “difficult” ;)

-w

On Jun 10, 2015, at 2:29 PM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:

Thanks to all who replied. I hate asking inter-digit questions because they are 
so basic BUT can be a real pain in the arse if it isn't transparent.

The issue was an all wildcard RP that ended with ! in the same PT as all the 
DNs. DNA didn't match on it because as Wes mentions, there were no numerical 
matches between that pattern and the called number that DNA used for analysis.

Thank you Brian, Wes and everyone else who replied. Ya'll will get a JibJab 
E-Card at Christmas from me.

Thanks,

-R


From: ws...@cisco.commailto:ws...@cisco.com
To: bmead...@vt.edumailto:bmead...@vt.edu
CC: ryanh...@outlook.commailto:ryanh...@outlook.com; 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Interesting off hook dial delay
Date: Wed, 10 Jun 2015 17:39:53 +

yup.

DNA is a close approximation, but in specific corner cases it deviates from 
dalib

I put my chips on an unexpected overlapping pattern

I troubleshot one of these long ago that turned out to be the intercept pattern 
for Cisco Messaging Interface (CMI for SMDI integration). When you consider 
“every dial able number” it is pretty encompassing.

Also, dump the verbose dialing forest and go to town looking around.

-w

On Jun 10, 2015, at 10:50 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

DNA won't show the overlapping pattern.  Set the Digit Analysis Complexity 
CallManager Service Parameter to TranslationAndAlternatePatternAnalysis on 
each node then pull CallManager traces for a test call.  The Digit Analysis 
section in the traces will now show the overlapping pattern.

On Wed, Jun 10, 2015 at 8:48 AM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:
Experiencing some strange off-hook delay when dialing.

Here are the tests I did:

When dialing an exact match (another DN) from the off-hook position, I get a 
full T302 delay before the digits are sent. If however, I mark the target DN as 
urgent priority, it routes immediately (as I would expect).

I've done some additional testing and placed 2 single DNs in a new PT/CSS and I 
get the same off-hook delay. If I dial the digits and send EnBloc with the dial 
softkey, everything works fine.

When trying to transfer a connected call to a third caller, the secondary 
transfer button and the target line description do not appear until the delay 
passes and the phone rings

I've used DNA and it isn't showing me anything unexpected; save this was a 
perfectly working system a few days ago.

I've tried a cluster reboot, and that didn't change the condition. It sounds 
like signaling delay except for the fact the I can override the issue with 
Urgent Priority, if it were signaling delay nothing would override it. So I am 
left with inter-digit timeout but can't see where it could be coming from, 
especially with a new PT/CSS that only has 2 DNs in it.

Thanks,

Ryan

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Re: [cisco-voip] Interesting off hook dial delay

[Wes Sisk (wsisk)]

yup.

DNA is a close approximation, but in specific corner cases it deviates from 
dalib

I put my chips on an unexpected overlapping pattern

I troubleshot one of these long ago that turned out to be the intercept pattern 
for Cisco Messaging Interface (CMI for SMDI integration). When you consider 
“every dial able number” it is pretty encompassing.

Also, dump the verbose dialing forest and go to town looking around.

-w

On Jun 10, 2015, at 10:50 AM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

DNA won't show the overlapping pattern.  Set the Digit Analysis Complexity 
CallManager Service Parameter to TranslationAndAlternatePatternAnalysis on 
each node then pull CallManager traces for a test call.  The Digit Analysis 
section in the traces will now show the overlapping pattern.

On Wed, Jun 10, 2015 at 8:48 AM, Ryan Huff 
ryanh...@outlook.commailto:ryanh...@outlook.com wrote:
Experiencing some strange off-hook delay when dialing.

Here are the tests I did:

When dialing an exact match (another DN) from the off-hook position, I get a 
full T302 delay before the digits are sent. If however, I mark the target DN as 
urgent priority, it routes immediately (as I would expect).

I've done some additional testing and placed 2 single DNs in a new PT/CSS and I 
get the same off-hook delay. If I dial the digits and send EnBloc with the dial 
softkey, everything works fine.

When trying to transfer a connected call to a third caller, the secondary 
transfer button and the target line description do not appear until the delay 
passes and the phone rings

I've used DNA and it isn't showing me anything unexpected; save this was a 
perfectly working system a few days ago.

I've tried a cluster reboot, and that didn't change the condition. It sounds 
like signaling delay except for the fact the I can override the issue with 
Urgent Priority, if it were signaling delay nothing would override it. So I am 
left with inter-digit timeout but can't see where it could be coming from, 
especially with a new PT/CSS that only has 2 DNs in it.

Thanks,

Ryan

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Re: [cisco-voip] Lead Second

[Wes Sisk (wsisk)]

Good call out Andrew.

adding a quote so that this might register a little more…

When the leap second update occurs it is possible for the kernel to hang or 
halt.” Kernel offline means all services/applications/processes offline. This 
is a little important.

-w


On Jun 8, 2015, at 8:53 PM, Andrew Grech 
agrec...@gmail.commailto:agrec...@gmail.com wrote:

Hi Guys,

As an advisory please check here for the up and coming leap second. Some UC 
versions are affected


http://www.cisco.com/web/about/doing_business/leap-second.html#~ProductInformation
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Re: [cisco-voip] MGCP CCM TFTP Download issue with snmp-server-chassis-id

[Wes Sisk (wsisk)]

Looks like others have found this as well. No indication if this is a problem 
or working as designed.

I’m still looking for external facing documentation.

-Wes

On Jun 5, 2015, at 3:05 PM, Erick Bergquist erick...@gmail.com wrote:

Has anyone seen the snmp-chassis-server-id interfere with CCM-manager
TFTP download?

I can't find a document explaining snmp-server-chassis-id setting and
MGCP behavior but it appears MGCP uses the snmp-serrver-chassis for
the TFTP download and not the MGCP hostname. Not finding anything on
google searches either.

For example, if my MGCP domain name is router1.domain.com and the
snmp-server-chassis-id doesn't match,  set to RouterA then CCM-manager
is trying to TFTP file for endpoint RouterA.

In debug ccm-manager configuration-download we see this,

: Unable to read file
tftp://10.1.1.10/aaln_s0_su...@routera.domain.com.cnf.xml, rc=-2

It's easy to fix but more curious if this is documented or if  anyone
else has seen this?

Erick
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Re: [cisco-voip] IMP Deployment Guide?

[Wes Sisk (wsisk)]

my first guess:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/presence.html

?

On May 29, 2015, at 4:07 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.commailto:avholloway+cisco-v...@gmail.com wrote:

I'm reading through all of the available IMP documentation I can find and I 
keep seeing references to the following:

Deployment Guide for IM and Presence Service on Cisco Unified Communications 
Manager

However, none of the references are hyperlinks, and a Google search returns no 
hits with that title.  Does anyone know if the document name has changed, or 
what's going on here?

Thanks.
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Re: [cisco-voip] Conf Phone?

[Wes Sisk (wsisk)]

i’m not aware of a phone with a workable mic profile. i did some work with 
cardioid mics in another life and those seem to fit the bill. a bit of a 
stretch but what about having multiple soft clients with cardioid mic inputs? 
one iPhone or iPad per conference with attached mic

-w

On May 28, 2015, at 9:39 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com 
wrote:

Have an exec that uses a room that can have multiple conferences at one time on 
different phones.  He would like the nearest microphone on the conference phone 
to be the only one that pics up sounds to mitigate some of the background noise 
of the other calls in the room. 

Aware of any 3rd party products?

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Re: [cisco-voip] MGCP Odd issue

[Wes Sisk (wsisk)]

Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Facility i = 0x9F8B0100A1110201010201008009485546462C5259414E
Protocol Profile =  Networking Extensions
0xA1110201010201008009485546462C5259414E
Component = Invoke component
Invoke Id = 1
Operation = CallingName
Name Presentation Allowed Extended
Name = HOWSER,BARRY
Calling Party Number i = 0x2180, 'DN-INTENTIONALLY-REMOVED'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, 'DN-INTENTIONALLY-REMOVED'
Plan:ISDN, Type:National
May 27 03:21:11.700: ISDN Se0/1/0:23 Q931: TX - CALL_PROC pd = 8  callref = 
0x805D
May 27 03:21:11.700: ISDN Se0/1/0:23 Q931: TX - ALERTING pd = 8  callref = 
0x805D
Progress Ind i = 0x8088 - In-band info or appropriate now available
May 27 03:21:11.716: ISDN Se0/1/0:23 Q931: RX - RELEASE_COMP pd = 8  callref = 
0x005D
Cause i = 0x82E018 - Mandatory information element missing
May 27 03:21:11.716: ISDN Se0/1/0:23 Q931: RX - RELEASE pd = 8  callref = 
0x005D
Cause i = 0x82D1 - Invalid call reference value

On Tue, May 26, 2015 at 7:12 PM, Dave Goodwin 
dave.good...@december.netmailto:dave.good...@december.net wrote:
Barry, if you have the q931 debug from when the error occurred, and if you are 
able to share it, that may help shed light on the error. The mandatory IE 
missing issue is an ISDN protocol error where the CUCM and telco switch are in 
disagreement about something. It is sometimes possible to determine which IE is 
missing from the debug of the entire failed call.

TAC may be able to provide help as well, if you can provide that debug for them.

On Tue, May 26, 2015 at 3:53 PM, Barry Howser 
bhowser5...@gmail.commailto:bhowser5...@gmail.com wrote:
Hi Wes. The mandatory missing IE message was at the end of a q931 debug right 
before the call goes busy. I may have over simplified my original explanation. 
I have several gateways that this exact same scenario happened to. All 
experienced the same condition, with the same configurations.

On Tue, May 26, 2015 at 3:33 PM, Wes Sisk (wsisk) 
ws...@cisco.commailto:ws...@cisco.com wrote:
a couple things here -

you say MGCP.. if using MGCP and d-channel bachaul then it is CCM’s ISDN stack 
in use. Where did you see the error “mandatory IE missing?” if it was with 
debugs on the gateway then it may have been generated by the gateway’s ISDN 
stack.

each isdn ‘switch type’ has subtle nuances in implementation. the right answer 
really depends on what physical equipment the telco is using as well as how 
they have the d-ch provisioned on their end.

it could be the telco changed config. or they might have upgraded the switch. 
or you may have started using a different call flow that added/removed IE’s.

also possible that a lingering reset/restart was not applied on the UCM side 
(CSCtw80866Reset Required flag in CCMAdmin for any device/trunk that has 
been )

-w

On May 26, 2015, at 2:44 PM, Barry Howser 
bhowser5...@gmail.commailto:bhowser5...@gmail.com wrote:

So I've had an MGCP/T1 gateway up and running with CCM, happy as a clam for 
several weeks.

Then all of the sudden today it stopped passing inbound communication. Egress 
works just fine, but ingress rings once then a fast busy.

In the ISDN logs I get mandatory information element missing.

I am using; EF, BZ8S, Primary-ni (which is telco settings). Again everything 
WAS fine. After some research I found that error to mean that the CCM side 
kicked the call back to the gateway because it didn't get everything it needed 
in the header.

A proposed suggestion was to use a different switch-type. So in the 
CCM/Gateway/PRI config page, I changed the switch type to PRI-4ESS - 
Saved/Applied/Reset (then restarted mgcp on the gateway) and presto, ingress is 
now working.

If I reverse the process and go back to the Primary-ni in the CCM/Gateway/PRI 
config, I get the same problem with ingress again.

Can anyone explain this to me? Does it sound like my telco changed something? 
Seems like something with MGCP is goofed right? Is this something that a telco 
would just arbitrarily change?

Thanks
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Re: [cisco-voip] MGCP Odd issue

[Wes Sisk (wsisk)]

a couple things here -

you say MGCP.. if using MGCP and d-channel bachaul then it is CCM’s ISDN stack 
in use. Where did you see the error “mandatory IE missing?” if it was with 
debugs on the gateway then it may have been generated by the gateway’s ISDN 
stack.

each isdn ‘switch type’ has subtle nuances in implementation. the right answer 
really depends on what physical equipment the telco is using as well as how 
they have the d-ch provisioned on their end.

it could be the telco changed config. or they might have upgraded the switch. 
or you may have started using a different call flow that added/removed IE’s.

also possible that a lingering reset/restart was not applied on the UCM side 
(CSCtw80866Reset Required flag in CCMAdmin for any device/trunk that has 
been )

-w

On May 26, 2015, at 2:44 PM, Barry Howser bhowser5...@gmail.com wrote:

So I've had an MGCP/T1 gateway up and running with CCM, happy as a clam for 
several weeks. 

Then all of the sudden today it stopped passing inbound communication. Egress 
works just fine, but ingress rings once then a fast busy.

In the ISDN logs I get mandatory information element missing.

I am using; EF, BZ8S, Primary-ni (which is telco settings). Again everything 
WAS fine. After some research I found that error to mean that the CCM side 
kicked the call back to the gateway because it didn't get everything it needed 
in the header.

A proposed suggestion was to use a different switch-type. So in the 
CCM/Gateway/PRI config page, I changed the switch type to PRI-4ESS - 
Saved/Applied/Reset (then restarted mgcp on the gateway) and presto, ingress is 
now working.

If I reverse the process and go back to the Primary-ni in the CCM/Gateway/PRI 
config, I get the same problem with ingress again.

Can anyone explain this to me? Does it sound like my telco changed something? 
Seems like something with MGCP is goofed right? Is this something that a telco 
would just arbitrarily change?

Thanks
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Re: [cisco-voip] Phones not showing registered on subscriber server, 10.5.1

[Wes Sisk (wsisk)]

RIS reads a shared memory segment populated by the ccm process. If no server in 
the cluster can see registration status when phones are registered to that sub 
then likely the ccm process on that node has stopped updating shared memory on 
that node. you can verify this with ‘utils query risdb’ from CLI on the sub.

once that happens there’s really no way to know why/how it happened. recovery 
is to restart the ccm process to reinitialize shared memory.

-Wes

On May 4, 2015, at 4:59 PM, Erick Bergquist erick...@gmail.com wrote:

Anyone have any thoughts as to why a subscriber server would not show
phones registered? Everything is working fine.

Just shows none, but if I force a phone to register to this server it
shows registered fine and works but then stays as Unregistered when
phone registers back to it's main server.

I've restarted the server and the RIS Data Collector service on all the nodes.

The database is synced and replicated fine cluster wide, and the utils
diagnose test passes all tests.

The version is 10.5.1.1

Regards, Erick
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Re: [cisco-voip] Number of Partitions in CSS max 1024 chars?

[Wes Sisk (wsisk)]

Checking, but not AFAIK. You might follow

CSCuc24135Increase the 1024 characters limitation in the partition list for 
a CSS

-Wes

On Apr 20, 2015, at 8:59 AM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:

Anyone familiar with planned changes to number of Partitions in a CSS in 10x or 
11x?

The maximum length of the combined CSS clause (device and pattern) comprises 
1024 characters that includes separator characters between partition names (for 
example, partition 1:partition 2:partition 3).

I have a customer needing more than 1024 characters in CUCM 10x and assume it’s 
still 1024 characters.

-jason


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Re: [cisco-voip] Cisco TAC Per Incident fee

[Wes Sisk (wsisk)]

Looks like it varies based on product. Typical Hardware/Software/Warranty 
discussion still applies. Software support that becomes hardware replacement 
may be up charge (re-entitlement)… etc.

-Wes

On Dec 17, 2014, at 3:50 PM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:

Thanks for the replies everyone. If anyone has any more information, or links, 
that would be great!

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

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


From: Brian Meade bmead...@vt.edumailto:bmead...@vt.edu
To: Lelio Fulgenzi le...@uoguelph.camailto:le...@uoguelph.ca
Cc: cisco-voip cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Sent: Wednesday, December 17, 2014 3:27:33 PM
Subject: Re: [cisco-voip] Cisco TAC Per Incident fee

I believe it depends on the technology.

On Wed, Dec 17, 2014 at 2:55 PM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:

Just wondering if there is any documentation as to how much a per incident fee 
would be with Cisco. Anyone have any ideas?

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

519‐824‐4120 Ext 56354tel:519%E2%80%90824%E2%80%904120%20Ext%2056354
le...@uoguelph.camailto:le...@uoguelph.ca
www.uoguelph.ca/ccshttp://www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


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Re: [cisco-voip] H245Interface Related Processes

[Wes Sisk (wsisk)]

Hmm,

These should help:
CSCsm26337need clear messages for h225 tcp session failure

In versions with fix h225 session setup is clearly indicated in cm sdi traces
with one of these messages:
H225Cdpc(%07d)::requestConnect_TcpStartSessionErr: H225 Tcp Start Session
request failed

H225 Call is aborted as no response received for H225 Tcp Start Session Request
within configured H225TcpTimer value

h225 session abort due to RST or FIN is clearly indicated in cm sdi traces with:
H225 Tcp session terminated abnormally


With fix for this defect TCP session aborts will be reported by one of the 
following statements in CallManager SDI trace:
requestConnect_TcpStartSessionErr: H225 Tcp Start Session request failed
requestConnect_H225TcpTimer: H225 Call is aborted as no response received for 
H225 Tcp Start Session Request within configured H225TcpTimer value
call_initiated1_TcpStopSessionInd: H225 Tcp session terminated abnormally
overlap_sending2_TcpStopSessionInd: H225 Tcp session terminated abnormally
outgoing_call_proceeding3_TcpStopSessionInd: H225 Tcp session terminated 
abnormally
call_delivered4_TcpStopSessionInd: H225 Tcp session terminated abnormally
call_present6_TcpStopSessionInd: H225 Tcp session terminated abnormally
incoming_call_proceeding9_TcpStopSessionInd: H225 Tcp session terminated 
abnormally
active10_TcpStopSessionInd: H225 Tcp session terminated abnormally
await_ann_complete_TcpStopSessionInd: H225 Tcp session terminated abnormally
active10a_TcpStopSessionInd: H225 Tcp session terminated abnormally
active10b_TcpStopSessionInd: H225 Tcp session terminated abnormally
GKRasARQH225Setup_TcpStopSessionInd: H225 Tcp session terminated abnormally
GKRasCcSetupRequestConnect_TcpStartSessionErr(%d, %d): TcpStartSessionErr from 
IP=%s
GKRasCcSetupRequestConnect_TcpStartSessionErr: H225 Tcp Start Session request 
failed
GKRasCcSetupRequestConnect_H225TcpTimer: H225 Call is aborted as no response 
received for H225 Tcp Start Session Request within configured H225TcpTimer value
overlap_receiving25_TcpStopSessionInd: H225 Tcp session terminated abnormally
wait_for_disconn_kluge_TcpStopSessionInd: H225 Tcp session terminated abnormally
paused_TcpStopSessionInd: H225 Tcp session terminated abnormally



CSCsm26355need clear messages for h245 tcp session failure
h.245 tcp session setup failure:
TCP ERROR: H245ListenReq or H245ConnectReq failure, or received SdlCloseInd
from H245Handler, Perform cleanup of H245 Session

established h.245 session experiences TCP keepalive timeout:
TranslateAndTransport(%d)::wait_SdlCloseInd - ERROR: H245 signaling connection
aborted

established h.245 session receives unexpected TCP FIN or RST:
TranslateAndTransport(%d)::wait_SdlCloseInd - ERROR: H245 signaling connection
aborted




Otherwise, consider enabling additional SDL trace flags like “enable network *” 
and “enable SDL TCP event trace”.  However, these will cause significant 
additional trace lines (one or more SDL signal per TCP segment received).


I believe one of the additional traces above will get you insight to socket 
requests down to the network layer in UCM.

-Wes



On Dec 1, 2014, at 5:12 PM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

Folks:

Hoping to get some insight on SDL process creation for H245…

Scenario is three CUCM clusters communicating over ICTs. Call is routed from 
Cluster-1 to Cluster-2… then Cluster-2 to Cluster-3. Cluster-3 sends the H245 
address  port info via H225 ALERTING to Cluster-2, which then sends its own to 
Cluster-1. Issue is Cluster-1 never establishes the H245 session with 
Cluster-2. The H245 address and port is received on Cluster-1 but no H245 
processes are being created for the MSD/TCS exchange. According to SDL traces 
on Cluster-2, the latest state of H245 on the node *sending* the ALERTING 
message is “waitForTransportEstablishment”. On Cluster-1, the H245Interface 
process is never created according to SDL traces, so we never even reach the 
opportunity for TCS media caps exchange. MXTimeout occurs shortly after.

Question is… For a node receiving an H245 address  port info via H225 (the 
calling cluster…), is creation of the H245Interface and/or related H245 process 
dependent on CUCM *first* establishing the new, 2nd TCP socket with the remote 
H.323 endpoint that advertised the H.245 port. In other words, at an SDL level, 
is H245Interface created only after the 2nd TCP session is successfully 
established at the transport level for H245 TCP communication? Knowing this 
would help me assess the likelihood of the issue being related to issues at the 
TCP level.

Thanks!

- Dan
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Re: [cisco-voip] RTMT not able to collect trace files in 10.0.1

[Wes Sisk (wsisk)]

The components displayed on the first page are defined by an XML file stored on 
the UCM server. Likely that file is missing or corrupt. Any unexpected power 
outages or filesystem issues?

Filenames on the server are documented in the footnote here:
https://supportforums.cisco.com/document/65651/communications-manager-rtmt-trace-locations-cli

-Wes

On Nov 14, 2014, at 1:39 PM, Erick Bergquist 
erick...@gmail.commailto:erick...@gmail.com wrote:

Anyone seen RTMT not show the first screen on Collect Traces function?

It does not show the screen for which components to select to gather
traces for and shows a hour glass for a second or two and goes right
to the second window where you set the path, zip, and duration you
want to collect for.  The next button on this screen does nothing as
well.

Remote Browse does not work either

CUCM version 10.0.1.1

Having to download files manually via SFTP is taking longer.
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Re: [cisco-voip] Remove lines on phones with BAT

[Wes Sisk (wsisk)]

You can reuse DN’s in unique partitions. What about just a garbage partition 
that every phone auto registers into? Give them a CSS that can only dial the 
local help desk and 911.

Poing being- when you use a discrete partition for it you can use or reuse any 
DN’s without creating conflicts. None of your CTI apps or CSS would need to 
work with this partition.

-Wes

On Oct 30, 2014, at 9:22 AM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:

OK I get the picture...

So if I'm going to put a line on all phones, can I put a shared line on all 
1300 phones?  If so, how does CER work?  will the PSAP call back the line or 
does it call back the phone?

I have a dialing plan that doesn't allow for much in the way of non DID numbers 
so have a very limited amount to use.

TIA

Scott


On Wed, Oct 29, 2014 at 2:17 PM, Wes Sisk (wsisk) 
ws...@cisco.commailto:ws...@cisco.com wrote:
Agreed with Ryan, Anthony, and Erick.

Give the phones DN’s, even just internal DN’s. It is “burning a DN” in the 
sense that number is allocated. That number does not have to be a DID. Give the 
phones/lines a CSS that PLAR’s to a help desk. Or at least allow them to invoke 
local emergency services - private or public.

-Wes

On Oct 29, 2014, at 3:13 PM, Erick Wellnitz 
ewellnitzv...@gmail.commailto:ewellnitzv...@gmail.com wrote:

Not having lines on a physical phone can be a career altering decision if 
someone needs to dial emergency services and they grab the nearest logged out 
phone.  There could also be legal issues but I'm not a lawyer so take that for 
what you will.

On Wed, Oct 29, 2014 at 1:03 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.commailto:avholloway+cisco-v...@gmail.com wrote:
That's a Yes for my experiences.  The logout profile or default state of the 
phone has a nonDID in most cases, DID in a few cases (with a CSS restriction 
for Internal and EMS), and the UDP of the user has a DID in most cases, nonDID 
in a few cases (with typical CSS).

On Wed, Oct 29, 2014 at 1:33 PM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:
So what are people doing in a Extension Mobility only environment?  Burning two 
DN's?  one for the user and one for every phone?

scott




On Wed, Oct 29, 2014 at 11:11 AM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
Auto-reg will provision lines as well, and I agree with Anthony.
I don't think anyone wants to deal with the potential issue of somebody picking 
up a phone to dial emergency services and not getting dialtone.

-Ryan

On Oct 29, 2014, at 2:05 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.commailto:avholloway+cisco-v...@gmail.com wrote:

Scott,

SIP or SCCP, either way, for emergency reasons you don't want phones laying 
about, unable to save a life when people depend on the reliability of 911.

Here's one option without knowing your environment design or limitations:  
Setup Auto Registration and then Bulk Delete all the phones.

On Tue, Oct 28, 2014 at 1:38 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
All phones? SIP phones won't register without a line.

-Ryan

On Oct 28, 2014, at 1:16 PM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:

is there a way to bulk delete all lines on all phones?

we are moving to extension mobility and want to remove all the lines on all the 
phones

Thanks

scott

CM 8.6.2
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Re: [cisco-voip] Remove lines on phones with BAT

[Wes Sisk (wsisk)]

Agreed with Ryan, Anthony, and Erick.

Give the phones DN’s, even just internal DN’s. It is “burning a DN” in the 
sense that number is allocated. That number does not have to be a DID. Give the 
phones/lines a CSS that PLAR’s to a help desk. Or at least allow them to invoke 
local emergency services - private or public.

-Wes

On Oct 29, 2014, at 3:13 PM, Erick Wellnitz 
ewellnitzv...@gmail.commailto:ewellnitzv...@gmail.com wrote:

Not having lines on a physical phone can be a career altering decision if 
someone needs to dial emergency services and they grab the nearest logged out 
phone.  There could also be legal issues but I'm not a lawyer so take that for 
what you will.

On Wed, Oct 29, 2014 at 1:03 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.commailto:avholloway+cisco-v...@gmail.com wrote:
That's a Yes for my experiences.  The logout profile or default state of the 
phone has a nonDID in most cases, DID in a few cases (with a CSS restriction 
for Internal and EMS), and the UDP of the user has a DID in most cases, nonDID 
in a few cases (with typical CSS).

On Wed, Oct 29, 2014 at 1:33 PM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:
So what are people doing in a Extension Mobility only environment?  Burning two 
DN's?  one for the user and one for every phone?

scott




On Wed, Oct 29, 2014 at 11:11 AM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
Auto-reg will provision lines as well, and I agree with Anthony.
I don't think anyone wants to deal with the potential issue of somebody picking 
up a phone to dial emergency services and not getting dialtone.

-Ryan

On Oct 29, 2014, at 2:05 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.commailto:avholloway+cisco-v...@gmail.com wrote:

Scott,

SIP or SCCP, either way, for emergency reasons you don't want phones laying 
about, unable to save a life when people depend on the reliability of 911.

Here's one option without knowing your environment design or limitations:  
Setup Auto Registration and then Bulk Delete all the phones.

On Tue, Oct 28, 2014 at 1:38 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.commailto:rratl...@cisco.com wrote:
All phones? SIP phones won't register without a line.

-Ryan

On Oct 28, 2014, at 1:16 PM, Scott Voll 
svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote:

is there a way to bulk delete all lines on all phones?

we are moving to extension mobility and want to remove all the lines on all the 
phones

Thanks

scott

CM 8.6.2
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Re: [cisco-voip] Mobile Connect and CDR's

[Wes Sisk (wsisk)]

I see suggestions that the last redirecting number is the desk number.

-Wes



On Oct 23, 2014, at 10:04 AM, Ed Leatherman ealeather...@gmail.com wrote:

Hello!

I'm trying to figure out how we can identify call legs from Mobile Connect that 
are going out to long distance destination numbers. We have a number of IT 
staff with home or mobile numbers that are long distance, and would like to 
start offering them the capability to use mobile connect to facilitate on-call 
and other communications needs.

Unfortunately this also means I need some way to track utilization and provide 
bill-back information for internal billing. 

I can't find any way to identify the final called party number for the remote 
destination except that it appears to be embedded in OutgoingProtocolCallRef, 
but the docs just call this a Globally unique call reference, not sure I want 
to rely on it.

So far the best I can do to pick out these calls seems to be searching for 
records where the finalcalledpartypartition matches the unrestricted long 
distance partition name, and the originalcalledpartypartition matches our 
internal DN partition. Wish there was a cleaner way to report on these that 
included the remote destination number.

Does anyone know of a better way to do this? CUCM 9.1.

BTW I am using splunk right now to sift through the records - at some point we 
will probably have this as part of our real billing software though.

-- 
Ed Leatherman
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Re: [cisco-voip] CSCup71611 - ccx 10.5

[Wes Sisk (wsisk)]

That is closed as being unreproducible in other environments. The workaround 
was not documented. This bug is submitted for review and cleanup.

Best route to forward progress is a TAC case.

-Wes

On Oct 15, 2014, at 2:31 PM, Erick Wellnitz ewellnitzv...@gmail.com wrote:

Any of our Cisco people able to shed a little more light on the workaround for 
CSCup71611?
 
It's been terminated  but I appear to have run into it.  I thought I'd check 
before running the gauntlet of getting a client's contract associated to my 
account.
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Re: [cisco-voip] 3905 gives up on voice VLAN

[Wes Sisk (wsisk)]

Both should be visible within 48 hours.

-Wes

On Oct 11, 2014, at 7:48 AM, Peter Slow peter.s...@gmail.com wrote:

both of the defect posted here appear as internal and can't be viewed,
at least at the moment,with the bug search tool...

On Fri, Oct 10, 2014 at 9:42 AM, James Andrewartha
jandrewar...@ccgs.wa.edu.au wrote:
 Hi Ryan,
 
 After setting some more LLDP options from that URL I managed to get it
 working. I then spent the rest of the day trying to get paging working,
 which turned out the be the documented CSCtq36901, for which there is
 fortunately a workaround that I'll test on Monday.
 
 Thanks for your help,
 
 James
 
 Sent from my Samsung Galaxy smartphone.
 
 
  Original message 
 From: Ryan Ratliff (rratliff)
 Date:2014/10/10 21:25 (GMT+08:00)
 To: James Andrewartha
 Cc: cisco-voip voyp list
 Subject: Re: [cisco-voip] 3905 gives up on voice VLAN
 
 https://tools.cisco.com/bugsearch/bug/CSCub09465
 
 There's an ES load you can get from TAC with the fix if you can't avoid
 LLDP.
 
 -Ryan
 
 On Oct 9, 2014, at 9:08 PM, James Andrewartha jandrewar...@ccgs.wa.edu.au
 wrote:
 
 On 10/10/14 00:47, Ryan Ratliff (rratliff) wrote:
 Is LLDAP enabled on the switch by any chance?
 
 It is, however I set LLDP to advertise the voice VID of 100 as well and
 it still doesn't help. Although if I then disable LLDP entirely the
 phone does stay on the voice VLAN. Looks like I need to set a bunch more
 LLDP options for the phone to get the right settings,
 https://community.extremenetworks.com/extreme/topics/g_d_c_b_a_series_f_w_6_03_use_of_the_lldp_med_network_policy_tlv_feature
 has the sufficient set.
 
 Thanks,
 
 --
 James Andrewartha
 Network  Projects Engineer
 Christ Church Grammar School
 Claremont, Western Australia
 Ph. (08) 9442 1757
 Mob. 0424 160 877
 
 
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Re: [cisco-voip] Redirect CUCM Hunt Group to CUC Call Handler

[Wes Sisk (wsisk)]

a few ideas to explore:

* hunt group login/logout
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_5_1/ccmsys/accm-851-cm/a03rp.html#wp1100559

* Incoming calls to a shared line, let users change the call forward all 
settings on the shared line

* build a web service that changes forwarding or routing as desired. Take input 
from http parameters. If no parameters input then start a wizard. If input 
parameters provided then immediately apply those changes. Then make the web 
service, with desired parameters, as a service on the phone under services or a 
services button assignment. I suspect one of the Cisco partners likely already 
has this developed if you’re looking for something more out-of-the-box vs. 
build-your-own.

-Wes

On Oct 9, 2014, at 1:42 PM, Ben Story 
ben.st...@gmail.commailto:ben.st...@gmail.com wrote:

I'm deploying a CUCM 10.5 cluster for a group of medical offices.  Each office 
acts independently and are used to traditional key system features like Night 
Service/Bell mode.  Right now I have inbound calls to a particular office using 
plar to ring to a hunt group that includes all of the phones for an office.  
The customers would like to be able to press a button or in some way toggle a 
mode where the hunt group goes to a call handler on CUC.

They have already ruled out time of day routing as they want it to be on 
demand.  They've also rejected logging the phones out of the hunt group.  Short 
of writing an AXL web application to set forwarding options on the hunt group, 
is there anything that can mimic the behavior they want?
--
Ben Story
CCSP, CCNA, CCNA Wireless, CCDA
ben.st...@gmail.commailto:ben.st...@gmail.com
@ntwrk80
http://showbrain.blogspot.comhttp://showbrain.blogspot.com/
http://rand0mw0rds.blogspot.comhttp://rand0mw0rds.blogspot.com/


From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. 
Teresa of Avila
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Re: [cisco-voip] Delete Log Files

[Wes Sisk (wsisk)]

+1 on what Daniel said.

Scheduled trace collection with delete is the only way I can currently think of 
to do this out of the box. Some files, like install logs have long term value, 
not sure what your retention options are for those?

* It is possible to automate an ssh/CLI login and file manipulation, but the 
CLI find/delete doesn’t allow identification by data AFAIK
* We have long requested CSCsi21579allow scheduling commands  but still 
waiting. Maybe hit up your account team on this one.
* Setting a reduced number of files or file size will not guarantee  7 days. 
That is always dependent on system utilization, traces configured, and the 
amount of tracing generated by the features used.
* RTMT serviceability API’s, which do file retrieval and delete, are now 
public: 
https://developer.cisco.com/site/collaboration/management/uc-manager-serviceability/develop-and-test/documentation/latest-version/
 . Retrieval includes delete option for RTMT. I don’t see it in the API, 
checking into that.

-Wes

On Sep 24, 2014, at 9:24 AM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

If you absolutely can’t have any log files older than seven days on disk, one 
option would be to configure and schedule trace archiving for all services and 
applications, but make sure the “delete log files from the server” option is 
enabled.

This would provide you with two things:

1.   Log files collected off CUCM will be deleted permanently. This won’t 
only include CCM but other services and applications as well such as CTI Mgr, 
LBM, Tomcat Security, syslogs, etc.

2.   The log files you archive to a separate disk and, more importantly, 
the length of time they’re stored on disk, can be managed on the archive server 
via the example provided by Wes below (if a *nix OS) or the forfiles command I 
mentioned in a previous email (if a Windows OS).

Keep in mind this has the potential to put the customer into a situation where 
reported issues might go nowhere due to missing trace information since only 
seven days are retained. I’d also keep in mind the disk space required on your 
trace archiving server and overhead placed on CUCM –  older version of CUCM 
don’t automatically zip trace files on disk and, depending on specs, gzip can 
and has contributed to higher-than-expected CPU utilization. It will likely 
also include a very large number of log files needing to be transferred over 
FTP or SFTP, so there’s that to consider as well. You can minimize these two 
factors by scheduling it to occur once a day and during an after-hours window 
while avoiding an overlap of any backup jobs. You can also try to avoid large 
LDAP sync jobs or the 3:15 AM garbage collection task but it’s probably 
unnecessary.

I personally have never seen or configured CUCM trace and log archiving that 
encompassed so many services so I can’t really recommend it or speak from 
experience, but it, in theory, would most certainly accomplish the goal of 
managing the duration of all CUCM log files on disk, not just CCM SDI/SDL.

Hope this helps

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Martin Schmuker
Sent: Tuesday, September 23, 2014 5:15 PM
To: Wes Sisk (wsisk)
Cc: Cisco VoIP Mailing List
Subject: Re: [cisco-voip] Delete Log Files

Guys, thank you very much for your answers.

Sorry that I did not explain, why we want to delete old files. The reason is 
stupid German law regarding protection of privacy. Customer asks to delete 
files after of 7 days. In this case it’s not really a law, but client feels 
better :-(

From: Wes Sisk (wsisk) [mailto:ws...@cisco.com]
Sent: Tuesday, September 23, 2014 5:04 PM
To: Martin Schmuker
Cc: Cisco VoIP Mailing List
Subject: Re: [cisco-voip] Delete Log Files

onbox logging is circular. It will consume as much space as allocated and then 
loop over that. If something goes awry then Log Partition Manager (LPM) will 
auto-delete files as necessary.

For Scheduled Trace Collection, 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rttlc.html#wp1048184

No, there is nothing built into CUCM to manage the consumed disk space on the 
trace archive server. If using a *nix box a cron’d ‘find’ command does pretty 
well.

some possible examples:
# find files modified in the last 1 day
find . -type f -mtime -1d

-1d within 1 day -mtime n[smhdw]

-Wes

On Sep 23, 2014, at 6:13 AM, Martin Schmuker 
m...@bilobit.commailto:m...@bilobit.com wrote:

Guys,

is there any way to delete CUCM log files (aka traces) after x days?

Thanks,  Martin
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Re: [cisco-voip] Delete Log Files

[Wes Sisk (wsisk)]

onbox logging is circular. It will consume as much space as allocated and then 
loop over that. If something goes awry then Log Partition Manager (LPM) will 
auto-delete files as necessary.

For Scheduled Trace Collection, 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rttlc.html#wp1048184

No, there is nothing built into CUCM to manage the consumed disk space on the 
trace archive server. If using a *nix box a cron’d ‘find’ command does pretty 
well.

some possible examples:
# find files modified in the last 1 day
find . -type f -mtime -1d

-1d within 1 day -mtime n[smhdw]

-Wes

On Sep 23, 2014, at 6:13 AM, Martin Schmuker 
m...@bilobit.commailto:m...@bilobit.com wrote:

Guys,

is there any way to delete CUCM log files (aka traces) after x days?

Thanks,  Martin
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Re: [cisco-voip] increasing SDL file size from 2MB to, say 5MB.

[Wes Sisk (wsisk)]

still going to increase total consumed disk space, why not just setup scheduled 
trace collection?

1 day is pretty short window to get from user report, through operations, to 
TAC case, and investigation.

AFAIK no issues with files/sizes in recent versions.

-Wes

On Sep 23, 2014, at 3:37 PM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:


There is a limit of 10,000 files on the CCM / SDL trace configuration. At 2MB, 
it's giving us just over one day of trace files on our busy servers.

I'd like to reduce the number of files, say to 7,500 but increase the size of 
the files to say 5MB. This should give us a bit more than what we have now, but 
reducing the number of files.

Any word on whether or not the TAC likes or dislikes large files? Any other 
caveats?

Lelio


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

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

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Re: [cisco-voip] increasing SDL file size from 2MB to, say 5MB.

[Wes Sisk (wsisk)]

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/6_0_1/rtmt/rtmt601/rttlc.html#wp1048087

-Wes

On Sep 23, 2014, at 4:32 PM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:


I never even knew that scheduled downloads was an option. I could try 
downloading files once a day. See how that turns out.

Thanks!

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

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


From: Wes Sisk (wsisk) ws...@cisco.commailto:ws...@cisco.com
To: Lelio Fulgenzi le...@uoguelph.camailto:le...@uoguelph.ca
Cc: cisco-voip 
(cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net) 
cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Sent: Tuesday, September 23, 2014 4:15:17 PM
Subject: Re: [cisco-voip] increasing SDL file size from 2MB to, say 5MB.

still going to increase total consumed disk space, why not just setup scheduled 
trace collection?

1 day is pretty short window to get from user report, through operations, to 
TAC case, and investigation.

AFAIK no issues with files/sizes in recent versions.

-Wes

On Sep 23, 2014, at 3:37 PM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:


There is a limit of 10,000 files on the CCM / SDL trace configuration. At 2MB, 
it's giving us just over one day of trace files on our busy servers.

I'd like to reduce the number of files, say to 7,500 but increase the size of 
the files to say 5MB. This should give us a bit more than what we have now, but 
reducing the number of files.

Any word on whether or not the TAC likes or dislikes large files? Any other 
caveats?

Lelio


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

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

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Re: [cisco-voip] SCCP/SDL trace question re:transfer

[Wes Sisk (wsisk)]

Ed,

Unity doesn’t parse the soft keys sent from UCM to Unity to understand if 
“transfer” is available. Unity just blindly sends transfer and hopes the call 
completes.

I believe Progress is insufficient in UCM to enable transfer, usually have to 
wait for Alerting.

Attempting to replicate this with an SCCP phone is marginal value - SCCP phones 
implement the softkey set received from UCM. Thus the phone won’t allow you to 
send transfer when UCM has not activated that soft key.

Why is the egress call stopped in the progress state?

-Wes

On Aug 31, 2014, at 2:58 PM, Ed Leatherman 
ealeather...@gmail.commailto:ealeather...@gmail.com wrote:

Hello!

I'm trying to help chase down a intermittent issue where Unity needs to 
transfer a caller off-site to an answering service, and sometimes the transfer 
doesn't complete and the caller gets left on-hold. I was hoping someone could 
explain a message i'm seeing in the traces during a failure.

SCCP integration to unity connection. 9.1 software versions on both CUCM and 
Unity. MGCP to PRI gateways. All gateways are set to offnet and service 
parameter is configured to allow transfers between offnet to rule that out as a 
issue.

On the trace side of things, for the transfer leg on a failure I see:
19:55:27.146 : Unity presses transfer , dials out the digits
19:55:29.853 : Q931 IN from PSTN for the transfer leg,  PROGRESS message
19:55:29.855 : CUCM OUT to Unity: Call State Ring out
19:55:41.020 : CUCM OUT to Unity: DisplayNotify timeOutValue=15 notify='Cannot 
Complete Transfer' content='Cannot Complete Transfer' ver=12

It looks like an abnormal amount of time for the call to connect, is that a 
possible reason for the Cannot Complete Transfer message? Is the timeout 
tweakable someplace?

On successful tries, the transfer leg connects faster (less than 10 seconds). 
So far we haven't found anything else different on our own; have a TAC case 
open on it but getting shuffled between groups now (unity team wants CUCM team 
to look at it).

Unity never seems to retrieve the caller from hold or try again, eventually the 
caller hangs up (I see the the DISCONNECT message from PSTN) at which point 
that call leg gets torn down.

Any ideas much appreciated

Ed

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Re: [cisco-voip] SCCP/SDL trace question re:transfer

[Wes Sisk (wsisk)]

Historically this has happened because:
* gateway attempts to setup audio but call signaling is not at a state that 
would allow transfer to complete
* the attempt to setup audio sends OpenReceiveChannel to Unity port
* Unity ignores because unity has already sent 2nd transfer and considers the 
call as transferred

Still, goes back to call not being in a valid state for the transfer to 
complete and Unity trying to send transfer anyway.

Address this specifically call signaling flow or use supervised transfer.

-Wes

On Sep 2, 2014, at 12:04 PM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

Ed:

Detailed CCM traces should suffice. If it indeed was a 12 second media exchange 
timeout, you should notice a missing SCCP or MGCP transaction after receiving 
the ISDN Call Proceeding event. I would check to make sure I see the 
OpenReceiveChannel, StartMediaTransmission, and OpenReceiveChannelACK on the 
SCCP call-leg followed by a MDCX with SDP to the MGCP gateway and a 200 
response – all immediately after ISDN Call Proceeding comes in. If you notice 
one of these missing then it’s likely an MX timeout issue. I’ve recently seen 
an issue where StationD doesn’t ACK an OpenReceiveChannel signal, resulting in 
a MX timeout. Doubt it’s related to this problem though… my issue was related 
to CTI ports.

- Dan

From: Ed Leatherman [mailto:ealeather...@gmail.com]
Sent: Tuesday, September 02, 2014 10:36 AM
To: Daniel Pagan
Cc: Sreekanth Narayanan; Mike Nickolich; Cisco VOIP
Subject: Re: [cisco-voip] SCCP/SDL trace question re:transfer

Dan,

I'm not seeing a MXTimeout, however the Cannot Complete Transfer is 12 
seconds after the ISDN Proceeding. Any special trace settings necessary to see 
that message?

22:58:38.411 |AppInfo  |In  Message -- PriCallProceedingMsg -- Protocol= 
PriNi2Protocol
..
22:58:50.310 |AppInfo  |StationD:(0331221) DisplayNotify timeOutValue=15 
notify='Cannot Complete Transfer' content='Cannot Complete Transfer' ver=12.




On Tue, Sep 2, 2014 at 9:34 AM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:
Ed:

As a test, are you able to recreate the issue when PSTN leg doesn’t answer for 
12 seconds after the transfer attempt? I ask because, based on the timestamps 
below, it seems the media exchange timer might be expiring. If you still have 
SDL traces, you can search for “MXTimeout”. If you find one, you should be able 
to backtrack 12 seconds and find the ISDN Call Proceeding message that triggers 
CUCM’s attempt at connecting media between the two call-legs.

- Dan


From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.netmailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Sreekanth Narayanan
Sent: Tuesday, September 02, 2014 2:22 AM
To: Ed Leatherman
Cc: Mike Nickolich; Cisco VOIP
Subject: Re: [cisco-voip] SCCP/SDL trace question re:transfer

Hi Ed,

If the Unity is sending the 2nd transfer command as soon as the initial call 
setup begins, it looks more like a blind transfer. The other transfer type 
'Supervise Transfer' is the consult transfer. Have you tried to do blind 
transfers from SCCP phones?

As per the RTS description, it's the responsibility of the CUCM to handle the 
call if the target of the transfer is busy or doesn't answer.

  *   Release to Switch—Unity Connection puts the caller on hold, dials the 
extension, and releases the call to the phone system. When the line is busy or 
is not answered, the phone system—not Unity Connection—forwards the call to the 
user or handler greeting. This transfer type allows Unity Connection to process 
incoming calls more quickly. Use Release to Switch only when call forwarding is 
enabled on the phone system.


Thanks
Sreekanth


On 1 September 2014 19:29, Ed Leatherman 
ealeather...@gmail.commailto:ealeather...@gmail.com wrote:
Hi Sreekanth,

The problem is inconsistent, but definitely more than say 20%. Load on our 
systems doesn't appear to be an issue, we did testing late at night/after hours 
and regular call volume is very low then. We were able to duplicate the issue 
just with cell phones as the target, so it seems the problem is not just the 
answering service not picking up; not had a problem where direct calls weren't 
answered promptly.

In looking through the trace files, it seems like Unity does a consult transfer 
even when set to Release to switch, its just sending the 2nd transfer command 
as soon as the initial call setup starts - IIRC it was doing it right after we 
got PROCEEDING from PSTN.

I did check out the T301 timer in CUCM but its still set to 3 minutes - so 
we're not hitting that one at least.

Your idea of reproducing the issue with a consultative transfer from a phone is 
a good one, we'll give that a try.

For now we just have their line directly forwarded after hours manually and 
skipping Unity completely and it works. They pay per call to the answering 
service though so they really want the front end IVR to pick up first. It is a 
suicide prevention 

Re: [cisco-voip] CCM Trace Question || MediaManager PTime

[Wes Sisk (wsisk)]

+1. Wouldn’t be the first time I’ve thought wrong but I think of these values 
as the max msec supported for each codec. I always thought there was an 
interesting underlying assumption that devices could do less than the 
advertised max msec. i.e. this device support codec 12 with maximum of 60 msec 
per packet.


-Wes

On Jul 15, 2014, at 5:41 PM, Peter Slow peter.s...@gmail.com wrote:

I've always seen medimanager use msec in that second field.
in sip there are ptime and maxptime parameters, ptime is supposed to
be msec per packet in the context of SIP SDP, which is really where
that term comes from. i dont think i've ever seen anything handle
maxptime correctly. ptime tends to get treated as max ptime, meaning
that if you told me you support 20 msec per packet, then sending you
10 msec pr packet should be okay.
This is almost a requirement, since various scenarios in a call can
cause a lower-volume packet to be sent. For instance, we've got a
ptime of 30 msec negotiated, and i hang up on you at 1 minute and 10
msec - I dont think a coder will insert 20 msec of silence, i beleive
you'd just expect to receive a from with one 10 msec sample in it.

Mediamanager is most likely (should) be showing you the ptime received
in media negotiation - not necessarily the CUCM defaults from CCM
service parameters. the Enforce Millisecond Packet Size and Always
Use Preferred G.729 Packet Size For SIP Trunk Answers service
parameters may have to do with whatever behavior you're seeing.

Is there something specific you're troubleshooting?

-Pete

On Tue, Jul 15, 2014 at 1:37 PM, Ryan Ratliff (rratliff)
rratl...@cisco.com wrote:
 I believe ptime is the number of frames per packet.
 
 -Ryan
 
 On Jul 15, 2014, at 10:28 AM, Daniel Pagan dpa...@fidelus.com wrote:
 
 Folks:
 
 
 
 Quick question on the preCheckCapabilities step for MediaManager. After a
 new MediaManager is created, I’ve often referred to the preCheckCapabilities
 line for determining codec and DTMF relay capabilities based on codec type
 numbers (Cap,ptime) and figured that ptime was simply the payload size in
 ms. While looking at a recent issue, I noticed that ptime didn’t exactly
 match up with packet time values specified in CCM service parameter
 settings. The environment was using default settings across the board for
 packet size in milliseconds, but advertised capabilities shown in
 preCheckCapabilities for an IP phone to IP phone call showed:
 
 
 
 (Cap,ptime)= (25,40) (4,40) (2,40) (15,60) (16,60) (11,60) (12,60)
 
 
 
 Here I’m seeing g.711 with a ptime value of 40 and g.729 with 60. In this
 environment, g.711 and g.729 is configured for 20 ms packet size. So for
 clarification, assuming my understanding is incorrect, can someone tell me
 what the ptime value represents? We also thought maybe maximum payload size
 in ms but service these values go beyond the allowable values in CUCM.
 
 
 
 Thanks ahead of time!
 
 
 
 - Daniel
 
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Re: [cisco-voip] Monitoring tool or best way to monitor

[Wes Sisk (wsisk)]

If MGCP you can see the calls per DS-1 (PRI / CAS) in RTMT and create alerts on 
the per-ds1 level but not per gateway and not in aggregate.

RTMT provides the starting point for customization. You can use ‘show risdc…’ 
from CLI on UCM to see the same counters. Or you can use AXL Serviceability API 
to poll UCM and pull the values into your programming language of choice. From 
there you can aggregate your sites or enterprise as necessary. You could even 
use AXL to push in config changes automatically.

The APIs are exposed natively. It takes a bit of programming to leverage them. 
Cisco partners may excel here.

-Wes

On Jun 18, 2014, at 8:25 AM, Michael Rose VOIP rose.michael.v...@gmail.com 
wrote:

My company frequently does mass meetings where one of the C level people will 
talk and employees will dial into a conference number. My group has to monitor 
gateways so that the active calls on each gateway does not exceed 80%. If it 
does we put up blocks via CUCM to drop the call.

What I am wondering is if there is a better way to do this. Currently what we 
do is split the offices among the team and each of us logs into the voice 
gateways and watches the active calls. How we monitor the active calls depends 
on the person and the size of the location. Each of us has to watch at least 
two gateways and we don't cover every location, just the major ones since we've 
got offices everywhere.

Is there a tool where we can plug in the voice gateway names and login 
credentials and it will just spit out active calls or number of active calls to 
a specific number? We've tried using RTMT but there are just so many gateways 
we can't monitor them all very well using that tool. I wanted to know what was 
out there before we try and build something on our own.

Do you guys have a better way to monitor gateway thresholds on over 11 sites 
with like 44 voice gateways?
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Re: [cisco-voip] Monitoring tool or best way to monitor

[Wes Sisk (wsisk)]

Hi Michael,

I am not aware of any means of differentiating inbound and outbound in Perfmon 
Counters or AXL serviceability API.

-Wes

On Jun 18, 2014, at 2:39 PM, Michael Rose VOIP 
rose.michael.v...@gmail.commailto:rose.michael.v...@gmail.com wrote:

Good info about pulling the information into a programming language to write a 
custom tool. I was thinking of this though no one in our department is a 
programmer. The alerts on a PRI basis might also work since we try to only send 
outbound on specific circuits and only overflow when required.

Can you get more granular with alerts? For example if all the outbound PRIs 
fill up and we start overflowing to the ones we reserve for inbounds can we 
tell RTMT to look at the circuit and say that we've got XX amount of outbound 
calls on that circuit, and alert us? Can it tell outgoing vs incoming on the 
alert level?


On Wed, Jun 18, 2014 at 10:40 AM, Wes Sisk (wsisk) 
ws...@cisco.commailto:ws...@cisco.com wrote:
If MGCP you can see the calls per DS-1 (PRI / CAS) in RTMT and create alerts on 
the per-ds1 level but not per gateway and not in aggregate.

RTMT provides the starting point for customization. You can use ‘show risdc…’ 
from CLI on UCM to see the same counters. Or you can use AXL Serviceability API 
to poll UCM and pull the values into your programming language of choice. From 
there you can aggregate your sites or enterprise as necessary. You could even 
use AXL to push in config changes automatically.

The APIs are exposed natively. It takes a bit of programming to leverage them. 
Cisco partners may excel here.

-Wes

On Jun 18, 2014, at 8:25 AM, Michael Rose VOIP 
rose.michael.v...@gmail.commailto:rose.michael.v...@gmail.com wrote:

My company frequently does mass meetings where one of the C level people will 
talk and employees will dial into a conference number. My group has to monitor 
gateways so that the active calls on each gateway does not exceed 80%. If it 
does we put up blocks via CUCM to drop the call.

What I am wondering is if there is a better way to do this. Currently what we 
do is split the offices among the team and each of us logs into the voice 
gateways and watches the active calls. How we monitor the active calls depends 
on the person and the size of the location. Each of us has to watch at least 
two gateways and we don't cover every location, just the major ones since we've 
got offices everywhere.

Is there a tool where we can plug in the voice gateway names and login 
credentials and it will just spit out active calls or number of active calls to 
a specific number? We've tried using RTMT but there are just so many gateways 
we can't monitor them all very well using that tool. I wanted to know what was 
out there before we try and build something on our own.

Do you guys have a better way to monitor gateway thresholds on over 11 sites 
with like 44 voice gateways?
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Re: [cisco-voip] CDR and CMR Dump Tables

[Wes Sisk (wsisk)]

With diagnostics enabled then it comes down to whether the endpoint supports 
CMRs and any insert errors that may occur when CAR attempts to insert raw 
records into the database.

1. what are the endpoints in question? what version of code?
2. any interesting CMR errors? have you checked the plain text files before 
loading into the database?

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_0_1/admin_master/servadmin/sacdrm.html

-Wes

On Jun 13, 2014, at 6:25 PM, Mark Rudholm 
m...@rudholm.commailto:m...@rudholm.com wrote:

Yeah, I just double checked that it's set to Enabled Regardless of CDR Enabled 
Flag

A lot of the fields are populated, just some are empty, including those 
important ones that would show me network issues.

On 06/13/2014 03:12 PM, Jason Aarons (AM) wrote:
Did you enable Diagnostic CDRs ?

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/5_1_3/car/car/caranrpt.html#wp1037699

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Mark 
Rudholm
Sent: Friday, June 13, 2014 5:30 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] CDR and CMR Dump Tables



I'm trying to debug some call quality problems. I took a look at a CDR
dump data, but the columns relating to packet counts, jitter, and packet
loss are all empty. Poring through Cisco docs and I can't seem to
figure out how to enable the logging of these fields.

orignumberOctetsReceived orignumberPacketsLost destnumberPacketsSent
destnumberOctetsSent destnumberPacketsReceived destnumberOctetsReceived
destnumberPacketsLost origjitter destjitter origlatency destlatency

Any pointers would be appreciated.

-Mark
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itevomcid

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Re: [cisco-voip] Heartbeat Failure SNRD

[Wes Sisk (wsisk)]

Hi Daniel,

Great find!

For the document:
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/46806-cm-crashes-and-shutdowns.html

The initialization process and timers have changed *significantly* since 4.x. 
Some examples include:
CSCsj76788cp-system request to remove initialization timers
“... remove the initialization timers that are started during CUCM 
initialization.  These timer would previously cause a system restart under 
certain circumstance…”

Still, there is a global maximum timeout. Individual Daemons must report start 
and successful initiation by that time.

Historically behavior like you discuss was triggered by service parameters 
being missing or having incorrect values. This may be a problem with connection 
to the database ( CSCsc72748 ) or problem with the contents of the database. 
Other problems include another process grabbing one of the TCP or UDP ports 
required by the ccm process.

ccm had many issues retrieving initialization information from the database in 
early linux versions. refinements to informix and in memory database (IMDB) 
have helped significantly.

-Wes


On May 21, 2014, at 9:33 AM, Daniel Pagan 
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:

Folks:

CUCM ES 8.6.2.24122-1 appears to be creating an issue where CallManager 
heartbeat fails to increment upon startup and the condition that must be met is 
very specific. On a problematic node, SDL traces show the following error 
exactly one hour after the start of the CCM service:

AppError  ||Local send blocked: SignalName: Start, DestPID: SNRD[1:100:61:1]

This error is followed by the SDL trace printing an error stating CallManager 
exceeded the permitted time for initialization and will restart the 
application. The CCM application restarts and additional SDL traces are printed 
showing the standard creation of critical processes – one hour later the same 
“Local send blocked” error is printed regarding the SNRD process.

I saw the DestPID: SNRD error, went to a completely different, non-problematic 
lab environment where 8.6.2.24122-1 is installed, created a single Remote 
Destination Profile, and then restarted the standalone node in order to force 
the creation of SNRD. CallManager heartbeats are now failing to increment in 
that environment and found another “Local send blocked” error regarding SNRD. 
Removing the single Remote Destination Profile from the standalone environment 
and rebooting the node resolves the problem. Re-inserting it again followed by 
a reboot recreates it, making SNRD the obvious culprit here.

I currently have a TAC case open where they’re attempting to recreate the 
problem. It seems no public facing defects are created for this. Just wanted to 
give you folks a heads up.

Related to this, can someone tell me if this document, specifally the section 
describing MMManInit and process creation, is still accurate? If so, then what 
I fail to see in SDL traces is a InitDone signal from SNRD to MMManInit during 
the 60 minutes between CCM startup and initialization timeout.

- Daniel

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Re: [cisco-voip] Multistage Dialing FAC

[Wes Sisk (wsisk)]

FAC/CMC is invoked after the route pattern is matched. FAC/CMC only work on the 
cluster where the device is registered.

I’m not aware of any conflict with multistage dialing offhand. What does 
multistage dialing mean to you?

-Wes

On May 2, 2014, at 2:14 PM, Heim, Dennis 
dennis.h...@wwt.commailto:dennis.h...@wwt.com wrote:

Is it possible with multistage dialing to enter in cucm forced authorization 
codes?

Dennis Heim | Solution Architect (Collaboration)
World Wide Technology, Inc. | 314-212-1814

PS Engineering:  Innovate  Ignite.


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Re: [cisco-voip] Multistage Dialing FAC

[Wes Sisk (wsisk)]

Thanks for the test Brian!

This was added with:
CSCts20551UCM 9.0(1): Speed dial support for pause and FAC/CMC

Looks like it should work with most endpoint types, YMMV. The core change is in 
UCM so all endpoints should be able to leverage it so long as they use the 
“standard speed dial” functionality. Some endpoints and applications implement 
their own speed dials - clearly those would not benefit from this change.

-Wes

On May 2, 2014, at 4:38 PM, Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu wrote:

Dennis,

I just tested this successfully with CUCM 9.1.2 and IP Communicator.

I built the speed-dial as ,1234 where  is my route pattern and 1234 
is my FAC.

The StationOffHook message looks like this:
mDialedDigits=,1234

I then see a digit analysis for my  route pattern:
00151412.008 |16:31:42.117 |AppInfo  |Digit analysis: match(pi=2, 
fqcn=1000, cn=1000,plv=5, pss=, TodFilteredPss=, dd=,dac=0)


CUCM successfully pulls in the FAC:
00151441.001 |16:31:42.121 |AppInfo  |FacCmc - DETAIL: valid authorization 
code, code(1234), desc(testfac), lvl(10), required lvl(10)

I then get my outgoing Invite to my SIP Trunk:
00151477.001 |16:31:42.138 |AppInfo  |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing 
SIP UDP message to 10.3.11.251:[5060]:
[9,NET]
INVITE sip:@10.3.11.251:5060http://sip:@10.3.11.251:5060/ SIP/2.0
Via: SIP/2.0/UDP 10.3.11.250:5060;branch=z9hG4bK2388981c3
From: 
sip:1000@10.3.11.250mailto:sip%3A1000@10.3.11.250;tag=4~aa150fb0-a9a9-4bd2-ad97-c9705eff01c2-25079840
To: sip:@10.3.11.251mailto:sip%3A@10.3.11.251
Date: Fri, 02 May 2014 20:31:42 GMT
Call-ID: 
c3ff9080-364100ae-3-fa0b030a@10.3.11.250mailto:c3ff9080-364100ae-3-fa0b030a@10.3.11.250
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-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: 
sip:10.3.11.250:5060http://10.3.11.250:5060/;method=NOTIFY;Event=telephone-event;Duration=500
Cisco-Guid: 3288305792-065536-03-4195025674
Session-Expires:  1800
P-Asserted-Identity: sip:1000@10.3.11.250mailto:sip%3A1000@10.3.11.250
Remote-Party-ID: 
sip:1000@10.3.11.250mailto:sip%3A1000@10.3.11.250;party=calling;screen=yes;privacy=off
Contact: sip:1000@10.3.11.250:5060http://sip:1000@10.3.11.250:5060/
Max-Forwards: 70
Content-Length: 0

This behavior may be dependent on phone model though.  You should try this on 
whatever model phones you use.

CUCM documented this feature for 8941s and 8945s as a new feature for CUCM 
9.0.1:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/rel_notes/9_0_1/delta/CUCM_BK_N38FD301_00_cucm-new-and-changed-90/CUCM_BK_N38FD301_00_cucm-new-and-changed-90_chapter_011.html#P415_RF_P7683F5B_00

Brian


On Fri, May 2, 2014 at 3:45 PM, Heim, Dennis 
dennis.h...@wwt.commailto:dennis.h...@wwt.com wrote:
Requirement is to have a speed dial that automatically enters the FAC.

Dennis Heim | Solution Architect (Collaboration)
World Wide Technology, Inc. | 314-212-1814tel:314-212-1814

PS Engineering:  Innovate  Ignite.


From: Wes Sisk (wsisk) [mailto:ws...@cisco.commailto:ws...@cisco.com]
Sent: Friday, May 02, 2014 2:55 PM
To: Heim, Dennis
Cc: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Multistage Dialing  FAC

FAC/CMC is invoked after the route pattern is matched. FAC/CMC only work on the 
cluster where the device is registered.

I’m not aware of any conflict with multistage dialing offhand. What does 
multistage dialing mean to you?

-Wes

On May 2, 2014, at 2:14 PM, Heim, Dennis 
dennis.h...@wwt.commailto:dennis.h...@wwt.com wrote:

Is it possible with multistage dialing to enter in cucm forced authorization 
codes?

Dennis Heim | Solution Architect (Collaboration)
World Wide Technology, Inc. | 314-212-1814tel:314-212-1814

PS Engineering:  Innovate  Ignite.


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Re: [cisco-voip] Run on All Active Unified CM Nodes in 8.5 and higher

[Wes Sisk (wsisk)]

I’m not aware of RouteLists doing “Run on all nodes” in 7.x. Going way back in 
time they did. In 7.x the route list should have registered on the nodes in the 
associated CM group.

I suspect you are hitting one of CSCtq10477, CSCul71689, CSCum85086

Checking the box just started using DeviceManager, 
SubscriptionManager,RouteListManager on the code where the call originated vs. 
the sequence of processes that was handling it before. If you registered a 
phone to each node in the cluster one at a time I suspect you would find at 
least one node where calls would still fail.

-Wes



On Apr 16, 2014, at 12:59 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:

Topic: Run on All Active Unified CM Nodes in 8.5 and higher

So prior to 8.5 was the default to Run on All Active Unified CM Nodes?

I upgraded a working customer from CallManager 7.0 to 9.1.2. They have 3 subs.  
After the upgrade they couldn’t make calls out a MGCP gateway.  We put a check 
in the Route List for Run on All Active Unified CM Nodes and the problem is 
fixed. They have 30+ Route lists for 30+ sites.  Each site has single PRI with 
MGCP.  Small spoke offices.

The have a single RL with a single RG pointed to a single MGCP gateway.

The upgrade has none of the Route Lists with Run on All Active Unified CM 
Nodes.  I can’t think of a reason in this scenario (note we have no SIP) not to 
put the checkmark in the Route Lists for Run on All Active Unified CM Nodes.

SRND Run on all active nodes
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/session_mgmt/deploy/8_5_1/overview.html




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Re: [cisco-voip] CUCM 10

[Wes Sisk (wsisk)]

With eDelivery you get a custom link to perform your download. You will need to 
obtain the software via the order fulfillment information provided. That is the 
only way to fulfill your order.

Regards,
Wes

On Apr 7, 2014, at 11:58 AM, Michel L. M. B. Perez 
michelmbpe...@gmail.commailto:michelmbpe...@gmail.com wrote:


Yes guys, i purchased NFR kit for partners on Cisco Marketplace.

It was edelivery, but the system is not making me download that, any sugestion?

Em 07/04/2014 10:20, Matt Slaga (AM) 
matt.sl...@dimensiondata.commailto:matt.sl...@dimensiondata.com escreveu:
Yes, you can order NFR versions.

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.netmailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Michel L. M. B. Perez
Sent: Saturday, April 5, 2014 10:04 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] CUCM 10


Anyone here already have access for download CUCM 10?

Thanks.

--
Michel Perez
Skype: michelmbperez
michelmbpe...@gmail.commailto:michelmbpe...@gmail.com
http://br.linkedin.com/in/michelmbperez


itevomcid
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Re: [cisco-voip] User still in corp directory that was removed from AD. 9.1.1.20000 (9.1.1a)

[Wes Sisk (wsisk)]

Are they still in the enduser table? What is their status?

run sql select userid,status from enduser

Garbage collection runs automatically at the fixed time of 3:15 AM and deletes 
the users marked inactive.

To keep the users after deleting LDAP just do run sql update enduser set 
status=1 where userid=id , they won't get deleted and will be able to login.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/directry.html#wp1045229


A few interesting points:
CSCuc58240LDAP: After Garbage Collection End-User and IMS Phone type not 
removed
CSCsq94034Need to add garbage collection Logs for serviceability

Based on that last one it looks like this is written to the ldap sync logs. 
Search for the text Gargabe collection routine started
https://supportforums.cisco.com/document/65651/communications-manager-rtmt-trace-locations-cli

Cisco DirSync
activelog cm/trace/dirsync/log4j/dirsync.log

-Wes

On Mar 17, 2014, at 5:32 PM, Erick Bergquist erick...@gmail.com wrote:

CUCM 9.1.1a (9.1.1.2) LDAP integrated with one directory.

There are a few users not listed in the end user directory anymore in CUCM, but 
on the phone corp directory and in Jabber you see these users still. The 
directory URL is pointing to the CUCM server and we have tried both servers in 
the cluster.

If we add a test user, they show up in the directory and when we delete the 
test user they go inactive and go away after 2-3 days when garbage collection 
job clears them out and the test user is no longer in the directory.

The users are no longer in LDAP anymore either.  Is there a way to 
purge/refresh the CUCM directory?  Not finding much in bug search tool yet.

If we remove the LDAP Directory and add it back think that may clear up 
something?
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[cisco-voip] UCM, UC, UCCX, CER, TP, VOS remote account usage statistics

[Wes Sisk (wsisk)]

I am working on serviceability enhancements across Cisco Voice products and 
currently focusing on use of remote accounts. These are situations where 
partners and customers must call Cisco to get into remote account (root) to 
diagnose and fix problems.

Having some real live remote access logs would help in this regard. If you are 
willing to share your remote access logs please collect them and unicast them 
to me, wsisk[at]cisco.com.

I will be parsing them to find the commands most often used from remote account 
shell. Feel free to review the logs for yourself to understand changes made to 
your system or to anonymize logs as needed before submitting them.


To collect remote access logs:
admin:file get activelog audit/vos/remote*

Thank you,
Wes
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