Re: [cisco-voip] BE7K or not?

[Ryan Huff]

Lelio,


Yes, you can remove the pre-installed VM's and deploy your own (with respect to 
the allowances of the TRC).


The reason adding additional hardware isn't supported is because the Business 
Edition UCS chassis's are sold as a TRC (Tested Reference Configuration); 
meaning that Cisco has tested a variety of approved configurations on the 
platform with which TAC will support end-to-end.


Configurations (hardware or software) outside those parameters have not been 
tested and could cause unplanned/unpredictable results with which TAC will not 
support. With the TRC stuff, you pretty much need to follow the DocWiki and 
deployment guides chapter and verse.


If you'd like to be able to customize your UCS chassis and perhaps capitalize 
on co -residency allowances (again, within supported parameters), you'd want to 
look at something called a specs-based configuration or a data center product 
like a B series chassis.


If you'd like to provide us with some detail on what your long-term objectives 
are with the upgrade and some demographics on your user/device population we 
might be able to give you some more informed insight.


Hope that helps,


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com



From: cisco-voip  on behalf of Lelio 
Fulgenzi 
Sent: Thursday, December 3, 2015 3:45 PM
To: cisco-voip voyp list
Subject: Re: [cisco-voip] BE7K or not?


Thanks for everyone's responses. I read on one of the pages that BE7K comes 
pre-loaded with UCCx 4 vCPU OVA, but I'm guessing that I can run whatever OVA I 
want as long as it fits?

We were told that the BE7K is not customizable, even after ordering, so adding 
memory or anything like that is not permitted. This sort of makes me cringe. I 
like having the opportunity to add memory and/or additional NICs as with the 
other TRCs. Can't really justify it now, but as future versions come up, I'd 
like to have the ability to add more memory if the CUCM v11 or v12 version 
requires it.

It's really only memory and NIC throughput that I can see having to increase 
and only as the apps demand more with each version.

Lelio


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

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


From: "Rob Dawson" 
To: "cisco-voip voyp list" 
Sent: Thursday, December 3, 2015 11:27:07 AM
Subject: Re: [cisco-voip] BE7K or not?


The newest BE7ks are built on C240 M4 hardware. The BE7H-M4-K9 is 20 cores, 128 
GB RAM, 4+ TB of available disk space, the BE7M-M4-K9 is 12 cores, 64 GB RAM, 
and around 2.6 TB available disk.



Cisco states on the data sheet that BE7k is “optimized for enterprise-scale 
organizations with 1000 to 5000 users and 3000 to 15,000 devices” but they also 
note “For more capacity to support larger sized deployments, simply stack 
additional servers. And in smaller sized deployments with less than 1000 users, 
typically more applications can be supported per server.”



I haven’t seen any practical limitations, and as pointed out by someone else 
you can use the VM Placement Tool to see what OVAs fit, and scale to size.



Rob



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Lamont, Joshua
Sent: Thursday, December 03, 2015 10:21 AM
To: Lelio Fulgenzi
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] BE7K or not?



Sorry for the confusion... We had originally planned to install C240 M3s but 
swapped them out for the C240 M4s instead.


Joshua Lamont

Senior Telecommunications Engineer

Brown University

office (401) 863-1003

cell(401) 749-6913



On Thu, Dec 3, 2015 at 10:14 AM, Lelio Fulgenzi 
> wrote:



There are BE7Ks built on the new UCS C240 M4s.



Or by "new" do you mean something else other than the 240M4s?





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



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





From: "Joshua Lamont" >
To: "Matthew Loraditch" 
>
Cc: "Lelio Fulgenzi" >, "cisco-voip 
voyp list" 

[cisco-voip] International CP tones

[Ryan Huff]

Nifty little tool I tripped over on the Internet for International CP tones ...


http://www.3amsystems.com/World_Tone_Database



= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

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Re: [cisco-voip] CUCM 10.5 Missing CLI commands

[Ryan Huff]

My initial thoughts are that somehow you are not using the default platform 
administrator account?


Can you run;


"show myself"

AND

"show account"


 What are the results?


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com



From: cisco-voip  on behalf of Dave Wolgast 

Sent: Wednesday, December 2, 2015 10:35 PM
To: Cisco VOIP Newsletter - puck.nether.net
Subject: [cisco-voip] CUCM 10.5 Missing CLI commands

I just did a 5- node 7.1(5)->10.5(2) PCD migration, which appeared to go pretty 
well. What I have noticed, though, is that two of my subscribers are missing 
commands from the CLI.

One thing I needed to do for the migrated cluster was to add DNS servers and 
add a domain, but on 2 of the 5 servers, there was no 'set network' option. I 
thought maybe a reboot would help, but alas, on the same servers, there was no 
'utils system' commands.

Anybody ever seen anything like this before? I assume I will just be rebuilding 
these subscribers manually, but it seems awfully strange, especially since it 
didn't happen to every server.

Baffled,
Dave Wolgast
Livonia, NY
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Re: [cisco-voip] CUCM 10.5 Missing CLI commands

[Ryan Huff]

If you do go the route of re-building your subs; be mindful of any custom moh 
files ... etc (they're easy to forget).


file list activelog mohprep

file get activelog mohprep/


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Ryan Huff 
<ryanh...@outlook.com>
Sent: Wednesday, December 2, 2015 10:43 PM
To: Dave Wolgast; Cisco VOIP Newsletter - puck.nether.net
Subject: Re: [cisco-voip] CUCM 10.5 Missing CLI commands


My initial thoughts are that somehow you are not using the default platform 
administrator account?


Can you run;


"show myself"

AND

"show account"


 What are the results?


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Dave Wolgast 
<dwolg...@rochester.rr.com>
Sent: Wednesday, December 2, 2015 10:35 PM
To: Cisco VOIP Newsletter - puck.nether.net
Subject: [cisco-voip] CUCM 10.5 Missing CLI commands

I just did a 5- node 7.1(5)->10.5(2) PCD migration, which appeared to go pretty 
well. What I have noticed, though, is that two of my subscribers are missing 
commands from the CLI.

One thing I needed to do for the migrated cluster was to add DNS servers and 
add a domain, but on 2 of the 5 servers, there was no 'set network' option. I 
thought maybe a reboot would help, but alas, on the same servers, there was no 
'utils system' commands.

Anybody ever seen anything like this before? I assume I will just be rebuilding 
these subscribers manually, but it seems awfully strange, especially since it 
didn't happen to every server.

Baffled,
Dave Wolgast
Livonia, NY
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Re: [cisco-voip] CCX 9.0.2 Licensing Issues

[Ryan Huff]

There was a terminated bug report in 8.5 for this, not much detail though: 
https://tools.cisco.com/bugsearch/bug/CSCto67019/?referring_site=bugquickviewredir

I would be curious as to what the clock strata on the primary server was at, 
prior to the reboot and the type of reference server?

If you have root access and are familiar with the CET tool, you can change the 
server's MAC. The official (and supported) way to do it is through a TAC case; 
sometimes faster than GLO.

-Ryan

On Dec 1, 2015, at 7:33 PM, Erick Bergquist 
<erick...@gmail.com<mailto:erick...@gmail.com>> wrote:

I've seen it once or twice on reboots. There is a licensemacchanges file under 
the install folder you can view.

File view install ...

I've also had the license Mac change in 8.5.1 when applying just a SU patch and 
no changes to the variables making up the hash.


Erick


On Tuesday, December 1, 2015, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

Dan,


The server's licensing MAC address is a hashed hexadecimal value based on 
several key settings in the operating system. Any changes to the following list 
of OS settings will cause the License MAC on the server to change, thereby 
invalidating the provisioned license file (which is bound to the server's 
licensing MAC address). Did anything change on the server, that would effect 
the licensing MAC, prior to the reboot? As you mentioned, re-hosting the 
license through GLO (or reverting the change) are the only remedial courses of 
action.


Things that impact the server's licensing MAC:

  *   Time zone
  *   NTP server 1 (or none)NIC speed (or auto)
  *   Hostname
  *   IP Address

  *   IP Mask

  *   Gateway Address

  *   Primary DNS

  *   SMTP server

  *   Certificate Information (Organization, Unit, Location, State, Country)


= Ryan =



Email: 
ryanthomash...@outlook.com<javascript:_e(%7B%7D,'cvml','ryanthomash...@outlook.com');>

Spark: 
ryanthomash...@outlook.com<javascript:_e(%7B%7D,'cvml','ryanthomash...@outlook.com');>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<javascript:_e(%7B%7D,'cvml','cisco-voip-boun...@puck.nether.net');>>
 on behalf of Dan Mason 
<dma...@winxnet.com<javascript:_e(%7B%7D,'cvml','dma...@winxnet.com');>>
Sent: Tuesday, December 1, 2015 10:23 AM
To: 
cisco-voip@puck.nether.net<javascript:_e(%7B%7D,'cvml','cisco-voip@puck.nether.net');>
Subject: [cisco-voip] CCX 9.0.2 Licensing Issues


Hi,


Has anyone run into licensing not working due to invalid MAC address after 
server reboot?  Ive had this occur twice over the last 12 months.  Single CCX 
server on VMWare using Cisco .ova template.  I had to get the licensing 
rehosted to resolve the issue.  I didn't see anything in caveats for the 
release.









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Re: [cisco-voip] Jabber phone mode outbound calls issue

[Ryan Huff]

In your case, it seems that video capabilities might have been sent to the 
provider. In the case of TDM providers, this will almost always result in the 
call being dropped by the provider.

For TDM, You can use "voice-cap speech" on the voice port to prevent video 
capabilities from being sent to the provider, for SIP you could use a sip 
profile.

Although, an even easier method, IMO, is to adjust CCM's regional settings to 
prevent video capabilities from making it to the gateway (unless there are 
other reasons to let video capabilities to make it to the gateway such as 
additional sip trunks that do allow video capabilities).

-Ryan


 Original message 
From: Abebe Amare
Date:11/27/2015 9:44 AM (GMT-05:00)
To: Ryan Huff
Cc: abbas wali ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue

I have encountered a similar problem in that I can't make outgoing calls to 
PSTN from jabber. The solution was to make the capability on the voice gateway 
voice port to voice only.

On Tue, Nov 24, 2015 at 10:34 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

It sounds like your solution may be changing or removing application dial 
rules. Application dial rules add or remove digits to dialed numbers from 
applications that use CCM as the call control server (e.g Unified 
Communications Manager IM & Presence) and do not function like traditional 
patterns. After an application dial rule applies its rule logic to a remote 
(called) number, that number must still match an egress-able pattern (or other 
type of onnet/offnet pattern/DN).


In practice, I do not use application dial rules for IM & Presence unless I 
have a specific use case; I typically allow the dialed digits from the Jabber 
client to enter CCM's numplan untreated by application dial rules. That said, I 
would not remove any of your application dial rules without first understanding 
why they are there and the impact it may have on other applications if they 
were to be removed.


The other thing to keep in mind about application dial rules is that Call 
Manager does not use the best match algorithm that it uses with other patterns, 
it applies the first dialing rule matched, which is determined by the 
application dial rule's priority.


You can research more on the topic on page 175 of 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100.pdf


Hope this helps,


= Ryan =



Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: abbas wali <abba...@gmail.com<mailto:abba...@gmail.com>>
Sent: Tuesday, November 24, 2015 2:05 PM
To: 'Ryan Huff'; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>

Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue


Its set to Standard Analysis.



Something else I have noticed.

There are application dial rules defined. On top (with top priority ) there was 
Default rule beginning with blank, 0 digit strip and append 8.



Some of the traces I found all my dials were appended by 8.



Okay I have now moved the default dial rule to the bottom and all the correct 
one are on top.



Now I can dial internally across cluster which is good. But cant dial external



If that’s the case and have to define full dial plan in the app dial rule that 
will become quiet messy.







From: Ryan Huff [mailto:ryanh...@outlook.com<mailto:ryanh...@outlook.com>]
Sent: 24 November 2015 18:30
To: abbas wali <abba...@gmail.com<mailto:abba...@gmail.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue



Also and if it isn't too late, before you pull the SDL trace on the test call, 
can you verify that the Digit Complexity Analysis is set to 
TranslationAndAlternatePatternAnalysis under Service Parameters->Cisco Call 
Manager?



= Ryan =





Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



____

From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Ryan Huff <ryanh...@outlook.com<mailto:ryanh...@outlook.com>>
Sent: Tuesday, November 24, 2015 1:00 PM
To: abbas wali; cisco-voip@puck.nether.

Re: [cisco-voip] Jabber phone mode outbound calls issue

[Ryan Huff]

Also and if it isn't too late, before you pull the SDL trace on the test call, 
can you verify that the Digit Complexity Analysis is set to 
TranslationAndAlternatePatternAnalysis under Service Parameters->Cisco Call 
Manager?


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Ryan Huff 
<ryanh...@outlook.com>
Sent: Tuesday, November 24, 2015 1:00 PM
To: abbas wali; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue

I would deffinatley look into getting the clock sync to a strata 3 on the pub 
and then restart ntp services.

Can you do a test call on one of the Jabber clients and pull of the SDL traces 
for the call?


Sent from my T-Mobile 4G LTE Device


 Original message 
From: abbas wali
Date:11/24/2015 12:13 PM (GMT-05:00)
To: 'Ryan Huff' ,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue

Hi Ryan,

Thanks for the detailed response.

Yes the issue is with Jabber clients and not the IP phones.

The line itself which is shared with many devices, can make calls on any other 
device but fails when made from Jabber.

I ran all the below Utils and all came out without any significant alarms
The NTP, though is at stratum 4. But again that’s for both the clusters and one 
of them can make calls with jabber.


Have ran some traces as below
These are multiple failed calls.
Not sure why there are so many REFER messages !!

Thank s

[cid:_com_android_email_attachmentprovider_2_2652_RAW@sec.galaxytab]

From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 24 November 2015 15:32
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue


I assume this is only with the Jabber clients and not IP phones as well?



The annunciation message you're getting from Call Manager is typically reserved 
for when the calling device does not have access to the called device (or 
pattern). If you're confident that you're CSS/Partitions are correct you may 
need to look at OS level items.



I recently assisted someone who presented with similar symptoms; everything 
worked fine except Jabber client egress and the solution there was NTP 
(incorrect/unsupported NTP can cause very, very strange behavior in UCOS).



I would give the cluster a quick health check (performed from the CLI of the 
publisher);



  *   utils dbreplication runtimestate

 *   Looking for everything to come back with a (2) Setup Completedi 
message in the Replication Setup column

  *   utils diagnose module validate_network

 *   Looking for it to come back with Passed (anything fails like reverse 
DNS ... etc and it will explain)

  *   utils ntp status

 *   Looking for it to show synchronized and a stratum 3 (or lower)

*   Windows servers (SNTP) are unsupported for NTP and may cause issues 
even if it shows synchronized

  *   utils ntp server list

 *   Looking for any ntp servers referenced by hostname/FQDN rather than IP 
address (you should reference ntp servers by IP address)

If everything comes back healthy, I would setup a test call scenario and pull 
traces off of CCM and follow the call flow. If one of the health checks fail, I 
would resolve that and then you may have to schedule a cluster restart (if 
possible).


= Ryan =





Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>


From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of abbas wali <abba...@gmail.com<mailto:abba...@gmail.com>>
Sent: Tuesday, November 24, 2015 9:58 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Jabber phone mode outbound calls issue


Hi all,



Jabber phone only mode (10.5.2) is unable to make any outbound calls including 
any internal calls even to reach the voicemail.

Inbound calls are working.



This is happening in CUCM 9.1



When dial anything , I get  the “your call cannt be completed as dialled please 
consult…”



I have checked via the DNA and the line settings are okay and calls permitted. 
Hence the CSSs\DP are okay.





Strangely, we have another cluster CM 9.1 with the same jabber version and 
setting and it has no issues making any calls.



Any sugges

Re: [cisco-voip] Jabber phone mode outbound calls issue

[Ryan Huff]

I would deffinatley look into getting the clock sync to a strata 3 on the pub 
and then restart ntp services.

Can you do a test call on one of the Jabber clients and pull of the SDL traces 
for the call?


Sent from my T-Mobile 4G LTE Device


 Original message 
From: abbas wali
Date:11/24/2015 12:13 PM (GMT-05:00)
To: 'Ryan Huff' ,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue

Hi Ryan,

Thanks for the detailed response.

Yes the issue is with Jabber clients and not the IP phones.

The line itself which is shared with many devices, can make calls on any other 
device but fails when made from Jabber.

I ran all the below Utils and all came out without any significant alarms
The NTP, though is at stratum 4. But again that’s for both the clusters and one 
of them can make calls with jabber.


Have ran some traces as below
These are multiple failed calls.
Not sure why there are so many REFER messages !!

Thank s

[cid:_com_android_email_attachmentprovider_2_2652_RAW@sec.galaxytab]

From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 24 November 2015 15:32
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue


I assume this is only with the Jabber clients and not IP phones as well?



The annunciation message you're getting from Call Manager is typically reserved 
for when the calling device does not have access to the called device (or 
pattern). If you're confident that you're CSS/Partitions are correct you may 
need to look at OS level items.



I recently assisted someone who presented with similar symptoms; everything 
worked fine except Jabber client egress and the solution there was NTP 
(incorrect/unsupported NTP can cause very, very strange behavior in UCOS).



I would give the cluster a quick health check (performed from the CLI of the 
publisher);



  *   utils dbreplication runtimestate

 *   Looking for everything to come back with a (2) Setup Completedi 
message in the Replication Setup column

  *   utils diagnose module validate_network

 *   Looking for it to come back with Passed (anything fails like reverse 
DNS ... etc and it will explain)

  *   utils ntp status

 *   Looking for it to show synchronized and a stratum 3 (or lower)

*   Windows servers (SNTP) are unsupported for NTP and may cause issues 
even if it shows synchronized

  *   utils ntp server list

 *   Looking for any ntp servers referenced by hostname/FQDN rather than IP 
address (you should reference ntp servers by IP address)

If everything comes back healthy, I would setup a test call scenario and pull 
traces off of CCM and follow the call flow. If one of the health checks fail, I 
would resolve that and then you may have to schedule a cluster restart (if 
possible).


= Ryan =





Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>


From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of abbas wali <abba...@gmail.com<mailto:abba...@gmail.com>>
Sent: Tuesday, November 24, 2015 9:58 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Jabber phone mode outbound calls issue


Hi all,



Jabber phone only mode (10.5.2) is unable to make any outbound calls including 
any internal calls even to reach the voicemail.

Inbound calls are working.



This is happening in CUCM 9.1



When dial anything , I get  the “your call cannt be completed as dialled please 
consult…”



I have checked via the DNA and the line settings are okay and calls permitted. 
Hence the CSSs\DP are okay.





Strangely, we have another cluster CM 9.1 with the same jabber version and 
setting and it has no issues making any calls.



Any suggestions.



Thanks
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Re: [cisco-voip] Call Queue while hunting

[Ryan Huff]

What I have done in the past (admittedly, not the most elegant); as Brian 
mentions, is send the ingress call to a Call Handler in CUC via CTI Route Point 
to play the message and then set the after greeting action to transfer to the 
hunt pilot and use the CTI Route Point for the hunt pilot's CFNA/CFB treatment 
options. Assuming you have access to Unity Connections though.




= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com



From: cisco-voip  on behalf of Brian Meade 

Sent: Tuesday, November 24, 2015 11:23 AM
To: Bob Fronk
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Call Queue while hunting

You could play the message with Unity Connection.

On Tue, Nov 24, 2015 at 11:18 AM, Bob Fronk 
> wrote:
We are being asked to create a method for the following:

Caller dials a DN - DN is answered with "all technicians are currently 
assisting other callers, please wait for the next available technician" - Hunt 
group starts - a technician answers - call is transferred to technician.

We tried the native call queuing, but this only works if all technicians are 
actually on a call.  We want the call queued even if everyone is available, and 
let the hunt start until someone actually answers.

Is this possible without 3rd party tools?

Bob Fronk
Manager Information Systems
Davis H. Elliot Construction Company, Inc.
*Note: We have a new email address domain. Please add 
@dhec.com to your whitelists and update your contact lists* 
This email and any attached files are confidential and intended solely for the 
intended recipient(s). If you are not the named recipient you should not read, 
distribute, copy or alter this email. Any views or opinions expressed in this 
email are those of the author and do not represent those of the Davis H. Elliot 
Company. Warning: Although precautions have been taken to make sure no viruses 
are present in this email, the company cannot accept responsibility for any 
loss or damage that arise from the use of this email or attachments. {dhe*}

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Re: [cisco-voip] Jabber phone mode outbound calls issue

[Ryan Huff]

It sounds like your solution may be changing or removing application dial 
rules. Application dial rules add or remove digits to dialed numbers from 
applications that use CCM as the call control server (e.g Unified 
Communications Manager IM & Presence) and do not function like traditional 
patterns. After an application dial rule applies its rule logic to a remote 
(called) number, that number must still match an egress-able pattern (or other 
type of onnet/offnet pattern/DN).


In practice, I do not use application dial rules for IM & Presence unless I 
have a specific use case; I typically allow the dialed digits from the Jabber 
client to enter CCM's numplan untreated by application dial rules. That said, I 
would not remove any of your application dial rules without first understanding 
why they are there and the impact it may have on other applications if they 
were to be removed.


The other thing to keep in mind about application dial rules is that Call 
Manager does not use the best match algorithm that it uses with other patterns, 
it applies the first dialing rule matched, which is determined by the 
application dial rule's priority.


You can research more on the topic on page 175 of 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100.pdf


Hope this helps,


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>



From: abbas wali <abba...@gmail.com>
Sent: Tuesday, November 24, 2015 2:05 PM
To: 'Ryan Huff'; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue


Its set to Standard Analysis.



Something else I have noticed.

There are application dial rules defined. On top (with top priority ) there was 
Default rule beginning with blank, 0 digit strip and append 8.



Some of the traces I found all my dials were appended by 8.



Okay I have now moved the default dial rule to the bottom and all the correct 
one are on top.



Now I can dial internally across cluster which is good. But cant dial external



If that’s the case and have to define full dial plan in the app dial rule that 
will become quiet messy.







From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 24 November 2015 18:30
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue



Also and if it isn't too late, before you pull the SDL trace on the test call, 
can you verify that the Digit Complexity Analysis is set to 
TranslationAndAlternatePatternAnalysis under Service Parameters->Cisco Call 
Manager?



= Ryan =





Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>





From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Ryan Huff <ryanh...@outlook.com<mailto:ryanh...@outlook.com>>
Sent: Tuesday, November 24, 2015 1:00 PM
To: abbas wali; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue



I would deffinatley look into getting the clock sync to a strata 3 on the pub 
and then restart ntp services.



Can you do a test call on one of the Jabber clients and pull of the SDL traces 
for the call?





Sent from my T-Mobile 4G LTE Device


---- Original message 
From: abbas wali
Date:11/24/2015 12:13 PM (GMT-05:00)
To: 'Ryan Huff' ,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue

Hi Ryan,



Thanks for the detailed response.



Yes the issue is with Jabber clients and not the IP phones.



The line itself which is shared with many devices, can make calls on any other 
device but fails when made from Jabber.



I ran all the below Utils and all came out without any significant alarms

The NTP, though is at stratum 4. But again that’s for both the clusters and one 
of them can make calls with jabber.





Have ran some traces as below

These are multiple failed calls.

Not sure why there are so many REFER messages !!



Thank s



[cid:image001.png@01D126EA.C20EFD20]



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 24 November 2015 15:32
To: abbas wali <abba...@gmail.com<mailto:abba...@gmail.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Jabber phone mode outbound cal

Re: [cisco-voip] Jabber phone mode outbound calls issue

[Ryan Huff]

I assume this is only with the Jabber clients and not IP phones as well?


The annunciation message you're getting from Call Manager is typically reserved 
for when the calling device does not have access to the called device (or 
pattern). If you're confident that you're CSS/Partitions are correct you may 
need to look at OS level items.


I recently assisted someone who presented with similar symptoms; everything 
worked fine except Jabber client egress and the solution there was NTP 
(incorrect/unsupported NTP can cause very, very strange behavior in UCOS).


I would give the cluster a quick health check (performed from the CLI of the 
publisher);


  *   utils dbreplication runtimestate
 *   Looking for everything to come back with a (2) Setup Completedi 
message in the Replication Setup column

  *   utils diagnose module validate_network
 *   Looking for it to come back with Passed (anything fails like reverse 
DNS ... etc and it will explain)

  *   utils ntp status
 *   Looking for it to show synchronized and a stratum 3 (or lower)
*   Windows servers (SNTP) are unsupported for NTP and may cause issues 
even if it shows synchronized

  *   utils ntp server list
 *   Looking for any ntp servers referenced by hostname/FQDN rather than IP 
address (you should reference ntp servers by IP address)

If everything comes back healthy, I would setup a test call scenario and pull 
traces off of CCM and follow the call flow. If one of the health checks fail, I 
would resolve that and then you may have to schedule a cluster restart (if 
possible).


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com



From: cisco-voip  on behalf of abbas wali 

Sent: Tuesday, November 24, 2015 9:58 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber phone mode outbound calls issue


Hi all,



Jabber phone only mode (10.5.2) is unable to make any outbound calls including 
any internal calls even to reach the voicemail.

Inbound calls are working.



This is happening in CUCM 9.1



When dial anything , I get  the "your call cannt be completed as dialled please 
consult..."



I have checked via the DNA and the line settings are okay and calls permitted. 
Hence the CSSs\DP are okay.





Strangely, we have another cluster CM 9.1 with the same jabber version and 
setting and it has no issues making any calls.



Any suggestions.



Thanks
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Re: [cisco-voip] Call to Huntgroup drop when Group only contains csf

[Ryan Huff]

Hi Dan!


Thanks for correcting me. For some reason I remembered it working differently? 
Anyway, my apologies to the list, I will work to improve [] .


= Ryan =



From: Daniel Pagan <dpa...@fidelus.com>
Sent: Monday, November 23, 2015 9:39 AM
To: Ryan Huff; rschukne...@gmx.de; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Call to Huntgroup drop when Group only contains csf


Just a heads up, the Use Forward Settings of Line Group Member should use the 
forward no coverage settings of the DN forwarding to the Hunt Pilot. No need to 
have an active line member in the LG. Just ran a quick test to confirm - the 
only member of a LG was logged out with the device unregistered and no coverage 
continued to work.



Not sure why this feature got renamed to “of Line Group Member” - it should 
have kept its original name instead (“use personal prefs”) considering how much 
confusion it caused when it was first changed.



- Dan



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Sunday, November 22, 2015 7:19 PM
To: rschukne...@gmx.de; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Call to Huntgroup drop when Group only contains csf



For the "Forward Hunt No Answer" setting on the hunt pilot did you select "Use 
Forward Settings of Line Group Member" or "Forward Unanswered Calls to 
Destination"? If you've selected "Use Forward Settings of Line Group Member", 
then an active line member will need to be present in the line group.



= Ryan =





Email: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Spark: ryanthomash...@outlook.com<mailto:ryanthomash...@outlook.com>

Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>

LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>

Web ryanthomashuff.com<http://ryanthomashuff.com>





From: cisco-voip 
<cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of rschukne...@gmx.de<mailto:rschukne...@gmx.de> 
<rschukne...@gmx.de<mailto:rschukne...@gmx.de>>
Sent: Sunday, November 22, 2015 4:40 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Call to Huntgroup drop when Group only contains csf





I am facing a strange Problem with Huntgroups which are containing only CSF 
(Client Services Framework) devices.

The huntpilot is configured to forward calls on busy and no Answer to a 
voicemail-Pilot. Which is working when at least one csf device is registered. 
It does not work when no csf device is registered. Then a call for the 
Huntgroup is silently dropped. No busy tone or announcement.

We are using CUCM Version 10.5.2 SU2.

Any Hints See welcome!





Regards,

Robert


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Re: [cisco-voip] Call to Huntgroup drop when Group only contains csf

[Ryan Huff]

For the "Forward Hunt No Answer" setting on the hunt pilot did you select "Use 
Forward Settings of Line Group Member" or "Forward Unanswered Calls to 
Destination"? If you've selected "Use Forward Settings of Line Group Member", 
then an active line member will need to be present in the line group.


= Ryan =



Email: ryanthomash...@outlook.com

Spark: ryanthomash...@outlook.com

Twitter: @ryanthomashuff

LinkedIn: ryanthomashuff

Web ryanthomashuff.com



From: cisco-voip  on behalf of 
rschukne...@gmx.de 
Sent: Sunday, November 22, 2015 4:40 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Call to Huntgroup drop when Group only contains csf



I am facing a strange Problem with Huntgroups which are containing only CSF 
(Client Services Framework) devices.

The huntpilot is configured to forward calls on busy and no Answer to a 
voicemail-Pilot. Which is working when at least one csf device is registered. 
It does not work when no csf device is registered. Then a call for the 
Huntgroup is silently dropped. No busy tone or announcement.

We are using CUCM Version 10.5.2 SU2.

Any Hints See welcome!



Regards,

Robert

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Re: [cisco-voip] Restart CUIC

[Ryan Huff]

Here is a second (or third perhaps) opinion, for what it is worth. "You can't 
break broke".

As Anthony said,  you are not impacting call processing and the service restart 
will be transparent to your users, reporting users not withstanding, obviously.


 Original message 
From: Anthony Holloway
Date:11/20/2015 3:05 PM (GMT-05:00)
To: Jose Colon II
Cc: Cisco VOIP
Subject: Re: [cisco-voip] Restart CUIC

2nd opinion from a stranger on the internet?

On Fri, Nov 20, 2015 at 2:04 PM, Jose Colon II 
> wrote:
I concur. Unfortunately my boss didn't and wanted me to get a 2nd opinion lol.

Thanks!!
Jose

On Fri, Nov 20, 2015 at 1:44 PM, Anthony Holloway 
> wrote:
I say, if you need to restart it, then restart it.  It's just reporting, so no 
calls will be impacted, and since the call details are not written to, nor 
controlled by CUIC, you wouldn't lose data while the service is stopped.

utils service restart Cisco Unified Intelligence Center Reporting Service

It should go without saying that this would impact any reporting abilities 
during the restart, but if you're restarting because it's not working, then 
what's the harm, right?

On Fri, Nov 20, 2015 at 10:37 AM, Jose Colon II 
> wrote:
Just a quick question, Can I restart the CUIC reporting service during 
production hours. Cant find any info on this.

Thanks

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Re: [cisco-voip] Restart CUIC

[Ryan Huff]

Dan is and continues to be one of my voices of reason!

Yes, the "can't break broke" approach is very cowboy'ish (which often times can 
be dangerous in the wrong hands and can lead to resume generating events); 
definitely know what is at stake before consuming that mantra. However, in the 
context of the CUIC reporting service restart, I don't the you're risking a 
great deal.

Big picture wise, I completely agree with Dan, you must understand everything 
associate to what you are doing (or about to do) and the propensity it has to 
make the issue worse or impact ancillary services.

= Ryan =

On Nov 20, 2015, at 4:36 PM, Daniel Pagan 
<dpa...@fidelus.com<mailto:dpa...@fidelus.com>> wrote:

Not to mention further impact caused by a “can’t break what’s broken” approach 
without having full knowledge of additional dependencies. I would add a third 
to your list:

3) Do I understand all the dependencies of this service I’m restarting, trunk 
I’m resetting, server I’m rebooting, or configuration pool I’m restarting? 
Further, am I certain the work I’m about to perform will have either a positive 
or, at least, a neutral effect on the impacted device or software. If not, 
repeat questions #1 and #2.

;)

- Dan


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Friday, November 20, 2015 3:34 PM
To: jcolon...@gmail.com<mailto:jcolon...@gmail.com>
Cc: Cisco VOIP <cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Restart CUIC

Amendment:

Well, You can break almost broke; at that point I would level it against 
business impact with two questions.

1.) Is this impacting the business (or capacity to do business) in a manner 
that dictates immediate resolution or can it wait till a scheduled maintenance 
window?

2.) What is the impact if I do nothing and the problem progresses, will there 
be other things impacted that will change my answer to question #1?

Thanks,

Ryan
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Re: [cisco-voip] MRA and Phone server selection

[Ryan Huff]

In your Expressway-C server, what do you have listed as your CUCM Neighbors?

On Nov 19, 2015, at 6:27 AM, Matthew Collins 
> wrote:

Hi All,

It seems that our phone services Via MRA are somewhat intermittent, It seems 
that the clients are trying to register with CUCM servers that are not running 
the CCM service. When I check servers under phone services the client seems to 
get a different pair each time the application is launched. When the servers 
reference the one of the two servers running CCM phone services connect. When a 
CCM server isn't referenced phone services fail to connect. Thought I'd just 
check here before opening a Tac case.

CUCM is 10.5.2
I have tested with Various Jabber clients 10.5, 10.6, 11.0 and 11.1 across 
multiple versions of Windows, Ipad, Iphone clients
Expressway is 8.5.1

Within this cluster we have

1 CUCM publisher
2 CUCM subscribers running TFTP
2 CUCM Subscribers running Media resources only (we use a lot of MTP for 3rd 
Party voicemail)
2 CUCM Subscribers running Call manger
2 IM Subscribers forming 1 presence cluster

I only have the 2 x CUCM Subscribers running Call manger set up as my CTI and 
Directory UC services, 2 of the IM Subscribers set up as my IM services.

They have a dedicated Device pool for Jabber clients and that only has the 2 x 
CUCM Subscribers running Call manger within the Call manager group.

The problem seems to be that when we log in via MRA the jabber client is 
attempting to register will any two of the 9 CUCM servers. It only successfully 
registers when one of the two servers are one of the one running CUCM. Presence 
and directory services (UDS) all ways register with the correct servers


Thanks in advance.

Matthew
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Re: [cisco-voip] MRA and Phone server selection

[Ryan Huff]

I remember an Expressway 8.5 deployment once; on the inside everything was fine 
but on the outside, my clients would only ever register with one CCM node 
(although there were three discovered).

In my case, it was a reboot of Expressway-C; not a software restart but a full 
VM reboot and then everything worked fine for me.

Thanks,

Ryan



On Nov 19, 2015, at 6:55 AM, Matthew Collins 
<mcoll...@block.co.uk<mailto:mcoll...@block.co.uk>> wrote:

Hi Ryan,

Just what the Expressway picked up during auto discovery.

That is the 2 subs running CCM and the 2 Subs running TFTP

Regards

Matthew





From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 19 November 2015 11:37
To: Matthew Collins <mcoll...@block.co.uk<mailto:mcoll...@block.co.uk>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] MRA and Phone server selection

In your Expressway-C server, what do you have listed as your CUCM Neighbors?
On Nov 19, 2015, at 6:27 AM, Matthew Collins 
<mcoll...@block.co.uk<mailto:mcoll...@block.co.uk>> wrote:
Hi All,

It seems that our phone services Via MRA are somewhat intermittent, It seems 
that the clients are trying to register with CUCM servers that are not running 
the CCM service. When I check servers under phone services the client seems to 
get a different pair each time the application is launched. When the servers 
reference the one of the two servers running CCM phone services connect. When a 
CCM server isn't referenced phone services fail to connect. Thought I'd just 
check here before opening a Tac case.

CUCM is 10.5.2
I have tested with Various Jabber clients 10.5, 10.6, 11.0 and 11.1 across 
multiple versions of Windows, Ipad, Iphone clients
Expressway is 8.5.1

Within this cluster we have

1 CUCM publisher
2 CUCM subscribers running TFTP
2 CUCM Subscribers running Media resources only (we use a lot of MTP for 3rd 
Party voicemail)
2 CUCM Subscribers running Call manger
2 IM Subscribers forming 1 presence cluster

I only have the 2 x CUCM Subscribers running Call manger set up as my CTI and 
Directory UC services, 2 of the IM Subscribers set up as my IM services.

They have a dedicated Device pool for Jabber clients and that only has the 2 x 
CUCM Subscribers running Call manger within the Call manager group.

The problem seems to be that when we log in via MRA the jabber client is 
attempting to register will any two of the 9 CUCM servers. It only successfully 
registers when one of the two servers are one of the one running CUCM. Presence 
and directory services (UDS) all ways register with the correct servers


Thanks in advance.

Matthew
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Re: [cisco-voip] MRA and Phone server selection

[Ryan Huff]

Matt,

Apologies if that is the impression I gave you. Expressway neighbors are just 
facilitating the communication between the core and the neighbor(s). I was just 
citing a quasi-similar situation where a core reboot worked for me.

In the configured CCMCIP profile, are all the mra users assigned to it?

In CCM, what is the primary server advertised for CTI Control?

Thanks,

Ryan

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Matthew Collins
Date:11/19/2015 9:02 AM (GMT-05:00)
To: Ryan Huff
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] MRA and Phone server selection

Hi Ryan,

There UCS servers were actually powered down last weekend due to power issues 
in the DC so the whole cluster has been power cycled. This issue has been 
ongoing prior to the shutdown but only coming to light as they are rolling 
Jabber out to mobile devices.

It seems odd that the jabber client would be told to register with one of the 
none CCM servers I don’t think this would come from the expressway.

Just been looking through the ipad log files and it looks like the jabber 
client uses the CCMCIP server to register the phone service to. In the first 
instance the CCMCIP servers are referencing the publisher, One of the TFTP 
server and one of the Media Servers.

In the below snippets the 5th character in the server name define their role.

P = Publisher
M =  Media Subscriber
S = CCM subscriber
T = TFTP Subscriber

The Below snippit is from the log file when the phone services don’t register

type=eSIP, isRelevant=true, server=cucms01.Company.co.uk:5060, 
connectionState=eNone, isEncrypted=false, serverType=ePrimary, role=eNone
type=eSIP, isRelevant=true, server=cucms02.Company.co.uk:5060, 
connectionState=eNone, isEncrypted=false, serverType=eSecondary, role=eNone
type=eUDS, isRelevant=true, vcse=expresswaye01.Company.co.uk:8443, 
server=cucmp01.Company.co.uk:8443, connectionState=eConnected, isEncrypted=true
type=eUDS, isRelevant=true, server=cucmm01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eUDS, isRelevant=true, server=cucmt01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eCCMCIP, isRelevant=true, server=cucmp01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eCCMCIP, isRelevant=true, server=cucmm01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eCCMCIP, isRelevant=true, server=cucmt01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmp01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmp01.Company.co.uk:8080, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmm01.Company.nhs.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmm01.Company.co.uk:8080, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmt01.Company.co.uk:8443, 
connectionState=eNotApplicable, isEncrypted=false
type=eEMAPI, isRelevant=true, server=cucmt01.Company.co.uk:8080, 
connectionState=eNotApplicable, isEncrypted=false
type=eConfigFile, isRelevant=true, vcse=expresswaye01.Company.co.uk:8443, 
server=cucmt01.Company.co.uk:6970, connectionState=eConnected, isEncrypted=true
type=eConfigFile, isRelevant=true, server=cucmt01.Company.co.uk:69, 
connectionState=eNotApplicable, isEncrypted=false
type=eConfigFile, isRelevant=true, server=cucmt02.Company.co.uk:6970, 
connectionState=eNotApplicable, isEncrypted=false
type=eConfigFile, isRelevant=true, server=cucmt02.Company.co.uk:69, 
connectionState=eNotApplicable, isEncrypted=false



14:02:25,198 INFO  [l/src/callcontrol/Authenticator.cpp(602)] [csf.ecc] 
[getLastCCMCIPServerUsed] - getLastCCMCIPServerUsed() = cucmp01.Company.co.uk
14:02:25,198 INFO  [/call/YLCTelephonyAuthentication.mm(308)] [UI.Telephony] 
[-[YLCTelephonyAuthentication onAuthenticationStatusChanged:]] - JCF Telephony 
service authentication status InProgress.
14:02:25,199 INFO  [fecycle/YLCAccountStatusPresenter.m(203)] 
[UI.Lifecycle.Login] [-[YLCAccountStatusPresenter 
cucmConnectionStatusChanged:]] - -[YLCAccountStatusPresenter 
cucmConnectionStatusChanged:] -- before changed, the cucm status is 2
14:02:25,200 INFO  [/impl/system/UserProfileManager.cpp(131)] 
[UserProfileManager] [getCredentialsForAuthenticator] -  for authenticator: 2100
14:02:25,200 INFO  [/impl/system/UserProfileManager.cpp(131)] 
[UserProfileManager] [getCredentialsForAuthenticator] -  for authenticator: 2100
14:02:25,205 INFO  [0x00016ed37000] 
[/tahiti/module/util/YLCMemoryInfo.mm(74)] [UI.Util] [-[YLCMemoryInfo 
logMemUsage:]] - (MEMORY MONITOR IN FUNCTION)-->: -[YLCTelephonyAuthentication 
onAuthenticationStatusChanged:] |current memory usage is 80.3 MB
14:02:25,230 INFO  [fecycle/YLCAccountStatusPresenter.m(205)] 
[UI.Lifecycle.Lo

Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Hi Jonathan,

Glad to hear a reboot on the publisher seems to have fixed your initial issue, 
I thought it might. You never know with reboots on those old MCS boxes though, 
so having a good DRS and recover plan prior is key.

I suspect, and if you are so inclined, "show hardware" on the CLI of that pub 
would show that write-cache is enabled once again.

Regarding the failure to connect to subscribers; one of the metrics used during 
UCOS installation to determine connectivity and reachability is response time 
and network visibility.  So if something takes to long to respond, the 
installer will say it isn't reachable but you can choose to ignore and 
installation will continue. Any latency deficit may, and likely will, cause 
issues in the cluster post install, but you can still complete the installation.

Another thought would be visibility, if the installer can't see all the hops 
between nodes (assuming nodes are on different segments) it will say it can't 
connect during installation but you can choose to ignore and installation will 
continue. Routing may work just fine between the nodes but the installer will 
error if there are blind hops between nodes or it can't see the complete end to 
end path between nodes (a traceroute between nodes is a quick way to check). 
This is another item that may cause issues post install.

-Ryan

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Jonathan Charles
Date:11/18/2015 9:02 AM (GMT-05:00)
To: Dave Wolgast
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...

A reboot of the Publisher fixed the time out issue after that the migration 
was successful... interestingly, the installation of subscribers fails due to 
'no network connection to the node'.. however, you can open its webpage, and 
its media resources are registered... I just hit continue... dbreplication 
looks fine.

Jonathan

On Tue, Nov 17, 2015 at 2:30 PM, Dave Wolgast 
> wrote:
On Tue, Nov 17, 2015 at 2:44 PM Heim, Dennis 
> wrote:
Are you running the latest rev of PCD? I ran into the issue, and I believe 
upgrading to the latest rev resolved the issue.
Thanks Dennis. A colleague suggested that as well and I am in the process of 
upgrading to PCD 11 now.

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[cisco-voip] dCloud FYI

[Ryan Huff]

If anyone is having issues with the Americas DC for dCloud; I just called and 
it seems one of the vCenters is having issues. Anyway, they are working on it. 
The specific symptom is that you'll start a content session and it will go from 
"Starting" to "Error" and then the session will get stuck in your "My 
Dashboard" session.


Not sure of the impact or if they'll clear once the dCloud team resolves the 
issue but I had to call to get my sessions released.


-Ryan
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Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Jonathan,

What timesout on the publisher? Are you referencing when PCD tries to do the 
cluster discovery on the existing cluster?

-Ryan


Sent from my T-Mobile 4G LTE Device


 Original message 
From: Jonathan Charles
Date:11/17/2015 12:06 PM (GMT-05:00)
To: Anthony Holloway
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...

Nope, not all good... new error... the upgrade failed so I deleted the cluster 
and re-added... it finds all of the subscribers, but it times out on the 
publisher

I have verified all services are running on the Pub and it looks clean...



Jonathan

On Mon, Nov 16, 2015 at 10:20 AM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
Looks like you're all good now, but as a heads up to everyone else, don't stop 
at checking NTP with "utils ntp status".  You will fail to upgrade if your NTP 
configuration has an FQHN for the NTP server which begins with a digit.

E.g., 0.pool.ntp.org<http://0.pool.ntp.org>

You will not see the hostname in the output of "utils ntp status", as it will 
only show you the resolved IP address.  So, you will also need to issue a 
"utils ntp config" to see what value was entered by the administrator.

This is the only defect reference I found, though my upgrade I hit it on was an 
8.6 to 10.5 Refresh Upgrade (RU) (Not PCD).

https://tools.cisco.com/bugsearch/bug/CSCtj07817

On Sun, Nov 15, 2015 at 10:43 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
OK, a reboot of CPCD got it passed that error...


Jonathan

On Sun, Nov 15, 2015 at 9:40 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
Yeah, the error I am getting is:

1 nodes(s) in Export task action ID #1127... on the Publisher...

I will try rebooting everything...



Jonathan

On Sun, Nov 15, 2015 at 9:35 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Looks healthy ...

I recall trying PCD once and I hit really strange issues too. For that upgrade, 
I ultimately abandoned PCD and built new VMs with the Answer File Generator 
then a DRS backup/restore.

Not sure where you are in your timeline or if it is that important but it is 
definitely something I would consider. Sometimes you can spend more time trying 
to get the silly tools to work, than to just do the work yourself.

Google is littered with PCD weirdness; great idea of an application, just not 
there yet IMO.

-Ryan



Sent from my iPad
On Nov 15, 2015, at 10:19 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

Everything looks good


admin:utils ntp status
ntpd (pid 19674) is running...

 remote   refid  st t when poll reach   delay   offset  jitter
==
 127.127.1.0 .LOCL.  10 l   21   64  3770.0000.000   0.001
 10.0.31.2   10.0.31.33 u  175 1024  3770.635   -5.771   2.042
*10.0.31.3   129.6.15.29  2 u  970 1024  3770.510  -11.340   0.449
 10.1.31.2   10.0.31.33 u  490 1024  3770.850   -9.114   4.881
+10.1.31.3   129.6.15.29  2 u  184 1024  3770.817   -4.085   5.355


synchronised to NTP server (10.0.31.3) at stratum 3
   time correct to within 68 ms
   polling server every 1024 s

Current time in UTC is : Mon Nov 16 03:16:14 UTC 2015
Current time in America/Chicago is : Sun Nov 15 21:16:14 CST 2015
admin:

admin:utils diagnose module validate_network

Log file: platform/log/diag1.log

Starting diagnostic test(s)
===
test - validate_network: Passed

Diagnostics Completed

admin:#

admin:utils dbreplication runtimestate

DB and Replication Services: ALL RUNNING

Cluster Replication State: Replication repair command started at: 
2014-06-20-23-22
 Replication repair command COMPLETED 541 tables processed out of 541
 Errors or Mismatches Were Found:

 Use 'file view activelog 
cm/trace/dbl/sdi/ReplicationRepair.2014_06_20_23_22_51.out' to see the details

DB Version: ccm8_6_2_2_2
Number of replicated tables: 541

Cluster Detailed View from PUB (5 Servers):

PINGREPLICATION REPL.   DBver&  
REPL.   REPLICATION SETUP
SERVER-NAME IP ADDRESS  (msec)  RPC?STATUS  QUEUE   TABLES  
LOOP?   (RTMT) & details
--- --  --- -   --- 
-   -
IPTCMS02  10.0.126.12 0.196   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMS01  10.0.126.11 0.151   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMP10.0.126.10 0.065   Yes Connected   0   match   Yes   
  (2) PUB Setup Completed
IPTCMS03  10.1.126.13   

Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Jonathan,

What timesout on the publisher? Are you referencing when PCD tries to do the 
cluster discovery on the existing cluster?

-Ryan


Sent from my T-Mobile 4G LTE Device


 Original message 
From: Jonathan Charles
Date:11/17/2015 12:06 PM (GMT-05:00)
To: Anthony Holloway
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...

Nope, not all good... new error... the upgrade failed so I deleted the cluster 
and re-added... it finds all of the subscribers, but it times out on the 
publisher

I have verified all services are running on the Pub and it looks clean...



Jonathan

On Mon, Nov 16, 2015 at 10:20 AM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
Looks like you're all good now, but as a heads up to everyone else, don't stop 
at checking NTP with "utils ntp status".  You will fail to upgrade if your NTP 
configuration has an FQHN for the NTP server which begins with a digit.

E.g., 0.pool.ntp.org<http://0.pool.ntp.org>

You will not see the hostname in the output of "utils ntp status", as it will 
only show you the resolved IP address.  So, you will also need to issue a 
"utils ntp config" to see what value was entered by the administrator.

This is the only defect reference I found, though my upgrade I hit it on was an 
8.6 to 10.5 Refresh Upgrade (RU) (Not PCD).

https://tools.cisco.com/bugsearch/bug/CSCtj07817

On Sun, Nov 15, 2015 at 10:43 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
OK, a reboot of CPCD got it passed that error...


Jonathan

On Sun, Nov 15, 2015 at 9:40 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
Yeah, the error I am getting is:

1 nodes(s) in Export task action ID #1127... on the Publisher...

I will try rebooting everything...



Jonathan

On Sun, Nov 15, 2015 at 9:35 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Looks healthy ...

I recall trying PCD once and I hit really strange issues too. For that upgrade, 
I ultimately abandoned PCD and built new VMs with the Answer File Generator 
then a DRS backup/restore.

Not sure where you are in your timeline or if it is that important but it is 
definitely something I would consider. Sometimes you can spend more time trying 
to get the silly tools to work, than to just do the work yourself.

Google is littered with PCD weirdness; great idea of an application, just not 
there yet IMO.

-Ryan



Sent from my iPad
On Nov 15, 2015, at 10:19 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

Everything looks good


admin:utils ntp status
ntpd (pid 19674) is running...

 remote   refid  st t when poll reach   delay   offset  jitter
==
 127.127.1.0 .LOCL.  10 l   21   64  3770.0000.000   0.001
 10.0.31.2   10.0.31.33 u  175 1024  3770.635   -5.771   2.042
*10.0.31.3   129.6.15.29  2 u  970 1024  3770.510  -11.340   0.449
 10.1.31.2   10.0.31.33 u  490 1024  3770.850   -9.114   4.881
+10.1.31.3   129.6.15.29  2 u  184 1024  3770.817   -4.085   5.355


synchronised to NTP server (10.0.31.3) at stratum 3
   time correct to within 68 ms
   polling server every 1024 s

Current time in UTC is : Mon Nov 16 03:16:14 UTC 2015
Current time in America/Chicago is : Sun Nov 15 21:16:14 CST 2015
admin:

admin:utils diagnose module validate_network

Log file: platform/log/diag1.log

Starting diagnostic test(s)
===
test - validate_network: Passed

Diagnostics Completed

admin:#

admin:utils dbreplication runtimestate

DB and Replication Services: ALL RUNNING

Cluster Replication State: Replication repair command started at: 
2014-06-20-23-22
 Replication repair command COMPLETED 541 tables processed out of 541
 Errors or Mismatches Were Found:

 Use 'file view activelog 
cm/trace/dbl/sdi/ReplicationRepair.2014_06_20_23_22_51.out' to see the details

DB Version: ccm8_6_2_2_2
Number of replicated tables: 541

Cluster Detailed View from PUB (5 Servers):

PINGREPLICATION REPL.   DBver&  
REPL.   REPLICATION SETUP
SERVER-NAME IP ADDRESS  (msec)  RPC?STATUS  QUEUE   TABLES  
LOOP?   (RTMT) & details
--- --  --- -   --- 
-   -
IPTCMS02  10.0.126.12 0.196   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMS01  10.0.126.11 0.151   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMP10.0.126.10 0.065   Yes Connected   0   match   Yes   
  (2) PUB Setup Completed
IPTCMS03  10.1.126.13   

Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Just as a note of practice; I wouldn't do anything to a running CCM at the OS 
level during production, at least not without proper maintenance declaration to 
the business unit [] . "A maintenance window a day, keeps the resumes away" 
 lol.


If you've got RAID issues; I would get that solved before doing anything else. 
If DRF services are still working (50/50 chance) I would get a good DRS on that 
cluster ASAP. If DRS doesn't work, hopefully BAT still does, I would do a BAT 
export on everything ASAP.


An upgrade would be the least of my worries at this point. He is a reference to 
the particular error message you show in the 
image.(http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/9_1_1/CUCM_BK_UAEC4331_00_upgrade-guide-cucm-91/CUCM_BK_UAEC4331_00_upgrade-guide-cucm-91_chapter_0110.html#CUP0_RF_WD01249D_00).


-Ryan



From: Jonathan Charles <jonv...@gmail.com>
Sent: Tuesday, November 17, 2015 12:45 PM
To: Dave Goodwin
Cc: Ryan Huff; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...

I did notice this on the Publisher OS Admin - Install/Upgrade page:

[Inline image 1]

I wonder if that's causing it...



Jonathan

On Tue, Nov 17, 2015 at 11:40 AM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
Yep, didn't work... same error. Let me check the other nodes...

On Tue, Nov 17, 2015 at 11:34 AM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
Logged into OS Admin on the pub, and it didn't give me the option to assume 
control... can I run discovery/migration during production? Just want to make 
sure I don't pop anything



Jonathan

On Tue, Nov 17, 2015 at 11:30 AM, Dave Goodwin 
<dave.good...@december.net<mailto:dave.good...@december.net>> wrote:
I had a similar symptom a couple months ago - discovery timed out on one or 
more nodes. I found a supportforum post from another individual having that 
issue, and it appeared that PCD got "stuck" installing the 
ciscocm.ucmap_platformconfig.cop file in the middle. The way to work around it 
was to log into the Software Install/Upgrade page on the affected node(s) and 
you can see you'll have the option to Assume Control of a currently running 
install. Do that, and click through to complete the install. Then try to do the 
discovery again. Unknown if there was an existing bug causing this problem, but 
the above workaround steps solved my problem.

On Tue, Nov 17, 2015 at 12:25 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
I just posted the log... I am not sure from reading it...

discovery node 1617 failed with errorcode 1-3


Jonathan

On Tue, Nov 17, 2015 at 11:23 AM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Jonathan,

What timesout on the publisher? Are you referencing when PCD tries to do the 
cluster discovery on the existing cluster?

-Ryan


Sent from my T-Mobile 4G LTE Device


 Original message 
From: Jonathan Charles
Date:11/17/2015 12:06 PM (GMT-05:00)
To: Anthony Holloway
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] CUCM Upgrade failure...

Nope, not all good... new error... the upgrade failed so I deleted the cluster 
and re-added... it finds all of the subscribers, but it times out on the 
publisher

I have verified all services are running on the Pub and it looks clean...



Jonathan

On Mon, Nov 16, 2015 at 10:20 AM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway+cisco-v...@gmail.com>> wrote:
Looks like you're all good now, but as a heads up to everyone else, don't stop 
at checking NTP with "utils ntp status".  You will fail to upgrade if your NTP 
configuration has an FQHN for the NTP server which begins with a digit.

E.g., 0.pool.ntp.org<http://0.pool.ntp.org>

You will not see the hostname in the output of "utils ntp status", as it will 
only show you the resolved IP address.  So, you will also need to issue a 
"utils ntp config" to see what value was entered by the administrator.

This is the only defect reference I found, though my upgrade I hit it on was an 
8.6 to 10.5 Refresh Upgrade (RU) (Not PCD).

https://tools.cisco.com/bugsearch/bug/CSCtj07817

On Sun, Nov 15, 2015 at 10:43 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
OK, a reboot of CPCD got it passed that error...


Jonathan

On Sun, Nov 15, 2015 at 9:40 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
Yeah, the error I am getting is:

1 nodes(s) in Export task action ID #1127... on the Publisher...

I will try rebooting everything...



Jonathan

On Sun, Nov 15, 2015 at 9:35 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Looks healthy ...

I recal

Re: [cisco-voip] BFL Pickup button

[Ryan Huff]

This doc is a bit dated but describes BLF configuration in relevant detail:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucmbe/rel_notes/6_1_2/cucmbe-rel_note-612/cucmbe-rel_note-612_2.html#wp308676


When you press the BLF button in the alerting state (flashing amber), and you 
hear the reorder tone, it is because it is not associate with any pickup 
groups. In essence, it is trying to pickup a call but can't find a call to 
pickup.


Thanks,


Ryan


From: cisco-voip  on behalf of Haas, Neal 

Sent: Monday, November 16, 2015 10:19 AM
To: 'Michael T. Voity'; 'voip puck'
Subject: Re: [cisco-voip] BFL Pickup button


Why don't you configure it as a Line if you want to pick it up?



Thank you,

Neal Haas





From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Michael T. Voity
Sent: Monday, November 16, 2015 5:42 AM
To: voip puck
Subject: [cisco-voip] BFL Pickup button



Hello,

My system is CUCM 10.5.2.12900-14

I have 1 7962 (67522)  that has 2 BLF buttons on it for 2 other phones, 7962 & 
7942 phones,  (67520 and 67521).   The BLF works fine for when i see them off 
hook and when I press the button to call them.

In the BLF config I clicked "Call Pickup"

On my 7962 (67522)  the amber light flashes when  the other phone is ringing .  
  I press the flashing amber button (67520)  and I get a fast busy and on the 
bottom it says"Key is not active"

I have made sure the lines and phones subscribe-css is correct in all the right 
places.   I am scratching my head

I have stalked around google and the cisco forums and can not figure it out

I did see a post out there that mentions that I need to configure a "Call 
Pickup Group"When that is done, I loose my extensions. unless I'm doing it 
wrong?

Ideas?

Thanks,

-Mike


--

Michael T. Voity

Network Engineer

University of Vermont
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Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Dan: Good reference  Been awhile since I've seen a VM cluster trying to 
sync with the host

Anthony: Very good point []


I think the best way to stay in the good graces of UCOS NTP, IMO, is to use an 
IOS device, synced to an S1 or S2 server via IP . everything else either 
isn't supported or has bugs seems like. That or Cisco starts selling cesium 
clocks.



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Daniel Pagan 
<dpa...@fidelus.com>
Sent: Monday, November 16, 2015 12:05 PM
To: Anthony Holloway; Jonathan Charles
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...


On the topic of NTP and upgrades/migrations, I would advise to also watch out 
for CSCur94973.



https://tools.cisco.com/bugsearch/bug/CSCur94973



- Dan

-



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Monday, November 16, 2015 11:20 AM
To: Jonathan Charles <jonv...@gmail.com>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM Upgrade failure...



Looks like you're all good now, but as a heads up to everyone else, don't stop 
at checking NTP with "utils ntp status".  You will fail to upgrade if your NTP 
configuration has an FQHN for the NTP server which begins with a digit.



E.g., 0.pool.ntp.org<http://0.pool.ntp.org>



You will not see the hostname in the output of "utils ntp status", as it will 
only show you the resolved IP address.  So, you will also need to issue a 
"utils ntp config" to see what value was entered by the administrator.



This is the only defect reference I found, though my upgrade I hit it on was an 
8.6 to 10.5 Refresh Upgrade (RU) (Not PCD).



https://tools.cisco.com/bugsearch/bug/CSCtj07817



On Sun, Nov 15, 2015 at 10:43 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

OK, a reboot of CPCD got it passed that error...





Jonathan



On Sun, Nov 15, 2015 at 9:40 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

Yeah, the error I am getting is:



1 nodes(s) in Export task action ID #1127... on the Publisher...



I will try rebooting everything...







Jonathan



On Sun, Nov 15, 2015 at 9:35 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

Looks healthy ...



I recall trying PCD once and I hit really strange issues too. For that upgrade, 
I ultimately abandoned PCD and built new VMs with the Answer File Generator 
then a DRS backup/restore.



Not sure where you are in your timeline or if it is that important but it is 
definitely something I would consider. Sometimes you can spend more time trying 
to get the silly tools to work, than to just do the work yourself.



Google is littered with PCD weirdness; great idea of an application, just not 
there yet IMO.

-Ryan




Sent from my iPad

On Nov 15, 2015, at 10:19 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

Everything looks good





admin:utils ntp status

ntpd (pid 19674) is running...



 remote   refid  st t when poll reach   delay   offset  jitter

==

 127.127.1.0 .LOCL.  10 l   21   64  3770.0000.000   0.001

 10.0.31.2   10.0.31.33 u  175 1024  3770.635   -5.771   2.042

*10.0.31.3   129.6.15.29  2 u  970 1024  3770.510  -11.340   0.449

 10.1.31.2   10.0.31.33 u  490 1024  3770.850   -9.114   4.881

+10.1.31.3   129.6.15.29  2 u  184 1024  3770.817   -4.085   5.355





synchronised to NTP server (10.0.31.3) at stratum 3

   time correct to within 68 ms

   polling server every 1024 s



Current time in UTC is : Mon Nov 16 03:16:14 UTC 2015

Current time in America/Chicago is : Sun Nov 15 21:16:14 CST 2015

admin:



admin:utils diagnose module validate_network



Log file: platform/log/diag1.log



Starting diagnostic test(s)

===

test - validate_network: Passed



Diagnostics Completed



admin:#



admin:utils dbreplication runtimestate



DB and Replication Services: ALL RUNNING



Cluster Replication State: Replication repair command started at: 
2014-06-20-23-22

 Replication repair command COMPLETED 541 tables processed out of 541

 Errors or Mismatches Were Found:



 Use 'file view activelog 
cm/trace/dbl/sdi/ReplicationRepair.2014_06_20_23_22_51.out' to see the details



DB Version: ccm8_6_2_2_2

Number of replicated tables: 541



Cluster Detailed View from PUB (5 Servers):



PINGREPLICATION REPL.   DBver&  
REPL.   REPLICATION SETUP

SERVER-NAME IP ADDRESS  (msec)  RPC?STATUS  QUEUE   TABLES  
LOOP?   (RTMT) & details

--- -

Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

If the FROM CCM version was unrestricted, it would say "Unrestricted" after the 
version number on the "Active Master Version" line. If it does not say 
"Unrestricted", then it is the more common restricted version.

As to your original issue, I would start with all the usual suspects. Is the 
FROM CCM cluster healthy to start with; dns, ntp, replication ... etc?

>From CCM:

1.) #utils diagnose module validate_network
(Should see Passed)

2.) #utils ntp status
(Pub should be synchronized and Strata 3 or better)

3.) #utils dbreplication runtimestate
(Should see 2 - Setup Completed for all nodes)

If PCD is moving the apps to a new platform/chassis, make sure the target 
environment can reach all the same network assets as the from environment.

Thanks,

Ryan

On Nov 15, 2015, at 9:34 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

It just says the version number...

admin:show version active
Active Master Version: 8.6.2.2-2
Active Version Installed Software Options:
cmterm-7942_7962-sccp.9-3-1ES27-rel.cop
cmterm-devicepack8.6.2.24118-1.cop
ciscocm.refresh_upgrade_v1.1.cop
ciscocm.ucmap_platformconfig.cop
ciscocm.migrate-export-v1.12.cop
admin:


Jonahan

On Sun, Nov 15, 2015 at 7:26 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Did you actually dump the logs to a serial interface (curious what it shows)?

On the FROM CCM, goto the CLI of pub (or sub) and do a "show version active"; 
it will tell you if you have the UNREST.

Thanks,

Ryan

> On Nov 15, 2015, at 7:55 PM, Jonathan Charles 
> <jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
>
> Using PCD on CUCM 10.5.2.11901 got the following error:
>
> 
>
> It seems to imply I am not matching restricted vs. unrestricted...
>
> Any easy way to find out?
>
>
>
> Jonathan
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip

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Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Also worth noting that if CCM NTP is synchronized to a Windows server (even if 
it shows Stratum 3 or better); that is a problem you'll need to correct as SNTP 
can play hell with UCOS and do some pretty weird stuff.

Thanks,

Ryan

On Nov 15, 2015, at 10:06 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

If the FROM CCM version was unrestricted, it would say "Unrestricted" after the 
version number on the "Active Master Version" line. If it does not say 
"Unrestricted", then it is the more common restricted version.

As to your original issue, I would start with all the usual suspects. Is the 
FROM CCM cluster healthy to start with; dns, ntp, replication ... etc?

>From CCM:

1.) #utils diagnose module validate_network
(Should see Passed)

2.) #utils ntp status
(Pub should be synchronized and Strata 3 or better)

3.) #utils dbreplication runtimestate
(Should see 2 - Setup Completed for all nodes)

If PCD is moving the apps to a new platform/chassis, make sure the target 
environment can reach all the same network assets as the from environment.

Thanks,

Ryan

On Nov 15, 2015, at 9:34 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

It just says the version number...

admin:show version active
Active Master Version: 8.6.2.2-2
Active Version Installed Software Options:
cmterm-7942_7962-sccp.9-3-1ES27-rel.cop
cmterm-devicepack8.6.2.24118-1.cop
ciscocm.refresh_upgrade_v1.1.cop
ciscocm.ucmap_platformconfig.cop
ciscocm.migrate-export-v1.12.cop
admin:


Jonahan

On Sun, Nov 15, 2015 at 7:26 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Did you actually dump the logs to a serial interface (curious what it shows)?

On the FROM CCM, goto the CLI of pub (or sub) and do a "show version active"; 
it will tell you if you have the UNREST.

Thanks,

Ryan

> On Nov 15, 2015, at 7:55 PM, Jonathan Charles 
> <jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:
>
> Using PCD on CUCM 10.5.2.11901 got the following error:
>
> 
>
> It seems to imply I am not matching restricted vs. unrestricted...
>
> Any easy way to find out?
>
>
>
> Jonathan
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip

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Re: [cisco-voip] CUCM Upgrade failure...

[Ryan Huff]

Looks healthy ...

I recall trying PCD once and I hit really strange issues too. For that upgrade, 
I ultimately abandoned PCD and built new VMs with the Answer File Generator 
then a DRS backup/restore.

Not sure where you are in your timeline or if it is that important but it is 
definitely something I would consider. Sometimes you can spend more time trying 
to get the silly tools to work, than to just do the work yourself.

Google is littered with PCD weirdness; great idea of an application, just not 
there yet IMO.

-Ryan



Sent from my iPad
On Nov 15, 2015, at 10:19 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

Everything looks good


admin:utils ntp status
ntpd (pid 19674) is running...

 remote   refid  st t when poll reach   delay   offset  jitter
==
 127.127.1.0 .LOCL.  10 l   21   64  3770.0000.000   0.001
 10.0.31.2   10.0.31.33 u  175 1024  3770.635   -5.771   2.042
*10.0.31.3   129.6.15.29  2 u  970 1024  3770.510  -11.340   0.449
 10.1.31.2   10.0.31.33 u  490 1024  3770.850   -9.114   4.881
+10.1.31.3   129.6.15.29  2 u  184 1024  3770.817   -4.085   5.355


synchronised to NTP server (10.0.31.3) at stratum 3
   time correct to within 68 ms
   polling server every 1024 s

Current time in UTC is : Mon Nov 16 03:16:14 UTC 2015
Current time in America/Chicago is : Sun Nov 15 21:16:14 CST 2015
admin:

admin:utils diagnose module validate_network

Log file: platform/log/diag1.log

Starting diagnostic test(s)
===
test - validate_network: Passed

Diagnostics Completed

admin:#

admin:utils dbreplication runtimestate

DB and Replication Services: ALL RUNNING

Cluster Replication State: Replication repair command started at: 
2014-06-20-23-22
 Replication repair command COMPLETED 541 tables processed out of 541
 Errors or Mismatches Were Found:

 Use 'file view activelog 
cm/trace/dbl/sdi/ReplicationRepair.2014_06_20_23_22_51.out' to see the details

DB Version: ccm8_6_2_2_2
Number of replicated tables: 541

Cluster Detailed View from PUB (5 Servers):

PINGREPLICATION REPL.   DBver&  
REPL.   REPLICATION SETUP
SERVER-NAME IP ADDRESS  (msec)  RPC?STATUS  QUEUE   TABLES  
LOOP?   (RTMT) & details
--- --  --- -   --- 
-   -
IPTCMS02  10.0.126.12 0.196   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMS01  10.0.126.11 0.151   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMP10.0.126.10 0.065   Yes Connected   0   match   Yes   
  (2) PUB Setup Completed
IPTCMS03  10.1.126.13 0.545   Yes Connected   0   match   Yes   
  (2) Setup Completed
IPTCMS04  10.1.126.14 0.527   Yes Connected   0   match   Yes   
  (2) Setup Completed

admin:

On Sun, Nov 15, 2015 at 9:14 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:
Also worth noting that if CCM NTP is synchronized to a Windows server (even if 
it shows Stratum 3 or better); that is a problem you'll need to correct as SNTP 
can play hell with UCOS and do some pretty weird stuff.

Thanks,

Ryan

On Nov 15, 2015, at 10:06 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

If the FROM CCM version was unrestricted, it would say "Unrestricted" after the 
version number on the "Active Master Version" line. If it does not say 
"Unrestricted", then it is the more common restricted version.

As to your original issue, I would start with all the usual suspects. Is the 
FROM CCM cluster healthy to start with; dns, ntp, replication ... etc?

>From CCM:

1.) #utils diagnose module validate_network
(Should see Passed)

2.) #utils ntp status
(Pub should be synchronized and Strata 3 or better)

3.) #utils dbreplication runtimestate
(Should see 2 - Setup Completed for all nodes)

If PCD is moving the apps to a new platform/chassis, make sure the target 
environment can reach all the same network assets as the from environment.

Thanks,

Ryan

On Nov 15, 2015, at 9:34 PM, Jonathan Charles 
<jonv...@gmail.com<mailto:jonv...@gmail.com>> wrote:

It just says the version number...

admin:show version active
Active Master Version: 8.6.2.2-2
Active Version Installed Software Options:
cmterm-7942_7962-sccp.9-3-1ES27-rel.cop
cmterm-devicepack8.6.2.24118-1.cop
ciscocm.refresh_upgrade_v1.1.cop
ciscocm.ucmap_platformconfig.cop
ciscocm.migrate-export-v1.12.cop
admin:


Jonahan

On Sun, Nov 15, 2015 at 7:26 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

Re: [cisco-voip] solution for multiple sip carriers?

[Ryan Huff]

Often times set providers Will allow you to mask caller ID however you choose. 
In cases where they do not, They should be able to remove the restriction If 
you ask. You may have to sign some forms in order for them to do that, If the 
provider services e911.

A few moons ago when I work for a regional service provider, We had a SBC that 
had a connection from level 3 And a connection from bandwidth.com.  We would 
prefer the level 3 connection for egress and failover to the bandwidth.com 
connection when the level 3 connection was down.

Basically we just sent all egress to the SBC and we let HSRP sort out Whether 
or not it was going out the primary connection or the failover connection.
Sent from my T-Mobile 4G LTE Device


 Original message 
From: "Norton, Mike"
Date:11/06/2015 4:26 PM (GMT-05:00)
To: "Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Have you asked either carrier if they can remove the outbound caller ID 
restriction? With my carrier (PRI, not SIP, but same issue) it was basically 
just a matter of asking them.

-mn

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Barnett, Nick
Sent: November-06-15 11:17 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] solution for multiple sip carriers?

I’m looking for some advice on how people handle situations with multiple sip 
carriers. We have a mix throughout our company of DNs ported from 2 different 
sip providers. If only the first provider could have ported everything, but 
that didn’t happen. I’m trying to NOT use a proxy as I don’t want to maintain a 
giant list of DNs. Also of note is that the DNs are not in ranges…

I have a single CUCM 10.0 cluster and 2 CUBEs (one at each data center).  But 
since I’m just labbing this up, I’m just dealing with a testbed cube and test 
cluster.

I have 3 ideas on how to handle this. First was to use a CSS on the line that 
added sig/steering digits on the front… say… 01 for carrier 1 and 2 for 
carrier 2. Then I could peel them off and send them out to the correct SBCs for 
each provider. I don’t like this, it makes it confusing for support staff to 
see how a call SHOULD exit.

My other idea may be a pipe dream… I’ve added a URI to the DN, so, let’s say 
1...@carrier1.example.com and 
1...@carrier2.example.com on the 2nd carrier 
DN. Then I changed the sip trunk to the CUBE to pass this info to CUBE. I see 
the correct stuff flowing through, and it works with my standard incoming 
called number and destination pattern dial peers. Now, I’m trying to figure out 
how to write a URI dial peer that will do magic things for me. I need to be 
able to route based on calling party URI… is that even possible? I haven’t been 
able to find an example like this, so I’m beginning to grow skeptical.

The 3rd idea is to use CUSP, which I don’t want to do…

Incoming calls is not an issue. The problem arises when making sure that an 
outbound call placed from a number from carrier1 goes out a sip trunk to 
carrier1. If it doesn’t go to the same carrier that owns the TN, either the 
caller id will be overwritten with the sip trunks BTN, or I have to apply a 
diversion header and all calls show up as long distance (and no longer roll up 
into billing codes).

Is there some other preferred method that people use?

Thanks in advance,
Nick
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Re: [cisco-voip] solution for multiple sip carriers?

[Ryan Huff]

I should probably further explain;  we would originate all calls through one 
carrier (the less expensive one) and terminate all calls from both carriers. We 
would use HSRP In cases where there were outages from the primary link.

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Ryan Huff
Date:11/06/2015 4:45 PM (GMT-05:00)
To: "Norton, Mike" ,"Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Often times set providers Will allow you to mask caller ID however you choose. 
In cases where they do not, They should be able to remove the restriction If 
you ask. You may have to sign some forms in order for them to do that, If the 
provider services e911.

A few moons ago when I work for a regional service provider, We had a SBC that 
had a connection from level 3 And a connection from bandwidth.com.  We would 
prefer the level 3 connection for egress and failover to the bandwidth.com 
connection when the level 3 connection was down.

Basically we just sent all egress to the SBC and we let HSRP sort out Whether 
or not it was going out the primary connection or the failover connection.
Sent from my T-Mobile 4G LTE Device


 Original message 
From: "Norton, Mike"
Date:11/06/2015 4:26 PM (GMT-05:00)
To: "Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Have you asked either carrier if they can remove the outbound caller ID 
restriction? With my carrier (PRI, not SIP, but same issue) it was basically 
just a matter of asking them.

-mn

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Barnett, Nick
Sent: November-06-15 11:17 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] solution for multiple sip carriers?

I’m looking for some advice on how people handle situations with multiple sip 
carriers. We have a mix throughout our company of DNs ported from 2 different 
sip providers. If only the first provider could have ported everything, but 
that didn’t happen. I’m trying to NOT use a proxy as I don’t want to maintain a 
giant list of DNs. Also of note is that the DNs are not in ranges…

I have a single CUCM 10.0 cluster and 2 CUBEs (one at each data center).  But 
since I’m just labbing this up, I’m just dealing with a testbed cube and test 
cluster.

I have 3 ideas on how to handle this. First was to use a CSS on the line that 
added sig/steering digits on the front… say… 01 for carrier 1 and 2 for 
carrier 2. Then I could peel them off and send them out to the correct SBCs for 
each provider. I don’t like this, it makes it confusing for support staff to 
see how a call SHOULD exit.

My other idea may be a pipe dream… I’ve added a URI to the DN, so, let’s say 
1...@carrier1.example.com<mailto:1...@carrier1.example.com> and 
1...@carrier2.example.com<mailto:1...@carrier2.example.com> on the 2nd carrier 
DN. Then I changed the sip trunk to the CUBE to pass this info to CUBE. I see 
the correct stuff flowing through, and it works with my standard incoming 
called number and destination pattern dial peers. Now, I’m trying to figure out 
how to write a URI dial peer that will do magic things for me. I need to be 
able to route based on calling party URI… is that even possible? I haven’t been 
able to find an example like this, so I’m beginning to grow skeptical.

The 3rd idea is to use CUSP, which I don’t want to do…

Incoming calls is not an issue. The problem arises when making sure that an 
outbound call placed from a number from carrier1 goes out a sip trunk to 
carrier1. If it doesn’t go to the same carrier that owns the TN, either the 
caller id will be overwritten with the sip trunks BTN, or I have to apply a 
diversion header and all calls show up as long distance (and no longer roll up 
into billing codes).

Is there some other preferred method that people use?

Thanks in advance,
Nick
___
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cisco-voip@puck.nether.net
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Re: [cisco-voip] solution for multiple sip carriers?

[Ryan Huff]

I should probably further explain;  we would originate all calls through one 
carrier (the less expensive one) and terminate all calls from both carriers. We 
would use HSRP In cases where there were outages from the primary link.

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Ryan Huff
Date:11/06/2015 4:45 PM (GMT-05:00)
To: "Norton, Mike" ,"Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Often times set providers Will allow you to mask caller ID however you choose. 
In cases where they do not, They should be able to remove the restriction If 
you ask. You may have to sign some forms in order for them to do that, If the 
provider services e911.

A few moons ago when I work for a regional service provider, We had a SBC that 
had a connection from level 3 And a connection from bandwidth.com.  We would 
prefer the level 3 connection for egress and failover to the bandwidth.com 
connection when the level 3 connection was down.

Basically we just sent all egress to the SBC and we let HSRP sort out Whether 
or not it was going out the primary connection or the failover connection.
Sent from my T-Mobile 4G LTE Device


 Original message 
From: "Norton, Mike"
Date:11/06/2015 4:26 PM (GMT-05:00)
To: "Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Have you asked either carrier if they can remove the outbound caller ID 
restriction? With my carrier (PRI, not SIP, but same issue) it was basically 
just a matter of asking them.

-mn

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Barnett, Nick
Sent: November-06-15 11:17 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] solution for multiple sip carriers?

I’m looking for some advice on how people handle situations with multiple sip 
carriers. We have a mix throughout our company of DNs ported from 2 different 
sip providers. If only the first provider could have ported everything, but 
that didn’t happen. I’m trying to NOT use a proxy as I don’t want to maintain a 
giant list of DNs. Also of note is that the DNs are not in ranges…

I have a single CUCM 10.0 cluster and 2 CUBEs (one at each data center).  But 
since I’m just labbing this up, I’m just dealing with a testbed cube and test 
cluster.

I have 3 ideas on how to handle this. First was to use a CSS on the line that 
added sig/steering digits on the front… say… 01 for carrier 1 and 2 for 
carrier 2. Then I could peel them off and send them out to the correct SBCs for 
each provider. I don’t like this, it makes it confusing for support staff to 
see how a call SHOULD exit.

My other idea may be a pipe dream… I’ve added a URI to the DN, so, let’s say 
1...@carrier1.example.com<mailto:1...@carrier1.example.com> and 
1...@carrier2.example.com<mailto:1...@carrier2.example.com> on the 2nd carrier 
DN. Then I changed the sip trunk to the CUBE to pass this info to CUBE. I see 
the correct stuff flowing through, and it works with my standard incoming 
called number and destination pattern dial peers. Now, I’m trying to figure out 
how to write a URI dial peer that will do magic things for me. I need to be 
able to route based on calling party URI… is that even possible? I haven’t been 
able to find an example like this, so I’m beginning to grow skeptical.

The 3rd idea is to use CUSP, which I don’t want to do…

Incoming calls is not an issue. The problem arises when making sure that an 
outbound call placed from a number from carrier1 goes out a sip trunk to 
carrier1. If it doesn’t go to the same carrier that owns the TN, either the 
caller id will be overwritten with the sip trunks BTN, or I have to apply a 
diversion header and all calls show up as long distance (and no longer roll up 
into billing codes).

Is there some other preferred method that people use?

Thanks in advance,
Nick
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Re: [cisco-voip] solution for multiple sip carriers?

[Ryan Huff]

I should stop using speech to text ... TERMINATE calls through one carrier and 
accept origination from both.

 Original message 
From: Ryan Huff
Date:11/06/2015 5:05 PM (GMT-05:00)
To: "Norton, Mike" ,"Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

I should probably further explain;  we would originate all calls through one 
carrier (the less expensive one) and terminate all calls from both carriers. We 
would use HSRP In cases where there were outages from the primary link.

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Ryan Huff
Date:11/06/2015 4:45 PM (GMT-05:00)
To: "Norton, Mike" ,"Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Often times set providers Will allow you to mask caller ID however you choose. 
In cases where they do not, They should be able to remove the restriction If 
you ask. You may have to sign some forms in order for them to do that, If the 
provider services e911.

A few moons ago when I work for a regional service provider, We had a SBC that 
had a connection from level 3 And a connection from bandwidth.com.  We would 
prefer the level 3 connection for egress and failover to the bandwidth.com 
connection when the level 3 connection was down.

Basically we just sent all egress to the SBC and we let HSRP sort out Whether 
or not it was going out the primary connection or the failover connection.
Sent from my T-Mobile 4G LTE Device


 Original message 
From: "Norton, Mike"
Date:11/06/2015 4:26 PM (GMT-05:00)
To: "Barnett, Nick" ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] solution for multiple sip carriers?

Have you asked either carrier if they can remove the outbound caller ID 
restriction? With my carrier (PRI, not SIP, but same issue) it was basically 
just a matter of asking them.

-mn

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Barnett, Nick
Sent: November-06-15 11:17 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] solution for multiple sip carriers?

I’m looking for some advice on how people handle situations with multiple sip 
carriers. We have a mix throughout our company of DNs ported from 2 different 
sip providers. If only the first provider could have ported everything, but 
that didn’t happen. I’m trying to NOT use a proxy as I don’t want to maintain a 
giant list of DNs. Also of note is that the DNs are not in ranges…

I have a single CUCM 10.0 cluster and 2 CUBEs (one at each data center).  But 
since I’m just labbing this up, I’m just dealing with a testbed cube and test 
cluster.

I have 3 ideas on how to handle this. First was to use a CSS on the line that 
added sig/steering digits on the front… say… 01 for carrier 1 and 2 for 
carrier 2. Then I could peel them off and send them out to the correct SBCs for 
each provider. I don’t like this, it makes it confusing for support staff to 
see how a call SHOULD exit.

My other idea may be a pipe dream… I’ve added a URI to the DN, so, let’s say 
1...@carrier1.example.com<mailto:1...@carrier1.example.com> and 
1...@carrier2.example.com<mailto:1...@carrier2.example.com> on the 2nd carrier 
DN. Then I changed the sip trunk to the CUBE to pass this info to CUBE. I see 
the correct stuff flowing through, and it works with my standard incoming 
called number and destination pattern dial peers. Now, I’m trying to figure out 
how to write a URI dial peer that will do magic things for me. I need to be 
able to route based on calling party URI… is that even possible? I haven’t been 
able to find an example like this, so I’m beginning to grow skeptical.

The 3rd idea is to use CUSP, which I don’t want to do…

Incoming calls is not an issue. The problem arises when making sure that an 
outbound call placed from a number from carrier1 goes out a sip trunk to 
carrier1. If it doesn’t go to the same carrier that owns the TN, either the 
caller id will be overwritten with the sip trunks BTN, or I have to apply a 
diversion header and all calls show up as long distance (and no longer roll up 
into billing codes).

Is there some other preferred method that people use?

Thanks in advance,
Nick
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Re: [cisco-voip] External calls from Jabber

[Ryan Huff]

Hi Aaron!

You are correct to think about CSS and partitions ... etc; that particular 
annunciation messeage typically comes about when the dialed digits do not match 
a pattern accessible to the calling device.

Does the CSF profile's device and/or line CSS have access to a route pattern 
that matches the dialed digits?

When the user dials externally, are they manually typing the digits or are they 
clicking on an external work/mobile number in the contact's profile?

If they are clicking on a contact's work/mobile number in the client, that 
number must also match an egressable pattern. You can supersede this behavior 
with application dial rules.

Hope this helps,

Ryan

Sent from my T-Mobile 4G LTE Device


 Original message 
From: Aaron Banks
Date:11/05/2015 12:10 PM (GMT-05:00)
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] External calls from Jabber


inShare
The jabber questions never end.  I am struggling with Jabber.  Users can call 
each other internally, check voicemail and IM each other, no problem.  When 
they go to make an external call to any number, they hear "your call cannot be 
completed as dialed" which would tell me maybe a calling search space issue.  
Not that.  I've changed the jabber client to start calls with audio instead of 
video. I've checked regions and there are only defaults.  Would the 
jabber-config.xml file create issues?  I would think it might bugger up 
directory services but not actually calling.
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Re: [cisco-voip] External calls from Jabber

[Ryan Huff]

Hello Aaron,


We would love to help you troubleshoot this issue; to do that we're at the 
point of needing to see 'what is going on under the hood'.


Please set up a test case for us by selecting a jabber desktop client impacted 
by this issue and clearing the local cache content of the client.

  *   Cache location for PC Clients: 
http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-windows/116433-probsol-jabber-00.html
  *   Cache location for Mac Clients: 
http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-mac/116682-technote-jabber-00.html

After you've clear the cache files; use the Jabber client to attempt an 
external call (replicate the same method the business unit is attempting to use 
for external calls that are failing). Immediately after the call fails 
(assuming it still does fail) and you get that annunciation message, I need you 
to end the call with the Jabber client and collect the "Calls" PRT (Problem 
Reporting Tool) logs from that Jabber client. You can access the PRT menu from 
the Help->Report A Problem menu of the Jabber Client. Follow the on-screen 
instructions and it will eventually allow you to download an archive file.

[cid:976983cc-3df8-4817-81c1-d341cd020363]


The next thing I'd like for you to do is to open the Real-Time Reporting Tool 
(RTMT) application and point it to the publisher node of your CCM cluster (if 
you do not have the RTMT application, you can download it from the 
Applications->Plugins section of CUCM). If needed, you can research RTMT at 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rtinst.html.
 Once you are logged into the tool, please replicate the following workflow and 
follow the on-screen prompts to download the traces (make sure you get the 
traces from ALL CUCM nodes for a time period covering the time of the example 
call). You can research more about getting trace files at 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rttlc.html.


[cid:4b623c9c-04cc-49ce-be22-d938309539ff]


Once you have all the files gathered, upload them to dropbox/box .. etc and 
send a linkcinto this list along with the called and calling party number of 
the example call and the device name of the CSF profile used by the Jabber 
Desktop client for the example call.


Hope this help,


-Ryan



From: cisco-voip  on behalf of Aaron Banks 

Sent: Thursday, November 5, 2015 1:51 PM
To: Anthony Holloway
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] External calls from Jabber


Per Charles' suggestion, I used the DNA tool to analyze the dialed call from 
jabber.  Nothing is being blocked.  Digits are not being dialed by clicking on 
the contact, the dial pad is being used in jabber or entered in the calling 
window.

Date: Thu, 5 Nov 2015 12:01:13 -0600
Subject: Re: [cisco-voip] External calls from Jabber
From: avholloway+cisco-v...@gmail.com
To: amichaelba...@hotmail.com
CC: cisco-voip@puck.nether.net

It's pretty straight forward, if you already have a grasp on CUCM core 
functionality.

Your reported error message sounds like the Annunciator telling you there were 
no matches in the dial plan for what you dialed.  You didn't mention what was 
dialed, so we cannot help you there.  If however, the error message was more 
like fast busy or ringback then call drop, I would say it was a failure tCacheo 
establish bi-directional media.  That's not that case though, so I would focus 
on your dialing habits and your matched patterns.

Also, Cisco Jabber does not use en bloc dialing, dispite the fact there is no 
dial tone.  Seems a bit backwards, but that's how it is.

On Thu, Nov 5, 2015 at 11:10 AM, Aaron Banks 
> wrote:

inShare
The jabber questions never end.  I am struggling with Jabber.  Users can call 
each other internally, check voicemail and IM each other, no problem.  When 
they go to make an external call to any number, they hear "your call cannot be 
completed as dialed" which would tell me maybe a calling search space issue.  
Not that.  I've changed the jabber client to start calls with audio instead of 
video. I've checked regions and there are only defaults.  Would the 
jabber-config.xml file create issues?  I would think it might bugger up 
directory services but not actually calling.

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Re: [cisco-voip] Jabber - auto detect server

[Ryan Huff]

Hi Aaron,


The auto-discovery feature in the Jabber client that you are speaking of uses 
DNS service discovery records (SRV). For the Jabber client, internally, it is 
looking for _cisco-uds._tcp.domain.tld (externally it would be looking for 
_collab-edge._tls.domain.tld). You can research the topic more at:


http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/Windows/9_7/CJAB_BK_C606D8A9_00_cisco-jabber-dns-configuration-guide/CJAB_BK_C606D8A9_00_cisco-jabber-dns-configuration-guide_chapter_010.html


If you already have the internal SRV record created per Cisco's Documentation, 
and it still isn't working correctly, the user's login suffix (@domain.tld) 
could potentially be provisioned by Cisco's cloud-based WebEx Meeting service, 
in which case the Jabber client will always prefer that as an authentication 
mechanism (unless you exclude the method in the jabber-config.xml file). The 
Jabber mobile clients work a little differently when it comes to preventing 
them to authenticate with Cisco's cloud-based WebEx Meeting service. My guess 
is though, if the domain were provisioned for the service, you would likely 
already know about it.


Beyond that, and assuming the SRV record is created correctly, the client 
itself may have other issues related to DNS lookups. You may try a simple 
client reset / clear the cache files, if you have not already. If you would 
like to collect the PRT logs and send them to this list I'm sure someone can 
look through them for you.


Hope this helps,


Ryan




From: cisco-voip  on behalf of Aaron Banks 

Sent: Wednesday, November 4, 2015 10:16 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber - auto detect server

I've had some trouble getting jabber to auto detect the IM server.  Every 
time a new user signs in, they have to put in the IP of the server.  Anyone 
have tricks to offer to avoid using the IP or FQDN?
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[cisco-voip] UC & Linux server for DRS SFTP DNS and NTP

[Ryan Huff]

Hey all,


Some time last week, I think, someone had posed a query about options for DRS 
mechanisms (SFTP ... etc); among many of the great suggestions was, as there 
always is, a suggestion to use Linux (I myself recommended).


I did some thinking and thought, for some, suggesting a Linux server may be 
akin to asking that soup be eaten with a fork. We're all busy professionals; 
taking the time away from work and family to learn how to use an unfamiliar 
operating system just for UC backups? I realise that might be an obvious 
proposition.


With regards to Cisco UC specifically, I think "Linux" is not just knowing that 
UCOS is built on it, I think it should be an essential tool in your UC utility 
belt! Linux can solve many issues and get you out of binds (no pun intended). 
In my opinion, it is a much more reliable and scalable solution than other 
options (and it can be more useful to your UC environment than just a "DRS 
Box").


So, for those in mind, and for my LOVE of the penguin; over the next week'ish 
(we'll see how work goes) I am doing a multi-part blog series covering the 
niche topic of Cisco UC and Linux utility servers and specific use cases. 
Specifically, I will cover using SFTP, DNS, NTP & Web Servers in Linux (all 
useful goodies for Cisco UC).


I am not, nor will not sell anything nor does the blog have ads or stuff like 
that. This is just a sincere and genuine effort to assist my fellow brothers 
and sisters that may not be ultra familiar with Linux but would like to know 
more and how Linux utility servers could be used with Cisco UC.


I will not be notifying the list each time I have a new entry in the series, so 
if you would like to follow along or do a fly-by at some point, please 
visit/bookmark http://ryanthomashuff.com/category/linux-aficionado-series/ or 
use the RSS link http://ryanthomashuff.com/feed/


Right now, the first entry is up which is a primer to the series.


Thanks,


Ryan

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Re: [cisco-voip] Baud rate on an analog port

[Ryan Huff]

Generally, the transmission rates will be controlled by the actual device 
transmitting the signal. My experience with credit card machines is that they 
have an option buried within their menu system for changing it or you have to 
call the machine's vendor and they can walk you through it.


What type of trouble are you experiencing with the machine?




From: cisco-voip  on behalf of 
norm.nichol...@kitchener.ca 
Sent: Tuesday, November 3, 2015 8:46 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Baud rate on an analog port




Is there a way to increase or decrease the baud rate on an analog port. I have 
some debit machines that are acting flaky and want to see if changing the baud 
rate helps.







Thanks











Norm Nicholson

Telecom Analyst

City of Kitchener

(519) 741-2200 x 7000




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Re: [cisco-voip] ESXi RAM Requirements

[Ryan Huff]

The tested reference solutions, I believe,  take into account only the specific 
VM memory overhead requirements for the number and size of UC VMs that are 
supported by the TRC; plus the overhead requirements of ESXi itself and, I 
believe, a certian amount of "buffer".

The additional features that you can access with enterprise licensing (vMotion 
 etc) have addition requirements that would scale beyond the minimum 4GB.

-Ryan


 Original message 
From: Anthony Holloway
Date:10/30/2015 9:58 PM (GMT-05:00)
To: Cisco VoIP Group
Subject: [cisco-voip] ESXi RAM Requirements

Hey All,

I'm trying to understand the concept of ESXi's RAM requirements, and with 5.5 
VMWare says the minimum is 4GB, but they do recommend 8GB for "full features."

When I look at this one guide as an example:
http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware#BE6000H.C2.A0Servers_and_Small_Plus_UConUCS_Tested.C2.A0Reference_Configurations

It shows that 48GB of RAM is installed inside, but only 44GB is available to 
VMs.  That makes me think Cisco is only accounting for a 4GB minimum, and not 
the 8GB recommended.

Is that what you read that as, as well?  And how does one allocate 8GB to ESXi? 
 Do you just not use it for your VMs and leave it on the table, a la CUC core 
reservation?

A little late Friday evening thoughts right before a holiday weekend.  That's 
how you know I'm in this job deep.  Thanks and have a great weekend.

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Re: [cisco-voip] Windows SFTP / FTP server recommendations?

[Ryan Huff]

IMO, FreeFTPd is one of those things that sounds great ... until you try it. My 
advice, take 20 minutes and spin up a Linux distribution with OpenSSH.

Thanks,

Ryan

Sent from my iPad

> On Oct 29, 2015, at 10:46 AM, Scott Voll  wrote:
> 
> What are others using for there UC backup server Application for SFTP and FTP?
> 
> I've started a pilot with FreeFTPd but it keeps crashing and then I have to 
> restart it and run my backups again.
> 
> What have others had good success using?
> 
> Thanks
> 
> Scott
> 
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Re: [cisco-voip] Windows SFTP / FTP server recommendations?

[Ryan Huff]

It is worth noting that SFTP and FTPS are completely different protocols.

FTPS is FTP with a control channel and opens new connections for the data 
transfer. As it uses SSL, it requires a certificate.

SFTP is an extension of SSH, so it usually uses only the SSH port for both data 
and control.

SSH server installations generally have SFTP support but FTPS typically needs a 
supported FTP server.

Sent from my iPad

On Oct 29, 2015, at 9:35 PM, Mike King 
> wrote:

http://secure.nerdster.com.au/knowledgebase/10075/How-to-enable-SFTP-with-Filezilla-Server.html

On Thu, Oct 29, 2015 at 9:30 PM, Charles Goldsmith 
> wrote:
same, i know it's an ftp server, but only the client can do sftp that i know 
about

On Thu, Oct 29, 2015 at 7:54 PM, Jason Aarons (AM) 
> wrote:
Love to know how you setup SFTP in Filezilla Server

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Mike King
Sent: Thursday, October 29, 2015 8:36 PM
To: Scott Voll >
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Windows SFTP / FTP server recommendations?


Filezilla is the FTP/ SFTP server I've been rolling with for awhile.

Mike

On Thu, Oct 29, 2015 at 10:45 AM, Scott Voll 
> wrote:
What are others using for there UC backup server Application for SFTP and FTP?

I've started a pilot with FreeFTPd but it keeps crashing and then I have to 
restart it and run my backups again.

What have others had good success using?

Thanks

Scott


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itevomcid

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Re: [cisco-voip] 7821 phones issue

[Ryan Huff]

Could you provide the "voice service voip" config section of HQ and a branch 
site?

Sent from my iPad

On Oct 28, 2015, at 8:08 AM, Ahmed Abd EL-Rahman 
> wrote:

I know that there is a lack of a response to the SIP invite message which is 
being sent from HQ 7821 phone to the branch 7821 phone during the call setup 
which may point to a network connectivity (or blocking) in the middle, but can 
anyone give me a reasonable answer for why the same messages are flowing 
normally when the 8831 HQ phone initiate the call to the 7821 branch phone ?? 
if something blocked it will be blocked for all.



Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - KSA

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ahmed 
Abd EL-Rahman
Sent: Wednesday, October 28, 2015 3:02 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 7821 phones issue

Hi Gents,


I have a setup of CME 10.5 on 2911 ISR in the HQ, then I have a DMVPN to the 
branch site.

I have the following phone types:

- 7945 skinny based phone.

- 8831 SIP conference station.

-7821 SIP phones.

- 7861 SIP phone.

the setup is working fine in the HQ (on the same LAN) but when i took some of 
the 7821 phones and placed them in the branch site the following happens:

- They are able to register to the CME in the HQ.

- Branch phones can call all HQ phones and can also call each others’ in the 
branch normally.

- The issue is that HQ 7821 & 7861 phones cannot call 7821 branch phones (no 
ring back is heard then the call disconnects after around 1 minute), despite 
the fact that HQ 7945 and 8831 (which is also SIP based) can call Branch 7821 
phones normally with no issues ???



I have upgraded the router IOS to the latest safe harbor version 15.4(3)M4 and 
upgraded the 7821 firmware to the latest version 10.3.1 with no success.

I have enabled the debug voice ccapi inout and collected the output from 2 
calls, 1 is from HQ 8831 SIP phone to a branch 7821 phone which was successful 
and the other from HQ 7821 phone to a branch 7821 phone which was unsuccessful 
and i compared the two outputs line by line and discovered that in the 
successful call the alerting singal (ring back) is received normally, but in 
the unsuccessful one it's not received and a disconnect signal is received as 
shown below:

- successful call:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678//CCAPI/cc_api_get_xcode_stream:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678//CCAPI/cc_api_get_xcode_stream:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678/1F64117E85C0/CCAPI/cc_api_call_proceeding:
   Interface=0x3E81FC6C, Progress Indication=NULL(0)
*Oct 25 10:43:55.466: //678/1F64117E85C0/CCAPI/cc_api_call_alert:
   Interface=0x3E81FC6C, Progress Indication=NULL(0), Signal Indication=SIGNAL 
RINGBACK(1)
*Oct 25 10:43:55.466: //678/1F64117E85C0/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=677



- unsuccessful call:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357//CCAPI/cc_api_get_xcode_stream:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357//CCAPI/cc_api_get_xcode_stream:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357/C9AD33DD82E1/CCAPI/cc_api_call_proceeding:
   Interface=0x3E81FC6C, Progress Indication=NULL(0)

*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnected:
   Cause Value=102, Interface=0x3E81FC6C, Call Id=357
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=102, Retry Count=0)
*Oct 25 08:19:24.863: //356/C9AD33DD82E1/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=357
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect 
Cause=102)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Call Entry(Responsed=TRUE, Cause Value=102)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3E81FC6C, Tag=0x0, Call Id=357,
   Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event 

Re: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery

[Ryan Huff]

UCOS does not support Windows time services. Cisco IOS based devices or Linux 
servers are recommended.

Sent from my iPad

On Oct 28, 2015, at 1:33 PM, Thomas LeMay 
<thomasle...@comcast.net<mailto:thomasle...@comcast.net>> wrote:

Hi,

Questions: What is the best business practice for the type of NTP server to use 
as the source: windows or some other operating system?

Thank you,

Tom

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Aaron 
Banks
Sent: Wednesday, October 28, 2015 12:39 AM
To: Ryan Huff; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery

Thank you for all of that.  You know what it was - NTP.  I shut down the HA.  
NTP was doing weird things on the primary node and I asked the customer if the 
NTP server address he gave me was a windows server.  Bingo.  I changed the NTP 
source, rebooted the primary, called voicemail and then powered on the HA.

Lesson learned.

From: ryanh...@outlook.com<mailto:ryanh...@outlook.com>
To: amichaelba...@hotmail.com<mailto:amichaelba...@hotmail.com>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: RE: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery
Date: Tue, 27 Oct 2015 17:25:47 +
1.) Shut down the HA node.

2.) Reboot the primary node

3.) Once the primary node is up, place a call into voicemail

4.) Power the HA node back on

5.) Once HA is up, verify HA status.



Sent from my T-Mobile 4G LTE Device


 Original message 
From: Aaron Banks
Date:10/27/2015 12:35 PM (GMT-05:00)
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery


Has anyone seen/resolved a split brain recovery in Unity Connection 10.5.2?  
The primary and secondary keep swapping back and forth every few minutes.  I 
can ping and trace to each server.  I restarted the primary but that did not 
resolve the issue.  In the RTMT system logs, the secondary sends an NTP query 
to the primary the response is the primary is inaccessible or down.  I'm 
stumped.
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Re: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery

[Ryan Huff]

1.) Shut down the HA node.

2.) Reboot the primary node

3.) Once the primary node is up, place a call into voicemail

4.) Power the HA node back on

5.) Once HA is up, verify HA status.



Sent from my T-Mobile 4G LTE Device


 Original message 
From: Aaron Banks
Date:10/27/2015 12:35 PM (GMT-05:00)
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery



Has anyone seen/resolved a split brain recovery in Unity Connection 10.5.2?  
The primary and secondary keep swapping back and forth every few minutes.  I 
can ping and trace to each server.  I restarted the primary but that did not 
resolve the issue.  In the RTMT system logs, the secondary sends an NTP query 
to the primary the response is the primary is inaccessible or down.  I'm 
stumped.
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Re: [cisco-voip] CUBE - Dial-peer multiple destination-pattern matching

[Ryan Huff]

A couple of items to keep in mind;

1.) Can only be used on VOIP dial peers

2.) When you have multiple pattern matches, the match with the longest prefix 
is considered the matching criteria which does not necessarily have to be the 
most specific match.

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/dialpeer/configuration/15-mt/vd-15-mt-book/vd-mdp-dialpeer.pdf

Unless there is an underlying reason for MPS I would suggest a SME or just 
stand up an IC/SIP trunk between the clusters and kick on ILS. The 
justification for SME would be scalability and number of connected calls.

Where MPS has served me well is in a deployment with an insane amount of e164 
numbers. It allowed me to exceed the "rule of 15" for voice translations on the 
router and also do some aggressive summarization without having to make CCM's 
numplan stack look like a toddler's playroom.

Sent from my iPad

On Oct 26, 2015, at 6:39 AM, Ed Leatherman 
> wrote:

I don't have any experience with the file based patterns. If they have room for 
a few more VM's, putting SME in the middle and hang CUBE off of that might be 
another way to do it cleanly. Then ils could take care of those patterns.

On Mon, Oct 26, 2015 at 5:42 AM, Boon 
> wrote:
I have a client who is planning on splitting their single CUCM cluster with 
CUBE and PSTN SIP into two separate clusters.

The challenge is that they want to share the CUBE solution and DID range 
between both clusters.

I can see an opportunity here to use the IOS dial-peer feature 'Multiple 
Destination Pattern' matching using a file hosted in the router flash.

Although the configuration looks pretty straight forward I wanted to find out 
if any of you guys had deployed this feature and whether there are any gotchas 
to be aware of? I'm aware of the minimum IOS version requirement. I'm wondering 
whether file maintenance can become an issue.

Also, has anyone used this with a CUBE HA solution?

Any help appreciated. Thanks

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Re: [cisco-voip] 3rd Party backup tools

[Ryan Huff]

While it is more than NOT supported; I have had success with cloning ucos vm's 
(while powered off). Then powering on the clones with the parents turned off. 
Again, clearly not supported but I have not noticed any behavior differences.

For kick's and curiosity's sake I have vmotioned a lab cluster (with sip pstn) 
before while powered on ... caused cpu spikes and the trunks had to be reset.

However, in practice I'd not wonder outside the bounds of the DocWiki for 
backup purposes.


Sent from my T-Mobile 4G LTE Device


 Original message 
From: Anthony Holloway
Date:10/26/2015 12:20 PM (GMT-05:00)
To: "Heim, Dennis"
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 3rd Party backup tools

As a Cisco partner, I will be sticking to the book on this one (DRS).

http://docwiki.cisco.com/wiki/Ongoing_Virtualization_Operations_and_Maintenance#Backup.2C_Restore.2C_and_Server_Recovery

However, I am very curious to know what else works, and how well it works.

On Sun, Oct 25, 2015 at 11:12 PM, Heim, Dennis 
> wrote:
Whats the current thought process around 3rd party backup tools to quickly 
restore a CUCM? I’ve played with Veeam in the past. If the VM is shutdown, 
would that be kosher?

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[cid:image001.png@01D10DD2.7FC81F90]
[cid:image002.png@01D10DD2.7FC81F90][cid:image003.png@01D10DD2.7FC81F90][cid:image004.png@01D10DD2.7FC81F90]
“There is a fine line between Wrong and Visionary. Unfortunately, you have to 
be a visionary to see it." – Sheldon Cooper
“The greatest danger for most of us is not that our aim is too high and we miss 
it, but that it is too low and we reach it.” -- Michelangelo Buonarroti
“We should tansform the way we work” -- RowanTrollope

Click here to join me in my Collaboration Meeting 
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Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

I have two approaches;

If I am remote I'll spinup a linux vm for dns, dhcp and sftp service on a 
seperate port group  this all assuming everything is on one network 
segment. If I am crossing network boundaries i'll throw in a cloud services 
router or something.

If I am onsite, I have an 800 series router I carry with me and use it as a 
'router on a stick' and trunk it to the UCS server so I can replicate all the 
production networks, dhcp and dns (using different DOT1Q tags). Important to 
make sure you prune the trunks correctly to avoid conflicts :).

When I am ready to cut over, its just a matter of changing port groups around 
on the vNICs.

Sounds, simple but it never fails that a VM or two will need restarted or at 
least some services restarted after the cut.

Thanks,

Ryan


Sent from my T-Mobile 4G LTE Device

 Original message 
From: Lelio Fulgenzi <le...@uoguelph.ca> 
Date:10/20/2015  10:40 AM  (GMT-05:00) 
To: Cisco VoIP Group <cisco-voip@puck.nether.net> 
Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade? 


My major concern is IP address conflicts. Right now, with my offline network, 
it's completely isolated, with the UCS solution, unless I buy a UCS server(s) 
that is sized accordingly to hold the VMs sized for my production environment 
(which is unlikely), I'll have to consider feeding the UCS servers with the 
offline network which is a duplicate of the production network.

While technically possible, it would mean that the entire team be aware of this 
special configuration and not mess with anything that could bridge the 
configurations and cause IP address conflicts. I'm envisioning a separate 
network cable plugged into our offline switch/routers that would connect to the 
UCS server(s) in a separate VLAN. Unfortunately, I'm not up to speed with the 
intricacies of the UCS systems and VM, etc. and whether or not that would cause 
issues.

As with other organizations, we're under pressure to keep systems up and 
running 24 hours a day. We're improving our service availability design to 
hopefully help us going forward. For example, building an almost exact 
duplicate of our auto-attendant and call processing on Unity Express, so that 
we can do work on Connection while peoples business can continue at 3 in the 
morning while we do work. 

We may revisit the offline upgrade in lieu of an inplace upgrade, but I'm not 
sure we're there yet.

Lelio


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

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

From: "Ryan Huff" <ryanh...@outlook.com>
To: "Lelio Fulgenzi" <le...@uoguelph.ca>
Cc: "Kevin Przybylowski" <kev...@advancedtsg.com>, "Anthony Holloway" 
<avholloway+cisco-v...@gmail.com>, "Cisco VoIP Group" 
<cisco-voip@puck.nether.net>
Sent: Sunday, October 18, 2015 7:04:42 PM
Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

Lelio, what challenges are you facing in your next upgrade on UCS?

I don't think staging is so much 'the old way of doing it' as much as it is 
depending on the engagement and timeline, in my opinion. 

If the target environment is only sized for the production VMs (and your coming 
from MCS)  it might be difficult to do a bridge in the target environment. 
In that case, I would advocate pulling the DRS and upgrading offnet, unless the 
customer can spin-up an sftp server that you can use to shuffle DRS on. At that 
point though, I'd say it is going to take just as long, one way or the other.

Virtual to virtual with plenty of room in the datastores can certainly, and 
easily be done onnet. In place upgrades are another great case for onnet 
upgrades without staging.

As Anthony mentioned earlier, PCD is only valuable (IMO) in a rather limited 
set of circumstances and has enough nuances that I don't bother with it. PCD 
has a ways to go before I'd consider using it in practice; it's a good idea 
-just too early for me.

Some organizations have change-controls that mandate major upgrades be ran in 
tandem/staged, then switched once signed off on. In those cases, I'll usually 
advocate upgrading the device loads on the current version before the switch, 
so at least the phones move over quick.

Thanks,

Ryan

Sent from my iPad

On Oct 18, 2015, at 6:36 PM, Lelio Fulgenzi <le...@uoguelph.ca> wrote:

We've built an offline network where we have staged the last couple of 
upgrades. It's worked out well. We basically test everything we can to ensure 
operability. The day of the cutover there's an overall downtime of about 90 
minutes but some things come up sooner. 

I'm hoping to come up with a similar approach to the next one. But it would be 
using ucs so I'm not sure how to ma

Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

Lelio, what challenges are you facing in your next upgrade on UCS?

I don't think staging is so much 'the old way of doing it' as much as it is 
depending on the engagement and timeline, in my opinion. 

If the target environment is only sized for the production VMs (and your coming 
from MCS)  it might be difficult to do a bridge in the target environment. 
In that case, I would advocate pulling the DRS and upgrading offnet, unless the 
customer can spin-up an sftp server that you can use to shuffle DRS on. At that 
point though, I'd say it is going to take just as long, one way or the other.

Virtual to virtual with plenty of room in the datastores can certainly, and 
easily be done onnet. In place upgrades are another great case for onnet 
upgrades without staging.

As Anthony mentioned earlier, PCD is only valuable (IMO) in a rather limited 
set of circumstances and has enough nuances that I don't bother with it. PCD 
has a ways to go before I'd consider using it in practice; it's a good idea 
-just too early for me.

Some organizations have change-controls that mandate major upgrades be ran in 
tandem/staged, then switched once signed off on. In those cases, I'll usually 
advocate upgrading the device loads on the current version before the switch, 
so at least the phones move over quick.

Thanks,

Ryan

Sent from my iPad

> On Oct 18, 2015, at 6:36 PM, Lelio Fulgenzi <le...@uoguelph.ca> wrote:
> 
> We've built an offline network where we have staged the last couple of 
> upgrades. It's worked out well. We basically test everything we can to ensure 
> operability. The day of the cutover there's an overall downtime of about 90 
> minutes but some things come up sooner. 
> 
> I'm hoping to come up with a similar approach to the next one. But it would 
> be using ucs so I'm not sure how to make that work just yet. 
> 
> Sent from my iPhone
> 
>> On Oct 16, 2015, at 3:52 PM, Kevin Przybylowski <kev...@advancedtsg.com> 
>> wrote:
>> 
>> It is very time consuming to stage in the lab… Installs, DRS’s, Upgrades, 
>> etc…  I’ve only done them in the past if there was a large gap in versions.  
>> It looks like PCD PCD is getting better so it looks like a valid option 
>> nowadays for bare metal to esx migration/upgrades.
>>  
>> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
>> Anthony Holloway
>> Sent: Friday, October 16, 2015 3:42 PM
>> To: Ryan Huff <ryanh...@outlook.com>
>> Cc: Cisco VoIP Group <cisco-voip@puck.nether.net>
>> Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?
>>  
>> You sound more organized than I am.  I would like to see what you have, 
>> sure.  Thanks for the offer.
>>  
>> I've never staged an upgrade in my lab, though I have heard of plenty of 
>> people doing this.  Is it really something to consider or is that a thing of 
>> the past?  Like pulling a drive from the array?  Not too mention, I rarely 
>> have time to perform two upgrades on a project like this.  I barely get 
>> enough time to upgrade the system once.
>>  
>> On Fri, Oct 16, 2015 at 1:56 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> I use an excel spread sheet with a hyperlink to the base doc in one sheet 
>> with notes and details gathered in the sheet.Then I create additional 
>> worksheets of subordinate documentation and notes and then make references 
>> from the base sheet to the subordinate sheets. I also have a sheet for 
>> customer discovery (current dns, ip, device loads  etc). It ends up 
>> looking a lot like a Gantt chart.
>>  
>> If you'd like, I can sanitize and send one to you, to compare notes and see 
>> if there is anything of use to you.
>>  
>> Also, If time permits, and it's feasible,  I like to stage a mock upgrade in 
>> my lab with customer data (drs ... etc) and do a dry run.
>> 
>> 
>>  Original message 
>> From: Anthony Holloway 
>> Date:10/16/2015 2:38 PM (GMT-05:00) 
>> To: Cisco VoIP Group 
>> Subject: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?
>> 
>> Does anyone else do this?  Gather all of the documentation ahead of time, 
>> because inevitably you're going to revisit a document more than once?  There 
>> are a lot of documents to gather!  Anything I could be doing better?  Tips?  
>> Tricks?
>>  
>> I create a spreadsheet of all of the pertinent documents I need to review or 
>> reference, like in this screenshot.  There's over 90 documents in this list. 
>>  Granted, I don't read them all front to back, but some I do, and  for 
>> others I need to reference information within them nonetheless. 

Re: [cisco-voip] Video Design Guide

[Ryan Huff]

Yes, I live on my iPad ... I have issues, I know.

I think the architecture guide you referenced is very good, especially the 
"Basic Video Concepts" section. 

For a learner just starting with video I might suggest something even more 
digestible and agnostic, just to get the feet wet with concepts like 
compression and encoding. It is a bit dated but still very applicable and 
teaches the "why" not just the "how". 
http://www.amazon.com/How-Video-Works-Analog-Definition/dp/0240809335

The CLN also has a decent "basics" course: 
https://learningnetwork.cisco.com/docs/DOC-6492

IMO, for someone well versed in all things Cisco, I think the architecture 
guide you have is spot on.

Sent from my iPad

> On Oct 18, 2015, at 12:15 PM, Anthony Holloway 
>  wrote:
> 
> Since there was only positive feedback from the group on the topic of 
> discussing video here, I'll just go ahead and ask the question I originally 
> wanted to ask
> 
> Is there a better document that this one, for learning the fundamentals of 
> video design?
> 
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/uc_system/design/guides/videodg/vidguide.html
> 
> It's a bit dated, but I like the material it covers.  I find the PAs and CVDs 
> to be too high level and too configuration focused, respectively.
> 
> I think it's important to understand things like frame rate, frame size, the 
> different compression methods, bitrates, etc.  Not too mention how Cisco's 
> implementation will scale the video up or down dynamically based on network 
> conditions.
> 
> Thanks for your input.
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Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

Here is the Excel spreadsheet that I use during UC upgrades. Don't beat me up 
too bad ;)  it is a collection of 'O wait, I need that too's'; it could 
probably use some consolidation / tweaking.
 
This is the version 1. In the 'version 2' I am making I am adding a testing 
plan section and I have also been experimenting with some AXL calls that will 
auto populate some of the 'discovery' stuff from the pre-upgraded nodes.
 
If anyone finds it to be useful in whole or part; have at it.
 
Thanks,
 
Ryan
Date: Fri, 16 Oct 2015 14:52:11 -0500
From: avholloway+cisco-v...@gmail.com
To: le...@uoguelph.ca
Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?
CC: cisco-voip@puck.nether.net

When you say Prime, I will assume you mean Prime Collaboration Deployment (PCD).
I have used PCD a few times now, but it's far from being the savior one might 
think it is.
First, PCD really only shines when migrating to v10 on new hardware.  Or, the 
same hardware, but you have twice the space.  You can jump straight from CUCM 
6.1(4) to 10.5(2) without COP files, or intermediate versions.  That's because 
PCD is actually installing v10 fresh, and just moving the data for you.  At 
least it tries to.  There are a few things it doesn't move yet.  E.g., DHCP 
Server TFTP Option 150.  Yes, I saw someone using that in CUCM!
Second, if your doing any other kind of upgrade in PCD, you're not really 
saving yourself from having to read all the documentation.  As your still bound 
to all the same restrictions and COP files, and whatever else.  You can look at 
PCD in this scenario as an intern who you've given instructions to and he/she 
just executes them while you go play GTA V on your Xbox you won in an 
Engineering Deathmatch.  The intern really isn't doing anything special for 
you, other than allowing you to look away while the upgrades happen.  And even 
then, I've seen them fail more times than they have succeeded.  YMMV.
Lastly, on the topic of PCD migrations, which are it's bread and butter, it 
only does this for CUCM and IM  Not CUC, nor CER, not UCCX, or anything 
else.  So, if you go migrate with PCD, then your stuck with COBRAS for CUC, BAT 
for CER, and who knows what else for the rest.  I'll leave that as an exercise 
to the reader.
At the end of the day, their maybe some environments where you can just pull 
the trigger and upgrade the system without reading any documentation, and just 
gamble, but for a professional of their craft, that's just not acceptable.
On Fri, Oct 16, 2015 at 2:02 PM, Lelio Fulgenzi  wrote:
I've never made a spreadsheet like yours, but I've done something similar. 
Typically with compatibility checking. You do have to visit a number of 
documents and/or links. It's quite frustrating to say the least.
The process of upgrading is a difficult one to say the least, especially with 
things like you mention, where gotchas are hidden deep in documents that you 
may not read front to back.
My biggest issue is when you skip versions, it's not really clear which 
documents to read with respect to changes. For example, when I upgraded from 
7.1 to 9.1, I found myself printing (k!) a number of documents which had 
duplicate information, but I wasn't sure on where to look.
And then there's the issue that each application will have different rules, so 
CUCM might say only print the latest minor version notes, any SU or a/b/c 
release will have everything you need. Where Connection or Unity Express might 
do something different.
It's not fun to say the least.
But isn't Prime supposed to make it easy to upgrade now?

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

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

From: "Anthony Holloway" 
To: "Cisco VoIP Group" 
Sent: Friday, October 16, 2015 2:38:40 PM
Subject: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

Does anyone else do this?  Gather all of the documentation ahead of time, 
because inevitably you're going to revisit a document more than once?  There 
are a lot of documents to gather!  Anything I could be doing better?  Tips?  
Tricks?
I create a spreadsheet of all of the pertinent documents I need to review or 
reference, like in this screenshot.  There's over 90 documents in this list.  
Granted, I don't read them all front to back, but some I do, and  for others I 
need to reference information within them nonetheless.  You never know when you 
might find a small font hidden note in there.
E.g., From the 8945 Release Notes
"Release 9.4(2)SR1 can only be upgraded from 9.3(4) and later. Releases prior 
to 9.3(4) have to be upgraded to 9.3(4) first."
Source: 8945 9.4(2)SR1 Release Notes
I actually missed this one recently, and unlike 7900 series phones, they 

Re: [cisco-voip] 79xx Ring in name display

[Ryan Huff]

I should further say that I am assuming you do have these set, as you stated an 
issue with one phone and not all phones (I tend to start at the beginning and 
work forward).

Another question would be in what call-flow scenarios is the user not seeing 
the display name (clid)? 

Is the issue in all call-flow scenarios or just one of these scenarios; 
- pstn->phone
- ip phone to ip phone
- when the call is forwarded from another ip phone or call is answer via line 
group

Thanks,

Ryan

> On Oct 17, 2015, at 8:40 AM, Ryan Huff <ryanh...@outlook.com> wrote:
> 
> Hello Martin,
> 
> On your ingress gateway/trunk, do you have "Display IE Delivery" or 
> "Connected Name Presentation" checked/allowed (respectively)?
> 
> Thanks,
> 
> Ryan
> 
> 
> From: m...@bilobit.com
> To: cisco-voip@puck.nether.net
> Date: Sat, 17 Oct 2015 10:25:26 +
> Subject: [cisco-voip] 79xx Ring in name display
> 
> Hi all,
>  
> a customer asks us for a solution for the following:
>  
> When a call comes in on a 7965 they want to have not only the calling party 
> number, but the name of the caller as well. We already have a MetaDirectory 
> there which collects contact information from several databases and presents 
> them via LDAP to 3rd party solutions.
>  
> Does anyone have an idea?
>  
> Thanks and nice weekend, Martin
> 
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Re: [cisco-voip] 79xx Ring in name display

[Ryan Huff]

Hello Martin,

On your ingress gateway/trunk, do you have "Display IE Delivery" or "Connected 
Name Presentation" checked/allowed (respectively)?

Thanks,

Ryan


From: m...@bilobit.com
To: cisco-voip@puck.nether.net
Date: Sat, 17 Oct 2015 10:25:26 +
Subject: [cisco-voip] 79xx Ring in name display

Hi all,
 
a customer asks us for a solution for the following:
 
When a call comes in on a 7965 they want to have not only the calling party 
number, but the name of the caller as well. We already have a MetaDirectory 
there which collects contact information from several databases and presents 
them via LDAP to 3rd party solutions.
 
Does anyone have an idea?
 
Thanks and nice weekend, Martin

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Re: [cisco-voip] Video Architecture

[Ryan Huff]

In my experience, I find the largest resistance to video from the customer's 
perspective is that it is still a "board room" or "conference room" feature. I 
think the industry is still a while off from video being adopted at the desk, 
or rather being used frequently at the desk. I have had engagements where it 
was heavily adopted, but more often am asked to disable it on the endpoints.

To the point of your thoughts; with Telepresence endpoints now being in the 
fold and the new changes to the Expressway series licensing it seems logical to 
me that Video is and will continue to be an increasingly present member of the 
UC family. To that end, I think it should be part of this list.

This list has a serious wealth of talent, more than I know is advertised. It 
has and continues to help me significantly. As Cisco video deployments become 
more standard, I think there will be a big need for the sort of ad-hoc support 
this list can offer for UC Video. 

As with Cisco voice, I think one of the many key contributors to widespread 
adoption is having forums like this, that are full of people who love what they 
do and like helping people.

I say bring on the Video!

Sent from my iPad

> On Oct 17, 2015, at 10:26 AM, Anthony Holloway 
>  wrote:
> 
> Does this list, being labeled "VoIP" and not "Voice", include Video in the 
> "V", or is the "V" for Voice only?  A bit of tongue-in-cheek there, but I 
> think you know what I mean.
> 
> With every day that passes, I feel like, as a Voice Engineer, I am being 
> asked to know and implement Video more and more.
> 
> From CUCM taking on more of the video call control (just about 100% now, 
> no?), to video being a part of the CCIE Collab (to some degree, don't flame 
> me for mentioning it), the line between voice and video engineering roles is 
> blurring.
> 
> I know there has been quite a bit of Expressway talk on here this year, but I 
> don't think I've explicitly read a thread which was 100% focused on video 
> technology.  What's the direction you see these two technologies heading, and 
> should/could this list serve as both a Voice and Video discussion group?
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Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

I use an excel spread sheet with a hyperlink to the base doc in one sheet with 
notes and details gathered in the sheet.Then I create additional worksheets of 
subordinate documentation and notes and then make references from the base 
sheet to the subordinate sheets. I also have a sheet for customer discovery 
(current dns, ip, device loads  etc). It ends up looking a lot like a Gantt 
chart.

If you'd like, I can sanitize and send one to you, to compare notes and see if 
there is anything of use to you.

Also, If time permits, and it's feasible,  I like to stage a mock upgrade in my 
lab with customer data (drs ... etc) and do a dry run.

 Original message 
From: Anthony Holloway  
Date:10/16/2015  2:38 PM  (GMT-05:00) 
To: Cisco VoIP Group  
Subject: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade? 

Does anyone else do this?  Gather all of the documentation ahead of time, 
because inevitably you're going to revisit a document more than once?  There 
are a lot of documents to gather!  Anything I could be doing better?  Tips?  
Tricks?

I create a spreadsheet of all of the pertinent documents I need to review or 
reference, like in this screenshot.  There's over 90 documents in this list.  
Granted, I don't read them all front to back, but some I do, and  for others I 
need to reference information within them nonetheless.  You never know when you 
might find a small font hidden note in there.

E.g., From the 8945 Release Notes

"Release 9.4(2)SR1 can only be upgraded from 9.3(4) and later. Releases prior 
to 9.3(4) have to be upgraded to 9.3(4) first."

Source: 8945 9.4(2)SR1 Release Notes

I actually missed this one recently, and unlike 7900 series phones, they phone 
will just brick itself and never register.  Causing you to walk to every phone 
and reset power to it, or walk the mac address tables of your layer 2 network 
and shut/no shut the ports.


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Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

Yeah, I am mobile at the moment,  Once I get home I'll clean one up and send it 
out to the list.

Thanks,

Ryan


Sent from my T-Mobile 4G LTE Device

 Original message 
From: Terry Oakley <terry.oak...@rdc.ab.ca> 
Date:10/16/2015  3:05 PM  (GMT-05:00) 
To: 'Ryan Huff' <ryanh...@outlook.com>,Anthony Holloway 
<avholloway+cisco-v...@gmail.com>,Cisco VoIP Group <cisco-voip@puck.nether.net> 
Subject: RE: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade? 

I would certainly be interested in a sanitized look.   Upgrades are fun (yeah 
right) but having some sort of tool to assist would at least make the light 
appear a little clearer.
 
 
 
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: October 16, 2015 12:57 PM
To: Anthony Holloway <avholloway+cisco-v...@gmail.com>; Cisco VoIP Group 
<cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?
 
I use an excel spread sheet with a hyperlink to the base doc in one sheet with 
notes and details gathered in the sheet.Then I create additional worksheets of 
subordinate documentation and notes and then make references from the base 
sheet to the subordinate sheets. I also have a sheet for customer discovery 
(current dns, ip, device loads  etc). It ends up looking a lot like a Gantt 
chart.
 
If you'd like, I can sanitize and send one to you, to compare notes and see if 
there is anything of use to you.
 
Also, If time permits, and it's feasible,  I like to stage a mock upgrade in my 
lab with customer data (drs ... etc) and do a dry run.


 Original message 
From: Anthony Holloway 
Date:10/16/2015 2:38 PM (GMT-05:00) 
To: Cisco VoIP Group 
Subject: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

Does anyone else do this?  Gather all of the documentation ahead of time, 
because inevitably you're going to revisit a document more than once?  There 
are a lot of documents to gather!  Anything I could be doing better?  Tips?  
Tricks?
 
I create a spreadsheet of all of the pertinent documents I need to review or 
reference, like in this screenshot.  There's over 90 documents in this list.  
Granted, I don't read them all front to back, but some I do, and  for others I 
need to reference information within them nonetheless.  You never know when you 
might find a small font hidden note in there.
 
E.g., From the 8945 Release Notes
 
"Release 9.4(2)SR1 can only be upgraded from 9.3(4) and later. Releases prior 
to 9.3(4) have to be upgraded to 9.3(4) first."
 
Source: 8945 9.4(2)SR1 Release Notes
 
I actually missed this one recently, and unlike 7900 series phones, they phone 
will just brick itself and never register.  Causing you to walk to every phone 
and reset power to it, or walk the mac address tables of your layer 2 network 
and shut/no shut the ports.
 
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Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade?

[Ryan Huff]

Typically, the only time that I'll stage an upgrade in a lab environment for a 
dry run is if I am doing something that I haven't done in a while or I have 
reason to think it might fail (you can screw up as much as you want in the lab 
;). I used to do it for almost every major upgrade.

I am a fan of doing the upgrade locally, grabbing the DRS from the local 
(upgraded) environment then restoring to a newly built environment in the 
customer network. This is a habit that extends from the pizza box days. 
However, with everything being virtualized now, I have become a fan of doing it 
all in the customer network.

I think each method has its pros and cons and the context of the customer 
engagement will generally point you to which method is better or more efficient 
for you. 

Thanks,

Ryan

Sent from my T-Mobile 4G LTE Device

 Original message 
From: Anthony Holloway <avholloway+cisco-v...@gmail.com> 
Date:10/16/2015  3:41 PM  (GMT-05:00) 
To: Ryan Huff <ryanh...@outlook.com> 
Cc: Cisco VoIP Group <cisco-voip@puck.nether.net> 
Subject: Re: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade? 

You sound more organized than I am.  I would like to see what you have, sure.  
Thanks for the offer.

I've never staged an upgrade in my lab, though I have heard of plenty of people 
doing this.  Is it really something to consider or is that a thing of the past? 
 Like pulling a drive from the array?  Not too mention, I rarely have time to 
perform two upgrades on a project like this.  I barely get enough time to 
upgrade the system once.

On Fri, Oct 16, 2015 at 1:56 PM, Ryan Huff <ryanh...@outlook.com> wrote:
I use an excel spread sheet with a hyperlink to the base doc in one sheet with 
notes and details gathered in the sheet.Then I create additional worksheets of 
subordinate documentation and notes and then make references from the base 
sheet to the subordinate sheets. I also have a sheet for customer discovery 
(current dns, ip, device loads  etc). It ends up looking a lot like a Gantt 
chart.

If you'd like, I can sanitize and send one to you, to compare notes and see if 
there is anything of use to you.

Also, If time permits, and it's feasible,  I like to stage a mock upgrade in my 
lab with customer data (drs ... etc) and do a dry run.


 Original message 
From: Anthony Holloway 
Date:10/16/2015 2:38 PM (GMT-05:00) 
To: Cisco VoIP Group 
Subject: [cisco-voip] How Many Docs Does it Take to Prep for an Upgrade? 

Does anyone else do this?  Gather all of the documentation ahead of time, 
because inevitably you're going to revisit a document more than once?  There 
are a lot of documents to gather!  Anything I could be doing better?  Tips?  
Tricks?

I create a spreadsheet of all of the pertinent documents I need to review or 
reference, like in this screenshot.  There's over 90 documents in this list.  
Granted, I don't read them all front to back, but some I do, and  for others I 
need to reference information within them nonetheless.  You never know when you 
might find a small font hidden note in there.

E.g., From the 8945 Release Notes

"Release 9.4(2)SR1 can only be upgraded from 9.3(4) and later. Releases prior 
to 9.3(4) have to be upgraded to 9.3(4) first."

Source: 8945 9.4(2)SR1 Release Notes

I actually missed this one recently, and unlike 7900 series phones, they phone 
will just brick itself and never register.  Causing you to walk to every phone 
and reset power to it, or walk the mac address tables of your layer 2 network 
and shut/no shut the ports.



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Re: [cisco-voip] Outbound IVR-UCCX and CUBE

[Ryan Huff]

Long shot here but have you double checked that voice activity detection (VAD) 
is off?
 
Thanks,
 
Ryan
 
From: nickolasjbr...@gmail.com
Date: Thu, 15 Oct 2015 14:14:59 +1100
To: rlafo...@cisco.com
Subject: Re: [cisco-voip] Outbound IVR-UCCX and CUBE
CC: cisco-voip@puck.nether.net

Sorry to Grave dig here, trying to configure CPA on UCCX on 10.6(1.1.1) 
with a CISCO2921/K9 on  Version 15.4(3)M4, RELEASE SOFTWARE (fc1).

It appears CSCui62525 was resolved on our version of UCCX but we tried removing 
RTP-NTE from all dialpeers with no luck (no update messages being sent from the 
GW to UCCX to advise that a live human is speaking connected)

Anything else we should check? TAC is next :(


On Fri, Apr 11, 2014 at 4:12 AM, Ryan LaFountain (rlafount) 
 wrote:





Hi Michele, 



As Bill said, only TDM is supported at this time. Although CUBE recently 
released support for CPA in DSPs that is not involved in call termination, 
there are still some things we have to work out. 



You can track CSCui62525. When this gets integrated I hope we’ll support CUBE 
for IVR-based Outbound Dialer with UCCX. 



You can also see this article which talks a little bit more about it. 



http://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express/116084-trouble-ivr-dialer-00.html


Thank you, 



Ryan LaFountain

Unified Contact Center

Cisco Services

Direct: +1 919 392 9898

Hours: M - F 9:00am - 5:00pm Eastern Time





On Apr 10, 2014, at 12:41 PM, Bill Talley  wrote:




PSTN access has to be TDM.  SIP trunking to telco is not supported in CCX 
9.0(2) for outbound dialer functionality.











On Thu, Apr 10, 2014 at 12:11 PM, Michele Russo (AM) 
 wrote:



I am configuring an Outbound IVR Progressive Campaign using UCCX version 9.0.2 
and CUBE.  Is this supported? I saw a note in this got a bit worried. 

http://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express/116084-trouble-ivr-dialer-00.html#anc8
 

Thanks.

 
 
Michele Russo Harttree
Consultant
Dimension Data NA
11730 Plaza America Drive Suite 350
Reston, Va 20190
202-460-3965 (cell)
571-203-4007 (desk)
michele.ru...@dimensiondata.com
 



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-- 
- Nick



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Re: [cisco-voip] Cisco CAR DB not running

[Ryan Huff]

Is it just sitting in the Starting state?
Is this the only service that won't start? 
Everything else in the cluster seems healthy (i.e no replication issues, the CM 
Server list reflects the new IP address, dns forward/reverse zones resolve the 
new address ... etc)?

Thanks,

Ryan

> On Oct 14, 2015, at 10:03 AM, Erick Bergquist  wrote:
> 
> Hello,
> 
> Does anyone know any tricks to get Cisco CAR DB started?  I can't get
> it to start and there was a IP address change done.
> 
> Call Manager version 9.1.2
> 
> Erick
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Re: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup.

[Ryan Huff]

Terry,

Sounds like you have a lot going on there!

How did you move into the 10.5 environment?  Did you do a bridge migration or a 
'stare and compare'?

A fast busy could be a few different things (css, partition ... etc) or dns 
based since you mentioned fqdn or resource based.

What codec are you trying using?

Have you pulled traces?

What is the disconnect cause code for one of the failed calls into the hunt 
pilot?

If you can reproduce a failed call and then send me the traces or the sip 
messages I can give you a much better answer.

Thanks,

Ryan


Sent from my T-Mobile 4G LTE Device

 Original message 
From: Terry Oakley  
Date:10/13/2015  6:24 PM  (GMT-05:00) 
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup. 

We currently are moving from Exchange 2007 to Exchange 2013.   We have three 
CAS servers and 3 Mailbox servers, all virtualized.   In our test environment, 
before we moved from CUCM 6.1 to 10.5 we are able to at least get Exchange 2013 
to answer a SIP trunk request from CUCM 6.1.   Now in CUCM 10.5 we just get a 
fast busy when we dial the VM pilot number.   Does anyone have experience with 
this and have a guide that we could follow?    We have followed a number of 
guides from Microsoft and they have not proven to be the magic answer.
 
We have a SIP trunk set to the CAS servers with all three individual servers 
listed in the Destination section (all FQDN) port 5060
We have three separate SIP trunks to the three mailbox servers with all three 
having the ports 5062 through 5068 listed and again FQDN for the destination 
address. 
The VM pilot (route pattern) is associated with the CAS trunk. 
Do we need a route list and hence a route group?
 
Thank you for your knowledge and wiliness to share.  And especially thanks to 
this forum for providing us the access.  
 
Cheers
 
Terry
 
Terry Oakley
Telecommunications Coordinator | Information Technology Services
Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5
work (403) 342-3521   |  FAX (403) 343-4034
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Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

Although it appears you have it enabled, not having the network-clock 
participation set correctly will render the network-clock synchronization 
ineffective for the NIM with the port your trying to clock source on.

http://www.cisco.com/c/en/us/support/docs/routers/4000-series-integrated-services-routers/118792-config-isr-00.html

Sent from my iPad

> On Oct 11, 2015, at 1:18 AM, Jonathan Charles <jonv...@gmail.com> wrote:
> 
> I put the primary in and set the input source... still getting slips...
> 
> 
> Jonathan
> 
>> On Sat, Oct 10, 2015 at 11:08 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> This is how I've setup clocking on 4th gens before, and it always seems to 
>> use the line as the clock source when I have done it this way (network clock 
>> synchronization is disabled by default, so you may double check that it is 
>> enabled). 
>> 
>> Using your slot reference ...
>> 
>> rtr(config)# network-clock participation 0/1
>> rtr(config)# network-clock synchronization automatic
>> rtr(config)# controller t1 0/1/0
>> rtr(config)# clock source line primary
>> rtr(config)# network-clock input-source 1 controller t1 0/1/0
>> 
>> I have also found this guide to be useful: 
>> http://www.cisco.com/c/en/us/td/docs/routers/access/4400/feature/guide/isr4400netclock.html#pgfId-1023586
>> 
>> Thanks,
>> 
>> Ryan
>> 
>> Sent from my iPad
>> 
>>> On Oct 10, 2015, at 10:49 PM, Jonathan Charles <jonv...@gmail.com> wrote:
>>> 
>>> ILVOIPISR01(config)#network-clock input-source 1 controller T1 0/1/0 
>>> % The clock source should be "line"
>>> 
>>> They are all set to line... however:
>> 
>>> 
>>> Why does it say internal:
>>> 
>>> ILNOVOIPISR01#show network-clocks synchronization 
>>> Symbols: En - Enable, Dis - Disable, Adis - Admin Disable 
>>>  NA - Not Applicable 
>>>  *  - Synchronization source selected 
>>>  #  - Synchronization source force selected 
>>>  &  - Synchronization source manually switched 
>>> 
>>> Automatic selection process : Enable
>>> Equipment Clock : 2048 (EEC-Option1)
>>> Clock Mode : QL-Disable
>>> ESMC : Disabled
>>> SSM Option : 1 
>>> T0 : Internal 
>>> Hold-off (global) : 300 ms
>>> Wait-to-restore (global) : 300 sec
>>> Tsm Delay : 180 ms
>>> Revertive : No
>>> 
>>> Nominated Interfaces
>>> 
>>>  InterfaceSigType Mode/QL  Prio  QL_IN  ESMC Tx  ESMC Rx
>>> *Internal NA  NA/Dis   251   QL-SECNANA 
>>>   
>>> ILNOVOIPISR01#
>>> 
>>> 
>>> 
>>> 
>>> Jonathan
>>> 
>>>> On Sat, Oct 10, 2015 at 9:11 PM, NateCCIE . <natec...@gmail.com> wrote:
>>>> Cisco changed everything again.  If you remember network-clock select with 
>>>> ISR G1, or the AIM-VOICE-30… When those first came out clocks were 
>>>> slipping all over the place with clock source line, because that wasn’t 
>>>> enough.
>>>> 
>>>>  
>>>> 
>>>> I am sorry if I am just behind everyone else and you’ve already done T1s 
>>>> on an ISR4k
>>>> 
>>>>  
>>>> 
>>>> http://www.cisco.com/c/en/us/support/docs/routers/4000-series-integrated-services-routers/118792-config-isr-00.html
>>>> 
>>>>  
>>>> 
>>>> I think this is right:
>>>> 
>>>>  
>>>> 
>>>> controller T1 0/1/0
>>>> 
>>>> framing esf
>>>> 
>>>> clock source line primary
>>>> 
>>>> linecode b8zs
>>>> 
>>>> cablelength long 0db
>>>> 
>>>> pri-group timeslots 1-24
>>>> 
>>>> !
>>>> 
>>>> controller T1 0/1/1
>>>> 
>>>> framing esf
>>>> 
>>>> clock source line secondary
>>>> 
>>>> linecode b8zs
>>>> 
>>>> cablelength long 0db
>>>> 
>>>> pri-group timeslots 1-24
>>>> 
>>>>  
>>>> 
>>>> network-clock synchronization automatic
>>>> 
>>>> network-clock input-source 1 controller T1 0/1/0
>>>> 
>>>> network-clock input-source 2 controller T1 0/1/1
>>>> 
>>>> 
>>>> 
&

Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

What is the output of show platform hardware subslot 0/1 module device 
networkclock

assuming the slot/subslot of the desired line clock is on 0/1

Sent from my iPad

> On Oct 11, 2015, at 12:42 PM, Jonathan Charles <jonv...@gmail.com> wrote:
> 
> Yeah, but I can't enable it:
> 
> 
> ILNOVOIPISR01(config)#network-clock participate 0 
> G.781 based clock selection process is enabled Please unconfigure G.781 based 
> configuration before configuring network-clock participate config command
> ILNOVOIPISR01(config)#network-clock participate 1 
> G.781 based clock selection process is enabled Please unconfigure G.781 based 
> configuration before configuring network-clock participate config command
> ILNOVOIPISR01(config)#
> 
> 
> 
> Jonathan
> 
>> On Sun, Oct 11, 2015 at 8:58 AM, Ryan Huff <ryanh...@outlook.com> wrote:
>> Although it appears you have it enabled, not having the network-clock 
>> participation set correctly will render the network-clock synchronization 
>> ineffective for the NIM with the port your trying to clock source on.
>> 
>> http://www.cisco.com/c/en/us/support/docs/routers/4000-series-integrated-services-routers/118792-config-isr-00.html
>> 
>> Sent from my iPad
>> 
>>> On Oct 11, 2015, at 1:18 AM, Jonathan Charles <jonv...@gmail.com> wrote:
>>> 
>>> I put the primary in and set the input source... still getting slips...
>>> 
>>> 
>>> Jonathan
>>> 
>>>> On Sat, Oct 10, 2015 at 11:08 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>>>> This is how I've setup clocking on 4th gens before, and it always seems to 
>>>> use the line as the clock source when I have done it this way (network 
>>>> clock synchronization is disabled by default, so you may double check that 
>>>> it is enabled). 
>>>> 
>>>> Using your slot reference ...
>>>> 
>>>> rtr(config)# network-clock participation 0/1
>>>> rtr(config)# network-clock synchronization automatic
>>>> rtr(config)# controller t1 0/1/0
>>>> rtr(config)# clock source line primary
>>>> rtr(config)# network-clock input-source 1 controller t1 0/1/0
>>>> 
>>>> I have also found this guide to be useful: 
>>>> http://www.cisco.com/c/en/us/td/docs/routers/access/4400/feature/guide/isr4400netclock.html#pgfId-1023586
>>>> 
>>>> Thanks,
>>>> 
>>>> Ryan
>>>> 
>>>> Sent from my iPad
>>>> 
>>>>> On Oct 10, 2015, at 10:49 PM, Jonathan Charles <jonv...@gmail.com> wrote:
>>>>> 
>>>>> ILVOIPISR01(config)#network-clock input-source 1 controller T1 0/1/0 
>>>>> % The clock source should be "line"
>>>>> 
>>>>> They are all set to line... however:
>>>> 
>>>>> 
>>>>> Why does it say internal:
>>>>> 
>>>>> ILNOVOIPISR01#show network-clocks synchronization 
>>>>> Symbols: En - Enable, Dis - Disable, Adis - Admin Disable 
>>>>>  NA - Not Applicable 
>>>>>  *  - Synchronization source selected 
>>>>>  #  - Synchronization source force selected 
>>>>>  &  - Synchronization source manually switched 
>>>>> 
>>>>> Automatic selection process : Enable
>>>>> Equipment Clock : 2048 (EEC-Option1)
>>>>> Clock Mode : QL-Disable
>>>>> ESMC : Disabled
>>>>> SSM Option : 1 
>>>>> T0 : Internal 
>>>>> Hold-off (global) : 300 ms
>>>>> Wait-to-restore (global) : 300 sec
>>>>> Tsm Delay : 180 ms
>>>>> Revertive : No
>>>>> 
>>>>> Nominated Interfaces
>>>>> 
>>>>>  InterfaceSigType Mode/QL  Prio  QL_IN  ESMC Tx  ESMC 
>>>>> Rx
>>>>> *Internal NA  NA/Dis   251   QL-SECNA
>>>>> NA   
>>>>> ILNOVOIPISR01#
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Jonathan
>>>>> 
>>>>>> On Sat, Oct 10, 2015 at 9:11 PM, NateCCIE . <natec...@gmail.com> wrote:
>>>>>> Cisco changed everything again.  If you remember network-clock select 
>>>>>> with ISR G1, or the AIM-VOICE-30… When those first came out clocks were 
>>>>>> slipping all over the place w

Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

If there are no calls active (and haven't been for awhile) the slips will 
usually clear. However, it will likely start slipping again once calls start 
signaling (assuming there is still an underlying issue with network clock 
synchronization).

Sent from my iPad

> On Oct 11, 2015, at 5:02 PM, Jonathan Charles <jonv...@gmail.com> wrote:
> 
> The slips cleared up and I am not sure why it may be that there are no 
> calls on them... 
> 
> Thanks!
> 
> 
> Jonathan
> 
>> On Sun, Oct 11, 2015 at 3:23 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> Here is the relevant portion of a isr4k humming along as we speak, with no 
>> clock slippage (excuse the scrunchiness of the output, SSHTerm on my iPad is 
>> limited and there is a Rocky marathon on TV ... so I ain't moving ;) ). 
>> Granted, this is one PRI and not 10 in two different modules but the concept 
>> should be the same.
>> 
>> The other thought I'd have is to make sure you are giving your desired clock 
>> interface the higher priority and not to have two clock sources at the same 
>> priority as this can cause issues with switch messaging.
>> 
>> card type t1 0 2
>> !
>> isdn switch-type primary-ni
>> !
>> network-clock synchronization automatic
>> network-clock synchronization participate 0/2
>> !
>> voice-card 0/2
>> dsp services dspfarm
>> !
>> controller T1 0/2/0
>> framing esf
>> linecode b8zs
>> !
>> clock source line primary
>> network-clock input-source 1 controller t1 0/2/0
>> cablelength long 0db
>> pri-group timeslots 1-24 voice-dsp
>> !
>> interface Serial0/2/0:23
>> encapsulation hdlc
>> isdn switch-type primary-ni
>> no cdp enable
>> !
>> rthlabrtr# show network-clocks synchronization
>> Symbols: En - Enable, Dis - Disable, Adis - Admin Disable 
>> NA - Not Applicable 
>> * - Synchronization source selected 
>> # - Synchronization source force selected 
>> & - Synchronization source manually switched 
>>  
>> Automatic selection process : Enable
>> Equipment Clock : 2048 (EEC-Option1)
>> Clock Mode : QL-Disable
>> ESMC : Disabled
>> SSM Option : 1 
>> T1 : 0/2/0
>> Hold-off (global) : 300 ms
>> Wait-to-restore (global) : 300 sec
>> Tsm Delay : 180 ms
>> Revertive : No
>>  
>> Nominated Interfaces
>>  
>> Interface SigType Mode/QL Prio QL_IN ESMC Tx ESMC Rx
>> Internal NA NA/Dis 251 QL-SEC NA NA 
>> *T1 0/2/0 NA NA/Dis 1 QL-SEC NA NA 
>>  
>> Sent from my iPad
>> 
>>> On Oct 11, 2015, at 3:15 PM, Jonathan Charles <jonv...@gmail.com> wrote:
>>> 
>>> I think the automatic already handles that
>>> 
>>> ILNOVOIPISR01(config)#network-clock synchronization participate 0/1  
>>> Slot 0 subslot 1 is already enabled for network clocking. Command Aborted.
>>> ILNOVOIPISR01(config)#
>>> 
>>> 
>>> 
>>> Jonathan
>>> 
>>>> On Sun, Oct 11, 2015 at 12:41 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>>>> I'm assuming you also have network-clock synchronization participate 0/1 
>>>> in your configuration?
>>>> 
>>>>  
>>>> Date: Sun, 11 Oct 2015 12:31:05 -0500
>>>> Subject: Re: [cisco-voip] slips on NIM-8MFT-T1/E1
>>>> From: jonv...@gmail.com
>>>> To: ryanh...@outlook.com
>>>> CC: natec...@gmail.com; cisco-voip@puck.nether.net
>>>> 
>>>> 
>>>> ILNOVOIPISR01#show platform hardware subslot 0/1 module device networkclock
>>>> 
>>>>  primary clock 0, secondary clock 255, ntwk_clk_selected Yes, 
>>>> ntwk_clk_participate Yes, current clock = 0
>>>> PLL status= 1, and cntl = 17 
>>>> 
>>>> 
>>>> ILNOVOIPISR01#
>>>> 
>>>> 
>>>> ILNOVOIPISR01#show platform hardware subslot 0/2 module device networkclock
>>>> 
>>>>  primary clock 0, secondary clock 255, ntwk_clk_selected No, 
>>>> ntwk_clk_participate Yes, current clock = 0
>>>> PLL status= 1, and cntl = 7 
>>>> 
>>>> 
>>>> ILNOVOIPISR01#
>>>> 
>>>> 
>>>> 
>>>> Jonathan
>>>> 
>>>> On Sun, Oct 11, 2015 at 11:53 AM, Ryan Huff <ryanh...@outlook.com> wrote:
>>>> What is the output of show platform hardware subslot 0/1 module device 
>>>> networkclock
>>>> 
>>>> assuming the slot/subslot of the desired line cloc

Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

Here is the relevant portion of a isr4k humming along as we speak, with no 
clock slippage (excuse the scrunchiness of the output, SSHTerm on my iPad is 
limited and there is a Rocky marathon on TV ... so I ain't moving ;) ). 
Granted, this is one PRI and not 10 in two different modules but the concept 
should be the same.

The other thought I'd have is to make sure you are giving your desired clock 
interface the higher priority and not to have two clock sources at the same 
priority as this can cause issues with switch messaging.

card type t1 0 2
!
isdn switch-type primary-ni
!
network-clock synchronization automatic
network-clock synchronization participate 0/2
!
voice-card 0/2
dsp services dspfarm
!
controller T1 0/2/0
framing esf
linecode b8zs
!
clock source line primary
network-clock input-source 1 controller t1 0/2/0
cablelength long 0db
pri-group timeslots 1-24 voice-dsp
!
interface Serial0/2/0:23
encapsulation hdlc
isdn switch-type primary-ni
no cdp enable
!
rthlabrtr# show network-clocks synchronization
Symbols: En - Enable, Dis - Disable, Adis - Admin Disable 
NA - Not Applicable 
* - Synchronization source selected 
# - Synchronization source force selected 
& - Synchronization source manually switched 
 
Automatic selection process : Enable
Equipment Clock : 2048 (EEC-Option1)
Clock Mode : QL-Disable
ESMC : Disabled
SSM Option : 1 
T1 : 0/2/0
Hold-off (global) : 300 ms
Wait-to-restore (global) : 300 sec
Tsm Delay : 180 ms
Revertive : No
 
Nominated Interfaces
 
Interface SigType Mode/QL Prio QL_IN ESMC Tx ESMC Rx
Internal NA NA/Dis 251 QL-SEC NA NA 
*T1 0/2/0 NA NA/Dis 1 QL-SEC NA NA 
 
Sent from my iPad

> On Oct 11, 2015, at 3:15 PM, Jonathan Charles <jonv...@gmail.com> wrote:
> 
> I think the automatic already handles that
> 
> ILNOVOIPISR01(config)#network-clock synchronization participate 0/1  
> Slot 0 subslot 1 is already enabled for network clocking. Command Aborted.
> ILNOVOIPISR01(config)#
> 
> 
> 
> Jonathan
> 
>> On Sun, Oct 11, 2015 at 12:41 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> I'm assuming you also have network-clock synchronization participate 0/1 in 
>> your configuration?
>> 
>>  
>> Date: Sun, 11 Oct 2015 12:31:05 -0500
>> Subject: Re: [cisco-voip] slips on NIM-8MFT-T1/E1
>> From: jonv...@gmail.com
>> To: ryanh...@outlook.com
>> CC: natec...@gmail.com; cisco-voip@puck.nether.net
>> 
>> 
>> ILNOVOIPISR01#show platform hardware subslot 0/1 module device networkclock
>> 
>>  primary clock 0, secondary clock 255, ntwk_clk_selected Yes, 
>> ntwk_clk_participate Yes, current clock = 0
>> PLL status= 1, and cntl = 17 
>> 
>> 
>> ILNOVOIPISR01#
>> 
>> 
>> ILNOVOIPISR01#show platform hardware subslot 0/2 module device networkclock
>> 
>>  primary clock 0, secondary clock 255, ntwk_clk_selected No, 
>> ntwk_clk_participate Yes, current clock = 0
>> PLL status= 1, and cntl = 7 
>> 
>> 
>> ILNOVOIPISR01#
>> 
>> 
>> 
>> Jonathan
>> 
>> On Sun, Oct 11, 2015 at 11:53 AM, Ryan Huff <ryanh...@outlook.com> wrote:
>> What is the output of show platform hardware subslot 0/1 module device 
>> networkclock
>> 
>> assuming the slot/subslot of the desired line clock is on 0/1
>> 
>> Sent from my iPad
>> 
>> On Oct 11, 2015, at 12:42 PM, Jonathan Charles <jonv...@gmail.com> wrote:
>> 
>> Yeah, but I can't enable it:
>> 
>> 
>> ILNOVOIPISR01(config)#network-clock participate 0 
>> G.781 based clock selection process is enabled Please unconfigure G.781 
>> based configuration before configuring network-clock participate config 
>> command
>> ILNOVOIPISR01(config)#network-clock participate 1 
>> G.781 based clock selection process is enabled Please unconfigure G.781 
>> based configuration before configuring network-clock participate config 
>> command
>> ILNOVOIPISR01(config)#
>> 
>> 
>> 
>> Jonathan
>> 
>> On Sun, Oct 11, 2015 at 8:58 AM, Ryan Huff <ryanh...@outlook.com> wrote:
>> Although it appears you have it enabled, not having the network-clock 
>> participation set correctly will render the network-clock synchronization 
>> ineffective for the NIM with the port your trying to clock source on.
>> 
>> http://www.cisco.com/c/en/us/support/docs/routers/4000-series-integrated-services-routers/118792-config-isr-00.html
>> 
>> Sent from my iPad
>> 
>> On Oct 11, 2015, at 1:18 AM, Jonathan Charles <jonv...@gmail.com> wrote:
>> 
>> I put the primary in and set the input source... still getting slips...
>> 
>> 
>> Jonathan
>> 
>> On Sat, Oct 1

Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

This is how I've setup clocking on 4th gens before, and it always seems to use 
the line as the clock source when I have done it this way (network clock 
synchronization is disabled by default, so you may double check that it is 
enabled). 

Using your slot reference ...

rtr(config)# network-clock participation 0/1
rtr(config)# network-clock synchronization automatic
rtr(config)# controller t1 0/1/0
rtr(config)# clock source line primary
rtr(config)# network-clock input-source 1 controller t1 0/1/0

I have also found this guide to be useful: 
http://www.cisco.com/c/en/us/td/docs/routers/access/4400/feature/guide/isr4400netclock.html#pgfId-1023586

Thanks,

Ryan

Sent from my iPad

> On Oct 10, 2015, at 10:49 PM, Jonathan Charles  wrote:
> 
> ILVOIPISR01(config)#network-clock input-source 1 controller T1 0/1/0 
> % The clock source should be "line"
> 
> They are all set to line... however:

> 
> Why does it say internal:
> 
> ILNOVOIPISR01#show network-clocks synchronization 
> Symbols: En - Enable, Dis - Disable, Adis - Admin Disable 
>  NA - Not Applicable 
>  *  - Synchronization source selected 
>  #  - Synchronization source force selected 
>  &  - Synchronization source manually switched 
> 
> Automatic selection process : Enable
> Equipment Clock : 2048 (EEC-Option1)
> Clock Mode : QL-Disable
> ESMC : Disabled
> SSM Option : 1 
> T0 : Internal 
> Hold-off (global) : 300 ms
> Wait-to-restore (global) : 300 sec
> Tsm Delay : 180 ms
> Revertive : No
> 
> Nominated Interfaces
> 
>  InterfaceSigType Mode/QL  Prio  QL_IN  ESMC Tx  ESMC Rx
> *Internal NA  NA/Dis   251   QL-SECNANA   
> 
> ILNOVOIPISR01#
> 
> 
> 
> 
> Jonathan
> 
>> On Sat, Oct 10, 2015 at 9:11 PM, NateCCIE .  wrote:
>> Cisco changed everything again.  If you remember network-clock select with 
>> ISR G1, or the AIM-VOICE-30… When those first came out clocks were slipping 
>> all over the place with clock source line, because that wasn’t enough.
>> 
>>  
>> 
>> I am sorry if I am just behind everyone else and you’ve already done T1s on 
>> an ISR4k
>> 
>>  
>> 
>> http://www.cisco.com/c/en/us/support/docs/routers/4000-series-integrated-services-routers/118792-config-isr-00.html
>> 
>>  
>> 
>> I think this is right:
>> 
>>  
>> 
>> controller T1 0/1/0
>> 
>> framing esf
>> 
>> clock source line primary
>> 
>> linecode b8zs
>> 
>> cablelength long 0db
>> 
>> pri-group timeslots 1-24
>> 
>> !
>> 
>> controller T1 0/1/1
>> 
>> framing esf
>> 
>> clock source line secondary
>> 
>> linecode b8zs
>> 
>> cablelength long 0db
>> 
>> pri-group timeslots 1-24
>> 
>>  
>> 
>> network-clock synchronization automatic
>> 
>> network-clock input-source 1 controller T1 0/1/0
>> 
>> network-clock input-source 2 controller T1 0/1/1
>> 
>> 
>> 
>> Thanks,
>> 
>> -Nate
>> 
>> 
>>> On Sat, Oct 10, 2015 at 4:01 PM, Jonathan Charles  wrote:
>>> We have two NIM-8MFT-T1/E1s in a router for PRI termination...
>>> 
>>> On one of the 8-port cards, we have 7 PRIs in the same trunk group, on the 
>>> other, 3 PRIs in the same trunk group.
>>> 
>>> We are seeing slips on all T1s.
>>> 
>>> This is a Ciisco 4351.
>>> 
>>> We have network-clock synchronization automatic, and I cannot find any 
>>> better config
>>> 
>>> All T1s are configured as B8ZS, ESF, Clock Source Line...
>>> 
>>> 
>>> controller T1 0/1/0
>>>  framing esf
>>>  linecode b8zs
>>>  cablelength long 0db
>>>  pri-group timeslots 1-24
>>> !
>>> 
>>> 
>>> Any ideas?
>>> 
>>> 
>>> 
>>> Jonathan
>>> 
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> 
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Re: [cisco-voip] slips on NIM-8MFT-T1/E1

[Ryan Huff]

I got some time tonight to take a look if you'd like ...

Are all 10 pri's from the same carrier?

Can you send me a sh run, sh isdn status and a sh controllers T1.

Also, a q931/q921 debug for a call would be helpful (even if it is a failed 
call).

Sent from my T-Mobile 4G LTE Device

 Original message 
From: Jonathan Charles  
Date:10/10/2015  6:01 PM  (GMT-05:00) 
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] slips on NIM-8MFT-T1/E1 

We have two NIM-8MFT-T1/E1s in a router for PRI termination...

On one of the 8-port cards, we have 7 PRIs in the same trunk group, on the 
other, 3 PRIs in the same trunk group.

We are seeing slips on all T1s.

This is a Ciisco 4351.

We have network-clock synchronization automatic, and I cannot find any better 
config

All T1s are configured as B8ZS, ESF, Clock Source Line...


controller T1 0/1/0
 framing esf
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-24
!


Any ideas?



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Re: [cisco-voip] DRS Backups 10.5

[Ryan Huff]

I think the best option is (and has been for awhile, imho) to spin up or box 
your favorite Linux flavor with openssh  (Arch, Ubuntu, Debian) come to mind. 
Not hard to setup at all and they just run, almost a set-it-and-forget-it.

-Ryan

Sent from my iPad

> On Oct 8, 2015, at 11:05 AM, Jason Aarons (AM) 
>  wrote:
> 
> Being only GlobalScape and Titan are the only two commercial SFTP TAC 
> supported products is a source of contention. Non-technical customers don’t 
> want to compile anything. I wish FTP and CIFS/SMB was supported. Keep it 
> simple and encrypt the file, not the transmission.
>  
> I haven’t been able to duplicate the FreeFTPD problem of 1GB file limitation 
> in my lab, nor does FreeFTPD indicate they have any file size limitation, 
> it’s some sort of Cisco made up release notes limitation that just has TAC 
> wash their hands.
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Buchanan, James
> Sent: Thursday, October 8, 2015 5:03 AM
> To: Heim, Dennis ; norm.nichol...@kitchener.ca; 
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] DRS Backups 10.5
>  
>  
> 
> Actually, it is only SFTP, right? FTP is not supported (a continual pain 
> point for customers).
>  
>  
> 
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Heim, Dennis
> Sent: Wednesday, October 07, 2015 3:17 PM
> To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] DRS Backups 10.5
>  
> SFTP/FTP Is the supported mechanism.
>  
> Dennis Heim | Emerging Technology Architect (Collaboration)
> World Wide Technology, Inc. | +1 314-212-1814
> 
> 
> “There is a fine line between Wrong and Visionary. Unfortunately, you have to 
> be a visionary to see it." – Sheldon Cooper
> “The greatest danger for most of us is not that our aim is too high and we 
> miss it, but that it is too low and we reach it.” -- Michelangelo Buonarroti
> “We should tansform the way we work” -- RowanTrollope
>  
> Click here to join me in my Collaboration Meeting Room
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> norm.nichol...@kitchener.ca
> Sent: Wednesday, October 07, 2015 1:52 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] DRS Backups 10.5
>  
>  
> I have been asked if Cisco supports SMB or NFS for our nightly DRS backups 
> instead of SFTP.
>  
>  
>  
> Thanks
>  
>  
>  
> Norm Nicholson
> Telecom Analyst
> City of Kitchener
> (519) 741-2200 x 7000
>  
>  
>  
> This message w/attachments (message) is intended solely for the use of the 
> intended recipient(s) and may contain information that is privileged, 
> confidential or proprietary. If you are not an intended recipient, please 
> notify the sender, and then please delete and destroy all copies and 
> attachments. Please be advised that any review or dissemination of, or the 
> taking of any action in reliance on, the information contained in or attached 
> to this message is prohibited.
> 
> 
> itevomcid
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Re: [cisco-voip] Using speed dials to access features

[Ryan Huff]

I should further add my suggestion would mean removing the line on the phone 
and creating it as a hunt pilot and then putting a pseudo number on the phone 
in place of the line, as the line group member.

Sent from my iPad

> On Oct 8, 2015, at 2:54 PM, Ryan Huff <ryanh...@outlook.com> wrote:
> 
> Off the top  (and this is by no means elegant);
> 
> 1.) Create a hunt group with the pilot's CFWNA action to forward to the cell 
> phone (using a Pstn egress patter). 
> 2.) Make the line a member of the line group serviced by the hunt list/hunt 
> group.
> 3.) Use an Hlog button on the phone to toggle whether the line is active in 
> the line group or not.
> 
> You may need to play around with the diversion header settings and answer 
> timers a bit ... etc, but in theory this should work.
> 
> Sent from my iPad
> 
>> On Oct 8, 2015, at 2:38 PM, norm.nichol...@kitchener.ca wrote:
>> 
>>  
>>  
>> I have a request to program a button to forward a line on a 7965/7916 to a 
>> cell phone automatically by pressing one button.  Do features like CFwdALL 
>> have a dialable number associated with them ?
>>  
>>  
>>  
>> Thanks
>>  
>>  
>>  
>>  
>>  
>> Norm Nicholson
>> Telecom Analyst
>> City of Kitchener
>> (519) 741-2200 x 7000
>>  
>>  
>> ___
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Re: [cisco-voip] Using speed dials to access features

[Ryan Huff]

Off the top  (and this is by no means elegant);

1.) Create a hunt group with the pilot's CFWNA action to forward to the cell 
phone (using a Pstn egress patter). 
2.) Make the line a member of the line group serviced by the hunt list/hunt 
group.
3.) Use an Hlog button on the phone to toggle whether the line is active in the 
line group or not.

You may need to play around with the diversion header settings and answer 
timers a bit ... etc, but in theory this should work.

Sent from my iPad

> On Oct 8, 2015, at 2:38 PM, norm.nichol...@kitchener.ca wrote:
> 
>  
>  
> I have a request to program a button to forward a line on a 7965/7916 to a 
> cell phone automatically by pressing one button.  Do features like CFwdALL 
> have a dialable number associated with them ?
>  
>  
>  
> Thanks
>  
>  
>  
>  
>  
> Norm Nicholson
> Telecom Analyst
> City of Kitchener
> (519) 741-2200 x 7000
>  
>  
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Re: [cisco-voip] CDR on CM 10.5.2.12901-1

[Ryan Huff]

Nothing I am aware of. I would verify the CDR Flag Enable is enabled on ALL 
nodes.

Thanks,

Ryan

Sent from my iPad

> On Oct 8, 2015, at 2:05 PM, Scott Voll  wrote:
> 
> Any bugs on this version with CM not sending CDR's to reporting server?
> 
> our Reporting vendor is saying they are not getting the CDR's.
> 
> we just upgraded from 8.6.2 to 10.5.2.12901-1
> 
> TIA
> 
> Scott
> 
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Re: [cisco-voip] Background url

[Ryan Huff]

I believe you need to specify the fully qualified tftp path; at least you did when I last played with it.Sent from my T-Mobile 4G LTE Device Original message From: "Louis Koekemoer (ZA)"  Date:10/06/2015  5:02 PM  (GMT-05:00) To: Justin Steinberg  Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Background url 

So in my case I should only put
Background.png ? As I say it is a new installation and at the moment it is only 2 phones that I’m busy testing with. We will be rolling out phones in about a week’s time.
 

 
Kind regards
 
Louis Koekemoer
Principle Systems Engineer – Converged Communications
Dimension Data Middle East & Africa
Tel: +27 (11) 575 4317
Fax: +27 (11) 576 4317
Mobile: +27 (71) 680 8790
louis.koekem...@dimensiondata.com
Planned Leave –

Planned Travel –



 


From: Justin Steinberg [mailto:jsteinb...@gmail.com]

Sent: 06 October 2015 10:06 PM
To: Louis Koekemoer (ZA)
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Background url


 
 

i believe you just put the file name in that field, so like below.

 


Background.png


 


I will say that I did this for the first time on 9.12 via the common phone profile and applied the config.   It dropped every call in progress.   This was an early 9.1 and
 the phones were 8945s.   I never investigated in further but it was the first time that I had an 'apply config' action drop calls in progress.


 


Justin



 

On Tue, Oct 6, 2015 at 1:23 AM, Louis Koekemoer (ZA)  wrote:


 

Hi all,
 
I’m busy with a deployment of 8841 phones. I created a background image as per the documentation and it works fine if you go to the physical phone and select settings>Wallpaper
 and set the new Background. I do however want to set this “globally” for all the phones. So on the 8841’s there is a Background Image option towards the bottom of the phone’s config. It is also in the Common Phone Profile. I tried setting it as I did in the
 List.xml file and a few other ways, without luck. Any Ideas what it should look like?
 
 
In the list.xml file it looks like this:

 
I tried the following. It does not accept the “ like above.
/Desktops/800x480x24/Background.png
TFTP:Desktops/800x480x24/Background.png
 
 
 

Kind regards
 
Louis Koekemoer
Principle Systems Engineer – Converged Communications
Dimension Data Middle East & Africa
Tel:
+27 (11) 575 4317
Fax:
+27 (11) 576 4317
Mobile:
+27 (71) 680 8790
louis.koekem...@dimensiondata.com
Planned Leave – 24/09/2015 – 27/09/2015
Planned Travel –


 



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itevomcid 



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Re: [cisco-voip] switch version from CUCM 9.1(1) to 10.5(2)

[Ryan Huff]

My suspicion (and this is just a guess really, without more detail); is that 
after the 10.5 update was applied, but before the switch version was completed, 
changes where made to the data in the 9.1 version.

When staging an upgrade in the inactive partition, changes made to the database 
in the active partition are not replicated to the database in the inactive 
partition. When you complete the switch version, the database from the inactive 
partition (created when you initially installed the upgrade) becomes active and 
any data differences between the databases are lost.

The same thing would happen if you had to revert the upgrade and switch-version 
back to 9.1. Any changes made in 10.5 would not replicate into 9.1.

Thanks,

Ryan

Sent from my iPad

> On Oct 6, 2015, at 3:20 AM, Abebe Amare  wrote:
> 
> Hi,
> 
> I upgraded CUCM over the weekend from version 9.1 to 10.5(2) and applied the 
> upgrade license on the PLM for version 10.x .
> I noticed some EM profiles, end users, phones and BLF speed dials disappeared 
> after the version switch over. Fortunately those missing items were not much 
> so I added them manually. what might have caused the configuration missing ?
> 
> thanks in advance
> 
> Abebe
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[cisco-voip] Odd call behavior CME,CUBE,CUCM

[Ryan Huff]

I have CME that has a SIP phone registered to it and a SCCP phone registered to 
it.
SIP=4002
SCCP=4001
 
I have a call path that looks like:
 
CUCM->SIPTrunk->CUBE->SIPTrunk->CME
 
When I dial the SIP phone (on CME) from call manager the call completes and 
goes fine. When I dial the skinny phone (on CME) from call manager the call 
immediately fails. However, from CME, using the skinny phone, I can dial 
anything in call manager and the call completes.
 
CCSIP on CME (for one of the failed calls) shows the INVITE then TRYING then 
SIP/2.0 500 Internal Server Error. CCAPI INOUT shows a cause code of 16, but 
this was anything but a normal clearing. Just for the hell of it, I also added 
a transcoder in CME and registered it in telephony-service, just to see if 
there was a codec flying around that I wasn't seeing  but alas, no.
 
I've attached the ccsip and ccapi inout; I think I have just been starring at 
it too long ... I can't see it.

Thanks,

Ryan
 
  Syslog logging: enabled (0 messages dropped, 8 messages rate-limited, 0 
flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.



No Inactive Message Discriminator.


Console logging: disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
 filtering disabled
Buffer logging:  level debugging, 298 messages logged, xml disabled,
filtering disabled
Exception Logging: size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

Trap logging: level informational, 106 message lines logged
Logging Source-Interface:   VRF Name:

Log Buffer (1000 bytes):

Oct  6 02:30:47.397: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:4001@142.102.66.254:5060 SIP/2.0
Via: SIP/2.0/UDP 142.102.65.254:5060;branch=z9hG4bKBDD75
Remote-Party-ID: "Site B Phone 1" 
;party=calling;screen=yes;privacy=off
From: "Site B Phone 1" ;tag=15CC463C-15AE
To: 
Date: Tue, 06 Oct 2015 02:30:47 GMT
Call-ID: 164EFFA5-6B0911E5-84D6F2F4-6AC111CC@142.102.65.254
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1057581824-065536-19-0188834958
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1444098647
Contact: 
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Length: 0


Oct  6 02:30:47.413: //11/3F096B00/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 142.102.65.254:5060;branch=z9hG4bKBDD75
From: "Site B Phone 1" ;tag=15CC463C-15AE
To: 
Date: Tue, 06 Oct 2015 02:30:47 GMT
Call-ID: 164EFFA5-6B0911E5-84D6F2F4-6AC111CC@142.102.65.254
Timestamp: 1444098647
CSeq: 101 INVITE

Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Oct  6 02:30:47.413: //11/3F096B00/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 142.102.65.254:5060;branch=z9hG4bKBDD75
From: "Site B Phone 1" ;tag=15CC463C-15AE
To: ;tag=11609C-F9E
Call-ID: 164EFFA5-6B0911E5-84D6F2F4-6AC111CC@142.102.65.254
CSeq: 101 INVITE
Timestamp: 1444098647
Content-Length: 0


Oct  6 02:30:47.441: //-1//SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:4001@142.102.66.254:5060 SIP/2.0
Via: SIP/2.0/UDP 142.102.65.254:5060;branch=z9hG4bKBDD75
From: "Site B Phone 1" ;tag=15CC463C-15AE
To: ;tag=11609C-F9E
Date: Tue, 06 Oct 2015 02:30:47 GMT
Call-ID: 164EFFA5-6B0911E5-84D6F2F4-6AC111CC@142.102.65.254
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
Syslog logging: enabled (0 messages dropped, 8 messages rate-limited, 0 
flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.



No Inactive Message Discriminator.


Console logging: disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
 filtering disabled
Buffer logging:  level debugging, 421 messages logged, xml disabled,
filtering disabled
Exception Logging: size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

Trap logging: level informational, 106 message lines logged
Logging Source-Interface:   VRF Name:

Log Buffer (1000 bytes):

Oct  6 02:32:24.988: //-1/78DA7180/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=3001
   - ccCallInfo IE subfields -
   cisco-ani=3001
   cisco-anitype=0
   cisco-aniplan=0
   

Re: [cisco-voip] Odd call behavior CME,CUBE,CUCM

[Ryan Huff]

HA! Thanks Brian ... orphaned passthru  this was the 28th (seriously) time 
I reviewed the config ... finally spotted it.

Is this because CUBE is presenting DTMF as part of the INVITE?

Thanks,

Ryan

Date: Mon, 5 Oct 2015 23:14:07 -0400
Subject: Re: [cisco-voip] Odd call behavior CME,CUBE,CUCM
From: bmead...@vt.edu
To: ryanh...@outlook.com
CC: cisco-voip@puck.nether.net

Make sure you don't have any of these on the dial-peer:voice-class sip 
pass-thru headers unsuppvoice-class sip pass-thru content unsuppvoice-class sip 
pass-thru content sdp

http://www.dslreports.com/forum/r27144625-Config-CME-8-Help-with-Incoming-calls-on-SIP

On Mon, Oct 5, 2015 at 11:00 PM, Ryan Huff <ryanh...@outlook.com> wrote:



I have CME that has a SIP phone registered to it and a SCCP phone registered to 
it.
SIP=4002
SCCP=4001
 
I have a call path that looks like:
 
CUCM->SIPTrunk->CUBE->SIPTrunk->CME
 
When I dial the SIP phone (on CME) from call manager the call completes and 
goes fine. When I dial the skinny phone (on CME) from call manager the call 
immediately fails. However, from CME, using the skinny phone, I can dial 
anything in call manager and the call completes.
 
CCSIP on CME (for one of the failed calls) shows the INVITE then TRYING then 
SIP/2.0 500 Internal Server Error. CCAPI INOUT shows a cause code of 16, but 
this was anything but a normal clearing. Just for the hell of it, I also added 
a transcoder in CME and registered it in telephony-service, just to see if 
there was a codec flying around that I wasn't seeing  but alas, no.
 
I've attached the ccsip and ccapi inout; I think I have just been starring at 
it too long ... I can't see it.

Thanks,

Ryan
 
  

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Re: [cisco-voip] ldaps authentication

[Ryan Huff]

Ed,

You may need to make sure that the common name of the ldap server also appears 
in the subject alternative name (SAN) of the certificate.

ldap dirsync will come from the cucm publisher node but ldap auth could 
potentially come from any cucm node.

My other thoughts besides a cert issue would be (since you reference an error 
about hostname mismatches) is; are there any stray/orphaned a records in dns, 
forward and reverse zones setup correctly, ALL cucm nodes are using dns servers 
that resolve the same forward and reverse zones. Also verify that every cucm 
node can talk to the directory server (in the case that nodes are in different 
network segments).

If you still have the CSR from the ldap server, take it to a CSR decoder 
(Google shows plenty) and I woul be interested to know if the cn is/is not in 
the san (or decode the ident cert on the server).

Thanks,

Ryan

Sent from my iPad

> On Oct 5, 2015, at 8:22 AM, Ed Leatherman  wrote:
> 
> Hello!
> 
> We turned up directory sync on cucm yesterday, and ran into some issues with 
> authentication; I ran out of maintenance window so we ended up converting the 
> small number of end users that were synced back into local accounts for now.
> 
> Our LDAP is front-ended by a load balancer that uses a wild-card certificate. 
> Yeah, I should have seen this coming.
> 
> What I have is my test cluster, running 10.5.2.1-5, integrated using 
> ldaps and working fine
> 
> My production system is slightly more recent 10.5.2.12901 (unrelated reason 
> as to why they don't match). Directory sync works fine using ldaps , but 
> authentication will not work, error message in the tomcat trace says that the 
> hostname doesn't match the certificate. I can see the wildcard cert CN's in 
> the trace.
> 
> I can't even see any entries in the test system trace file related the SSL 
> socket (nor could Tac), so i'm assuming that extra trace info was added in 
> the SU. I guess it also started enforcing the no wild-card rule on 
> certificates for other things - I was under the (apparently false) impression 
> that that rule was only related to signing CUCM certs. 
> 
> But why does my dir sync work ok, it uses SSL also to the same host? Tac 
> isn't interested in troubleshooting any further as they say it's unsupported. 
> 
> We tried changing LDAP on CUCM to use IP instead of hostname to skip the SSL 
> hostname check, this worked for authenticating Ucmuser webpage but it did not 
> work for Jabber. I wanted to troubleshoot this to see if this issue was not 
> SSL related but we ran out of maintenance window.
> 
> My action plan right now is to move my "beta" users off the test system and 
> get it to the same version as production, and try to reproduce the issue. 
> Also investigating getting a normal cert for our ldap but I'm not sure how 
> feasible this will be.
> 
> Any suggestions or am I SOL with that wildcard cert?
> 
> 
> -- 
> Ed Leatherman
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Re: [cisco-voip] ldaps authentication

[Ryan Huff]

When it comes to ssl integrations with CCM, I tend to subscribe to the idea 
that the "integrators" should play by the same safety rules that I use for CCM; 
CN in the SAN, no UPPER case letters or non alphanumeric characters ... etc
 
However, I'd be cautious to call that the smoking gun because you do have A 
node successfully negotiating with the LDAP server; my first peek would be at 
communications between all cucm nodes and the directory server. 
 
The ldap auth is using the tomcat service; I have seen a simple tomcat service 
bounce on all the ccm nodes do the trick. Especially if the cert on the 
directory server has recently changed; would be the equivalent of a "ipconfig 
/flushdns" in the windows world, although a bit more impactful ;).
 
Thanks,
 
-R
Date: Mon, 5 Oct 2015 09:03:08 -0400
Subject: Re: [cisco-voip] ldaps authentication
From: ealeather...@gmail.com
To: ryanh...@outlook.com
CC: cisco-voip@puck.nether.net

Thanks Ryan those are good suggestions.
CN is not in the cert, that much I can see from the trace 
files:impl.Certificates - getCNs : impl.LDAPHostnameVerifier - check : cns = 
[*.wvu.edu]impl.Certificates - getDNSSubjectAlts : impl.LDAPHostnameVerifier - 
check : subjectAlts = [*.wvu.edu, wvu.edu]
We're reaching out to the F5 and ldap teams here to see what impact is on 
making some changes to get rid of the wildcard - apparently they have related 
issues with other apps around this so I might get more traction on it than I 
expected.
Double checking dns on the other cm nodes is good idea - i'm pretty sure they 
are all using the DNS but close only counts in horseshoes and hand grenades as 
they say. I forgot that the secondary nodes were able to do auth on their own.

On Mon, Oct 5, 2015 at 8:41 AM, Ryan Huff <ryanh...@outlook.com> wrote:
Ed,



You may need to make sure that the common name of the ldap server also appears 
in the subject alternative name (SAN) of the certificate.



ldap dirsync will come from the cucm publisher node but ldap auth could 
potentially come from any cucm node.



My other thoughts besides a cert issue would be (since you reference an error 
about hostname mismatches) is; are there any stray/orphaned a records in dns, 
forward and reverse zones setup correctly, ALL cucm nodes are using dns servers 
that resolve the same forward and reverse zones. Also verify that every cucm 
node can talk to the directory server (in the case that nodes are in different 
network segments).



If you still have the CSR from the ldap server, take it to a CSR decoder 
(Google shows plenty) and I woul be interested to know if the cn is/is not in 
the san (or decode the ident cert on the server).



Thanks,



Ryan



Sent from my iPad



> On Oct 5, 2015, at 8:22 AM, Ed Leatherman <ealeather...@gmail.com> wrote:

>

> Hello!

>

> We turned up directory sync on cucm yesterday, and ran into some issues with 
> authentication; I ran out of maintenance window so we ended up converting the 
> small number of end users that were synced back into local accounts for now.

>

> Our LDAP is front-ended by a load balancer that uses a wild-card certificate. 
> Yeah, I should have seen this coming.

>

> What I have is my test cluster, running 10.5.2.1-5, integrated using 
> ldaps and working fine

>

> My production system is slightly more recent 10.5.2.12901 (unrelated reason 
> as to why they don't match). Directory sync works fine using ldaps , but 
> authentication will not work, error message in the tomcat trace says that the 
> hostname doesn't match the certificate. I can see the wildcard cert CN's in 
> the trace.

>

> I can't even see any entries in the test system trace file related the SSL 
> socket (nor could Tac), so i'm assuming that extra trace info was added in 
> the SU. I guess it also started enforcing the no wild-card rule on 
> certificates for other things - I was under the (apparently false) impression 
> that that rule was only related to signing CUCM certs.

>

> But why does my dir sync work ok, it uses SSL also to the same host? Tac 
> isn't interested in troubleshooting any further as they say it's unsupported.

>

> We tried changing LDAP on CUCM to use IP instead of hostname to skip the SSL 
> hostname check, this worked for authenticating Ucmuser webpage but it did not 
> work for Jabber. I wanted to troubleshoot this to see if this issue was not 
> SSL related but we ran out of maintenance window.

>

> My action plan right now is to move my "beta" users off the test system and 
> get it to the same version as production, and try to reproduce the issue. 
> Also investigating getting a normal cert for our ldap but I'm not sure how 
> feasible this will be.

>

> Any suggestions or am I SOL with that wildcard cert?

>

>

> --

> Ed Leatherman

> ___

Re: [cisco-voip] ILBC

[Ryan Huff]

I am studying for my IE in a few weeks so I am just working in the lab Had to 
use a 7941 cause it was all I had and I guess I just never tried to use ILBC on 
a 41 before. Definitelyhad me scratching my head for a few minutes.

Sent from my iPad

> On Oct 4, 2015, at 2:23 PM, James Buchanan <james.buchan...@gmail.com> wrote:
> 
> I never said anything about your psychological condition since anyone in 
> Cisco voice must be a little off-kilter, but you are most welcome.
> 
>> On Sun, Oct 4, 2015 at 2:19 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> Thanks for confirming that I wasn't going nuts.
>> 
>> Sent from my iPad
>> 
>>> On Oct 4, 2015, at 2:17 PM, James Buchanan <james.buchan...@gmail.com> 
>>> wrote:
>>> 
>>> One of the new features of the 7942/62 was the addition of iLBC support. 
>>> See 
>>> http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7962g/prod_qas0900aecd80699c20.html.
>>> 
>>>> On Sun, Oct 4, 2015 at 2:11 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>>>> Having some issues with a 7941 refusing ILBC and preferring G711ULAW 
>>>> (despite proper region/device pool config) and I think I know why, just 
>>>> looking for a confirmation here really
>>>> 
>>>> As I read it from;
>>>> 
>>>> http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7941g/product_data_sheet0900aecd802ff012.html
>>>> 
>>>> the 7941 does not support the ILBC codec.
>>>> 
>>>> The troubleshooting guide for the 7941,7942,7961,7962 at 
>>>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7962g_7961g_7961g-ge_7942g_7941g_7941g-ge/8_0/english/administration/guide/62adm80/62614241trb.html
>>>>  has troubleshooting steps in the case that a phone is not negotiating 
>>>> ILBC. I assume those troubleshooting steps are referring to the 7942 and 
>>>> 7962 and not the 41/61.
>>>> 
>>>> Thanks,
>>>> 
>>>> Ryan
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Re: [cisco-voip] ILBC

[Ryan Huff]

Thanks for confirming that I wasn't going nuts.

Sent from my iPad

> On Oct 4, 2015, at 2:17 PM, James Buchanan <james.buchan...@gmail.com> wrote:
> 
> One of the new features of the 7942/62 was the addition of iLBC support. See 
> http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7962g/prod_qas0900aecd80699c20.html.
> 
>> On Sun, Oct 4, 2015 at 2:11 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>> Having some issues with a 7941 refusing ILBC and preferring G711ULAW 
>> (despite proper region/device pool config) and I think I know why, just 
>> looking for a confirmation here really
>> 
>> As I read it from;
>> 
>> http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7941g/product_data_sheet0900aecd802ff012.html
>> 
>> the 7941 does not support the ILBC codec.
>> 
>> The troubleshooting guide for the 7941,7942,7961,7962 at 
>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7962g_7961g_7961g-ge_7942g_7941g_7941g-ge/8_0/english/administration/guide/62adm80/62614241trb.html
>>  has troubleshooting steps in the case that a phone is not negotiating ILBC. 
>> I assume those troubleshooting steps are referring to the 7942 and 7962 and 
>> not the 41/61.
>> 
>> Thanks,
>> 
>> Ryan
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[cisco-voip] ILBC

[Ryan Huff]

Having some issues with a 7941 refusing ILBC and preferring G711ULAW (despite 
proper region/device pool config) and I think I know why, just looking for a 
confirmation here really

As I read it from;

http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7941g/product_data_sheet0900aecd802ff012.html

the 7941 does not support the ILBC codec.

The troubleshooting guide for the 7941,7942,7961,7962 at 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7962g_7961g_7961g-ge_7942g_7941g_7941g-ge/8_0/english/administration/guide/62adm80/62614241trb.html
 has troubleshooting steps in the case that a phone is not negotiating ILBC. I 
assume those troubleshooting steps are referring to the 7942 and 7962 and not 
the 41/61.

Thanks,

Ryan
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Re: [cisco-voip] Presence status for third party app

[Ryan Huff]

Rob, 

Not sure what you're looking to do, but I have written an abstracted API layer 
in PHP to do things such as; text/email when contact is available/away ... Etc. 
I can share if interested.

Right now I am working on a script for when a contact goes away and the the 
away status exists for longer than 15 minutes will trigger an IO event on a 
fidget board that will in turn use relays to activate an electronic doorlock. 
The idea being as long as the contact stays active or available the door 
remains unlocked, if the contact is in away status for longer than 15 minutes 
the door locks. Sort of a proof of concept thing right now.

Sent from my iPad

> On Oct 2, 2015, at 3:35 PM, Rob Dawson  wrote:
> 
> Thanks. I actually just stumbled on the Jabber web SDK on the Dev site, which 
> will probably do what I need –
>  
> https://developer.cisco.com/site/jabber-websdk/overview/
>  
> The app is non-existent as of yet, just pondering some things – triggering 
> real world events based on status, that type of stuff.
>  
> Thanks again,
> Rob
>  
> From: avhollo...@gmail.com [mailto:avhollo...@gmail.com] On Behalf Of Anthony 
> Holloway
> Sent: Friday, October 02, 2015 3:23 PM
> To: Rob Dawson
> Cc: voip puck
> Subject: Re: [cisco-voip] Presence status for third party app
>  
> Have you looked into this:
>  
> https://developer.cisco.com/site/unified-presence/overview/
>  
> Or are you asking for a ready made solution?  What's the third party app?
>  
> On Fri, Oct 2, 2015 at 2:16 PM, Rob Dawson  wrote:
> Doing some tinkering and looking for a way to monitor a particular users 
> status from a third party app. I assume I could just do it via XMPP, but is 
> there an preferred method for doing this?
>  
> Thanks,
> Rob
> 
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> 
>  
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Re: [cisco-voip] 3650 PoE Issue

[Ryan Huff]

You may also try increasing the poe advertisement length on the port ( power 
inline delay shutdown 20 initial 300 ). 


Sent from my T-Mobile 4G LTE Device

 Original message 
From: "Casper, Steven" <scas...@mtb.com> 
Date:09/30/2015  5:18 PM  (GMT-05:00) 
To: 'Ryan Huff' <ryanh...@outlook.com>,cisco-voip voip list 
<cisco-voip@puck.nether.net> 
Subject: RE: [cisco-voip] 3650 PoE Issue 

Module   Available Used Remaining
 (Watts) (Watts)    (Watts)
--   -      -
1  1550.0 141.0  1409.0
2  1550.0    96.9  1453.1
3  1550.0    165.5   1384.5
4  1550.0   103.6    1446.4
5  1550.0  98.9  1451.1
 
Going to try power inline auto max 15400 on the ports supporting the 7940s to 
see if it makes a difference when the stack reloads.
 
Steve
 
From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: Wednesday, September 30, 2015 10:18 AM
To: Casper, Steven; cisco-voip voip list
Subject: RE: [cisco-voip] 3650 PoE Issue
 
How much available power do you have left on the POE backplane once all phones 
are up and working?

 
From: scas...@mtb.com
To: cisco-voip@puck.nether.net
Date: Wed, 30 Sep 2015 12:33:51 +
Subject: [cisco-voip] 3650 PoE Issue

Having an issue related to PoE negotiation between 3650 switches that we are 
beginning to deploy  and older cisco phones such as 7960s and 7940s . I think 
these are using the pre PoE standard. Symptoms are when the stack reloads  the  
switchports connected to Cisco 7960 telephones come up  in a hung state:
 
7960 Telephones showed no power
No link lights on switchport
Switchport status showed UP/UP
Switchport remained in UP/UP state even after disconnecting the cable from 
switchport
Shut/no shut port and the telephone would reboot normally
Moving to available switchport and telephone rebooted normally
Bypassed telephone and connected PC and the PC would reboot normally 
reconnected telephone and the telephone would reboot normally.
 
Newer phones such as 7942 come right up.
 
Once we performed a shut/no shut on all switch ports and all telephones 
recovered normally but still this is not normal behavior. Going to open a TAC 
case but wondering if anyone else has experienced  this?
Switches are WS-C3650-48PD on Version 03.03.05SE.
 
Thanks!
Steve
 
 
 
 

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Re: [cisco-voip] 3650 PoE Issue

[Ryan Huff]

How much available power do you have left on the POE backplane once all phones 
are up and working?

 
From: scas...@mtb.com
To: cisco-voip@puck.nether.net
Date: Wed, 30 Sep 2015 12:33:51 +
Subject: [cisco-voip] 3650 PoE Issue









Having an issue related to PoE negotiation between 3650 switches that we are 
beginning to deploy  and older cisco phones such as 7960s and 7940s . I think 
these are using the pre PoE standard. Symptoms are when the stack reloads  the  
switchports
 connected to Cisco 7960 telephones come up  in a hung state:
 
7960 Telephones showed no power
No link lights on switchport
Switchport status showed UP/UP 
Switchport remained in UP/UP state even after disconnecting the cable from 
switchport
Shut/no shut port and the telephone would reboot normally
Moving to available switchport and telephone rebooted normally
Bypassed telephone and connected PC and the PC would reboot normally 
reconnected telephone and the telephone would reboot normally.
 
Newer phones such as 7942 come right up.
 
Once we performed a shut/no shut on all switch ports and all telephones 
recovered normally but still this is not normal behavior. Going to open a TAC 
case but wondering if anyone else has experienced  this?
Switches are WS-C3650-48PD on Version 03.03.05SE.
 
Thanks!
Steve
 
 
 
 



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This communication may contain nonpublic personal information about consumers 
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Re: [cisco-voip] 10.5 DRS Restore problem?

[Ryan Huff]

Usually means you are attempting a restore from an incomplete backup set. Is 
the backup set you are attempting restore with, from a successfully completed 
DRS backup of a like ucos version?

Also, check the file permissions on your backup set and make sure the sftp user 
can read them.

Thanks,

Ryan
Sent from space

> On Sep 25, 2015, at 11:43 AM, Jason Aarons (AM) 
>  wrote:
> 
> Anyone hit this problem with a DRS restore in CUCM 10.5?  Open the file on 
> the SFTP server we can see them present.
>  
> 2015-09-24 16:49:37,927 ERROR [drfMasterAgentMsgWorker, MessageID: 2400] - 
> drfSftpManager:drfGetListOfBackups: No files with pattern _ drfComponent.xml 
> on the path /
>  
>  
>  
> Jason Aarons, CCIEx2 No 38564
> Consultant
> Dimension Data
> 904-338-3245 mobile
>  
>  
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Re: [cisco-voip] LDAP error on CUC

[Ryan Huff]

I'm assuming that you have performed a directory synchronization in Unity 
Connections first, then attempted the import (and the local Unity Connections 
account name matches the AD account name).
 
Thanks,
 
Ryan
 
Date: Thu, 24 Sep 2015 10:45:25 -0500
From: jonv...@gmail.com
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] LDAP error on CUC

Please note that this is for ALL users... 
On Thu, Sep 24, 2015 at 10:44 AM, Jonathan Charles  wrote:
We are integrated CUC/CUCM with AD (version 10.5)...
We can import users from AD (Users - Import Users - from LDAP) successfully.
The same user cannot be converted from local to AD (checking box Integrate with 
LDAP Directory on the CUC User page)...
The error returned is "This user cannot be associated with the LDAP Directory."
Google returns no results...
Any ideas?



Jonathan



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Re: [cisco-voip] LDAP error on CUC

[Ryan Huff]

To convert local accounts to AD synced you should perform the directory sync, 
then use the checkbox in the account to integrate with ldap (or use the bat 
tool). After you import an account from AD, the account is no longer in Unity's 
cached ad account list for Unity to match the local account with.

Thanks,

Ryan

Sent from my T-Mobile 4G LTE Device

 Original message 
From: Ryan Huff <ryanh...@outlook.com> 
Date:09/24/2015  12:01 PM  (GMT-05:00) 
To: Jonathan Charles <jonv...@gmail.com>,cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] LDAP error on CUC 

I'm assuming that you have performed a directory synchronization in Unity 
Connections first, then attempted the import (and the local Unity Connections 
account name matches the AD account name).
 
Thanks,
 
Ryan
 
Date: Thu, 24 Sep 2015 10:45:25 -0500
From: jonv...@gmail.com
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] LDAP error on CUC

Please note that this is for ALL users... 

On Thu, Sep 24, 2015 at 10:44 AM, Jonathan Charles <jonv...@gmail.com> wrote:
We are integrated CUC/CUCM with AD (version 10.5)...

We can import users from AD (Users - Import Users - from LDAP) successfully.

The same user cannot be converted from local to AD (checking box Integrate with 
LDAP Directory on the CUC User page)...

The error returned is "This user cannot be associated with the LDAP Directory."

Google returns no results...

Any ideas?




Jonathan


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Re: [cisco-voip] Feature Request to fix Licensing issues with EM

[Ryan Huff]

Well now  look at that shiny toy. 

I came up with my Excel spreadsheet back when 9.1 came out as I had 10k of 
phones to deal with at the time, simple and it just worked so I stuck with it. 
Nice to see there is an official tool for it though.

Good stuff Anthony, thanks!

-Ryan

Date: Thu, 17 Sep 2015 14:34:54 -0500
Subject: Re: [cisco-voip] Feature Request to fix Licensing issues with EM
From: avholloway+cisco-v...@gmail.com
To: ryanh...@outlook.com
CC: svoll.v...@gmail.com; le...@uoguelph.ca; cisco-voip@puck.nether.net

Ryan,
It sounds like you haven't seen/used the DAT?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/udat/userguide/UDAT_BK_U1401523_00_device-assignment-tool-user-guide.html

On Thu, Sep 17, 2015 at 2:30 PM, Ryan Huff <ryanh...@outlook.com> wrote:



Scott,

The DLU <-> CUWL license model change was and continues to be a significant 
change for customers; no arguments there. 

However, once you make it to the other side, the CUWL licensing model makes a 
lot more sense and is ultimately easier to manage and predict usage. Having 
said that; there are techniques you can use to make the 'true-up' (associating 
devices to users) process a little less painful. 

For instance, you can use BAT to export a user list and device list. By using 
Excel pivot tables and a little join magic you can usually come up with a 
device re-import that will slap user associations on the devices (beats 
clicking through 1,000+ devices in GUI). If you have IM & Presence, you could 
do something similar for line ownership (needed for line presence).


Thanks,

Ryan

Date: Thu, 17 Sep 2015 11:55:54 -0700
From: svoll.v...@gmail.com
To: avholloway+cisco-v...@gmail.com; le...@uoguelph.ca
Subject: Re: [cisco-voip] Feature Request to fix Licensing issues with EM
CC: cisco-voip@puck.nether.net

We are 100% EM in CM 8.6.  Now I have to associate phones to Users so I can get 
the licensing that I have support on.  1000 users rather than 1600 phones. (450 
are soft clients).  Since I'm licensed for 1000 I have to "true up" my 
licensing ($24k).  so if convert over to CUWL for 1000 why should I buy more 
licensing that I already have? thus working through the association of devices 
to users.
So now I'm stuck in a Associating user / device nightmare.  which I can work 
through.  But now I have to keep my associations current moving forward (we 
move users often) so I don't get out of Licensing complainants again.
Scott
On Thu, Sep 17, 2015 at 11:44 AM, Anthony Holloway 
<avholloway+cisco-v...@gmail.com> wrote:
Can you briefly explain this EM licensing nightmare?  I'm not sure I'm aware of 
it.
Also, it sounds like what you are describing, implies that license compliance 
checks are run frequently and multiple times throughout the day, and to my 
knowledge they are not.  How else would you keep up with all of the logging in 
and out of devices, unless the license compliance check was run in real time?  
Which was probably how it worked prior to ELM.  Just spitballing a few ideas.
On Thu, Sep 17, 2015 at 1:29 PM Scott Voll <svoll.v...@gmail.com> wrote:
OK, everyone has complained about the licensing nightmare that EM plays in CM 9 
and 10.
My coworker has come up with a fix..

In thinking this all through, I thought about how I would
like it all to work mechanically.  Here
is what I would like to request as a Feature/Enhancement Request:


 


When I log into a device for the first time with Extension
Mobility, I am assigned as the device owner. 
That ownership is retained until one of two things happen:


1.I log into another like device (i.e.: I have a
8861 at my desk, and I log into Scott’s 8861 at his desk) 


2.Someone else logged into my device.


 


This way, the ownership of the device is always assigned by
the extension mobility profile and follows the user.  If I have an iPhone, IPad 
or Jabber or other
such device, since those all require a user to log in, the ownership would be
added in the same way.  This would
mitigate the licensing issues caused by extension mobility (in my opinion). 


 


Using this would really simplify the licensing requirements
in my thinking.


 


Any thoughts?


Thanks


Scott



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Re: [cisco-voip] Feature Request to fix Licensing issues with EM

[Ryan Huff]

Scott,

The DLU <-> CUWL license model change was and continues to be a significant 
change for customers; no arguments there. 

However, once you make it to the other side, the CUWL licensing model makes a 
lot more sense and is ultimately easier to manage and predict usage. Having 
said that; there are techniques you can use to make the 'true-up' (associating 
devices to users) process a little less painful. 

For instance, you can use BAT to export a user list and device list. By using 
Excel pivot tables and a little join magic you can usually come up with a 
device re-import that will slap user associations on the devices (beats 
clicking through 1,000+ devices in GUI). If you have IM & Presence, you could 
do something similar for line ownership (needed for line presence).


Thanks,

Ryan

Date: Thu, 17 Sep 2015 11:55:54 -0700
From: svoll.v...@gmail.com
To: avholloway+cisco-v...@gmail.com; le...@uoguelph.ca
Subject: Re: [cisco-voip] Feature Request to fix Licensing issues with EM
CC: cisco-voip@puck.nether.net

We are 100% EM in CM 8.6.  Now I have to associate phones to Users so I can get 
the licensing that I have support on.  1000 users rather than 1600 phones. (450 
are soft clients).  Since I'm licensed for 1000 I have to "true up" my 
licensing ($24k).  so if convert over to CUWL for 1000 why should I buy more 
licensing that I already have? thus working through the association of devices 
to users.
So now I'm stuck in a Associating user / device nightmare.  which I can work 
through.  But now I have to keep my associations current moving forward (we 
move users often) so I don't get out of Licensing complainants again.
Scott
On Thu, Sep 17, 2015 at 11:44 AM, Anthony Holloway 
 wrote:
Can you briefly explain this EM licensing nightmare?  I'm not sure I'm aware of 
it.
Also, it sounds like what you are describing, implies that license compliance 
checks are run frequently and multiple times throughout the day, and to my 
knowledge they are not.  How else would you keep up with all of the logging in 
and out of devices, unless the license compliance check was run in real time?  
Which was probably how it worked prior to ELM.  Just spitballing a few ideas.
On Thu, Sep 17, 2015 at 1:29 PM Scott Voll  wrote:
OK, everyone has complained about the licensing nightmare that EM plays in CM 9 
and 10.
My coworker has come up with a fix..

In thinking this all through, I thought about how I would
like it all to work mechanically.  Here
is what I would like to request as a Feature/Enhancement Request:

 

When I log into a device for the first time with Extension
Mobility, I am assigned as the device owner. 
That ownership is retained until one of two things happen:

1.I log into another like device (i.e.: I have a
8861 at my desk, and I log into Scott’s 8861 at his desk) 

2.Someone else logged into my device.

 

This way, the ownership of the device is always assigned by
the extension mobility profile and follows the user.  If I have an iPhone, IPad 
or Jabber or other
such device, since those all require a user to log in, the ownership would be
added in the same way.  This would
mitigate the licensing issues caused by extension mobility (in my opinion). 

 

Using this would really simplify the licensing requirements
in my thinking.

 

Any thoughts?
Thanks
Scott

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Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

My last thought here would be that if you've recently had to reboot the cluster 
(due to replication/database issues) and now you had this strange anomaly where 
CTI control got out of step randomly; could be symptoms of other, less 
transparent issues.

I would do a heath check on my UC enterprise

Completely vet forward and reverse DNS lookups for my entire UC domainEnsure 
all publishers are synced with a reliable NTP clockCisco recommends Cisco IOS 
or Linux based NTP services for Unified Communications at Stratum-1, Stratum-2 
or Stratum-3Verify my IP phone networks have access to DNS that can do forward 
and reverse lookups on the entire UC domainVerify there isn't any unusually 
high latency between UC servers ... etc

Thanks,


RYan


Date: Wed, 16 Sep 2015 13:38:07 +0100
From: abba...@gmail.com
To: christine.see.ev...@chemeketa.edu; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX 9 EM agents cant login

thanks Christine/Ryan,

Cluster rebooted last time due to some db sync issues but its been fine since 
then. talking about 3/4 weeks back that was.

could have restarted the RIS Data Collector. but at this time creating and 
recreating new profiles will do.

thanks all.


On 15 September 2015 at 23:17, Christine See-Evans 
<christine.see.ev...@chemeketa.edu> wrote:
Delete the EM Profile then rebuild the profile, do the associations in both 
CUCM and UCCX, CTI re-start. 

Check the version for your agent/supervisor CAD desktop for your version of 
CUCM/UCCX (if you have them). Uninstall, re-install, restart.

That's my last ditch effort.



Christine See-Evans, BCS, MBA
Network Analyst
Chemeketa Community College
4000 Lancaster Drive NE,
Salem, OR 97305
christine.see.ev...@chemeketa.edu
(503)589-7776



 
"Make space in your life for the things that matter, for family and friends, 
love and generosity, fun and joy. Without this, you will burn out in mid-career 
and wonder where your life went."― Jonathan Sacks 


On Tue, Sep 15, 2015 at 11:23 AM, abbas wali <abba...@gmail.com> wrote:
Cti restarts and profile+ device re association been done already. No luck.  
From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 18:46
To: ealeather...@gmail.com; abba...@gmail.com
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
 If the cti service in ccm has lost the state of the device, reassociating the 
device as Ed suggests, or restarting the cti manager service in ccm is how I 
have resolved these types of issues before.
Thanks,
Ryan


 Original Message 
From: Ed Leatherman <ealeather...@gmail.com>
Sent: Tuesday, September 15, 2015 01:36 PM
To: abbas wali <abba...@gmail.com>
Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
CC: Ryan Huff <ryanh...@outlook.com>,Cisco VOIP 
<cisco-voip@puck.nether.net>I've had some weird, rare occasions where i've had 
to disassociate the device or profile from the rmjtapi app user and 
re-associate them. I'd suggest you try that if you haven't already just as a 
quick thing to do, although that won't tell you a root cause. On Tue, Sep 15, 
2015 at 10:21 AM, abbas wali <abba...@gmail.com> wrote:Sorry that will mean !!  
From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 15:18
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Have is the subsystem in 
partial service?From: abba...@gmail.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Date: Tue, 15 Sep 2015 15:11:33 +0100Thanks Ryan,  We have a dozen of other 
users who can login to them phones without any issues. Even I can do it on my 
CIPC  But these new 3 agents cant.  From: Ryan Huff 
[mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 15:05
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a 
new/upgrade UCCX and it isn't a JTAPI compatibility issue with CCM; assuming it 
was working and now it is not.

I would check these items;

Agent/phone checks;Make sure the RmCM user/password hasn't changed from what 
UCCX has recorded. NODoes the agent have the Standard CTI Role? YesDoes the 
agent have IPCCX defined in their profile? YesDoes the agent have CTI control 
of the phone? YesDoes the agent have control of the EM. profile? YesIs the 
physical phone associated to the RMCM user? Yes ( currently to log them in I am 
using my soft phone which is associated with RMCM user )

Server checks; yes as many other agent can login and are talking calls. DNS ... 
(are forward and reverse lookups working correctly) NTP ... NTP still working?

Thanks,


RyanFrom: abba...@gmail.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Date: Tue, 15 Sep 2015 14:53:54 +0100That is the case, the DN is exclus

Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

It likely indicates 'something' was out of sync. The act of creating a new 
account (assuming the parameters were the same) re inserted into the db with 
new associations.

Strange, what version of ccm? When was the last time that cluster was rebooted?

Thanks,

Ryan

 Original Message 
From: abbas wali <abba...@gmail.com>
Sent: Tuesday, September 15, 2015 05:54 PM
To: 'Ryan Huff' <ryanh...@outlook.com>,ealeather...@gmail.com
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
CC: cisco-voip@puck.nether.net

>
>
>This doesn’t make any sense .. 
>
> 
>
>Created a new account with the same parameters for that agent/user and it does 
>login. 
>
> 
>
>That’s absurd 
>
> 
>
> 
>
>From: abbas wali [mailto:abba...@gmail.com] 
>Sent: 15 September 2015 19:23
>To: 'Ryan Huff' <ryanh...@outlook.com>; ealeather...@gmail.com
>Cc: cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Cti restarts and profile+ device re association been done already. No luck. 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 18:46
>To: ealeather...@gmail.com; abba...@gmail.com
>Cc: cisco-voip@puck.nether.net
>Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>If the cti service in ccm has lost the state of the device, reassociating the 
>device as Ed suggests, or restarting the cti manager service in ccm is how I 
>have resolved these types of issues before.
>
>Thanks,
>
>Ryan
>
>
>
> Original Message 
>From: Ed Leatherman <ealeather...@gmail.com>
>Sent: Tuesday, September 15, 2015 01:36 PM
>To: abbas wali <abba...@gmail.com>
>Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
>CC: Ryan Huff <ryanh...@outlook.com>,Cisco VOIP <cisco-voip@puck.nether.net>
>
>I've had some weird, rare occasions where i've had to disassociate the device 
>or profile from the rmjtapi app user and re-associate them. I'd suggest you 
>try that if you haven't already just as a quick thing to do, although that 
>won't tell you a root cause.
>
> 
>
>On Tue, Sep 15, 2015 at 10:21 AM, abbas wali <abba...@gmail.com> wrote:
>
>Sorry that will mean !! 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 15:18
>
>
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Have is the subsystem in partial service?
>
>From: abba...@gmail.com
>To: ryanh...@outlook.com; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>Date: Tue, 15 Sep 2015 15:11:33 +0100
>
>Thanks Ryan, 
>
> 
>
>We have a dozen of other users who can login to them phones without any 
>issues. Even I can do it on my CIPC 
>
> 
>
>But these new 3 agents cant. 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 15:05
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility 
>issue with CCM; assuming it was working and now it is not.
>
>I would check these items;
>
>Agent/phone checks;
>
>Make sure the RmCM user/password hasn't changed from what UCCX has recorded. NO
>
>Does the agent have the Standard CTI Role? Yes
>
>Does the agent have IPCCX defined in their profile? Yes
>
>Does the agent have CTI control of the phone? Yes
>
>Does the agent have control of the EM. profile? Yes
>
>Is the physical phone associated to the RMCM user? Yes ( currently to log them 
>in I am using my soft phone which is associated with RMCM user )
>
>
>
>Server checks; yes as many other agent can login and are talking calls. 
>
>DNS ... (are forward and reverse lookups working correctly) 
>
>NTP ... NTP still working?
>
>
>
>Thanks,
>
>
>Ryan
>
>From: abba...@gmail.com
>To: ryanh...@outlook.com; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>Date: Tue, 15 Sep 2015 14:53:54 +0100
>
>That is the case, the DN is exclusive only to the profile – its not used on 
>any phy. Phone. 
>
> 
>
>does anyone know, if  want to get traces from RTMT which option should I use 
>i.e. Cisco Call Manager will suffice ?
>
> 
>
>thanks 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 14:35
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Since you

Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

If the cti service in ccm has lost the state of the device, reassociating the 
device as Ed suggests, or restarting the cti manager service in ccm is how I 
have resolved these types of issues before.

Thanks,

Ryan

 Original Message 
From: Ed Leatherman <ealeather...@gmail.com>
Sent: Tuesday, September 15, 2015 01:36 PM
To: abbas wali <abba...@gmail.com>
Subject: Re: [cisco-voip] UCCX 9 EM agents cant login
CC: Ryan Huff <ryanh...@outlook.com>,Cisco VOIP <cisco-voip@puck.nether.net>

>I've had some weird, rare occasions where i've had to disassociate the device 
>or profile from the rmjtapi app user and re-associate them. I'd suggest you 
>try that if you haven't already just as a quick thing to do, although that 
>won't tell you a root cause.
>
>
>On Tue, Sep 15, 2015 at 10:21 AM, abbas wali <abba...@gmail.com> wrote:
>
>Sorry that will mean !! 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 15:18
>
>
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Have is the subsystem in partial service?
>
>From: abba...@gmail.com
>To: ryanh...@outlook.com; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>Date: Tue, 15 Sep 2015 15:11:33 +0100
>
>Thanks Ryan, 
>
> 
>
>We have a dozen of other users who can login to them phones without any 
>issues. Even I can do it on my CIPC 
>
> 
>
>But these new 3 agents cant. 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 15:05
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility 
>issue with CCM; assuming it was working and now it is not.
>
>I would check these items;
>
>Agent/phone checks;
>
>Make sure the RmCM user/password hasn't changed from what UCCX has recorded. 
>NODoes the agent have the Standard CTI Role? YesDoes the agent have IPCCX 
>defined in their profile? YesDoes the agent have CTI control of the phone? 
>YesDoes the agent have control of the EM. profile? YesIs the physical phone 
>associated to the RMCM user? Yes ( currently to log them in I am using my soft 
>phone which is associated with RMCM user )
>
>
>
>Server checks; yes as many other agent can login and are talking calls. 
>
>DNS ... (are forward and reverse lookups working correctly) NTP ... NTP still 
>working?
>
>
>
>Thanks,
>
>
>Ryan
>
>From: abba...@gmail.com
>To: ryanh...@outlook.com; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>Date: Tue, 15 Sep 2015 14:53:54 +0100
>
>That is the case, the DN is exclusive only to the profile – its not used on 
>any phy. Phone. 
>
> 
>
>does anyone know, if  want to get traces from RTMT which option should I use 
>i.e. Cisco Call Manager will suffice ?
>
> 
>
>thanks 
>
> 
>
>From: Ryan Huff [mailto:ryanh...@outlook.com] 
>Sent: 15 September 2015 14:35
>To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
>Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
>
> 
>
>Since you mention using extension mobility 
>
>When the Agent logs in with their Ex. mobility profile, does the DN happen to 
>be on another IP phone? The only way to "share" an ACD extension between 
>multiple devices is to assign it to a Device Profile exclusively and then 
>login using Extension Mobility to whatever device they wish to use.
>
>
>Thanks,
>
>Ryan
>
>From: abba...@gmail.com
>To: cisco-voip@puck.nether.net
>Date: Tue, 15 Sep 2015 13:54:13 +0100
>Subject: [cisco-voip] UCCX 9 EM agents cant login
>
>Hi all, 
>
> 
>
>Urgent issue here.
>
> 
>
>Ext Mob. Enabled agents cant login. Getting “Login failed due to a  
>configuration error with your phone and JTAPI or UCM. Contact your admin..”
>
> 
>
>The users profile is in the controlled list for RM application user. 
>
> 
>
>The phone they are loggin in – is used by other agents with their profiles and 
>they can get through. 
>
> 
>
>Not sure what else I can check. 
>
> 
>
>Please help !!
>
> 
>
>Thanks 
>
>
>___ cisco-voip mailing list 
>cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>___
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>
>
>
>
>-- 
>
>Ed Leatherman
>
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Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

Have is the subsystem in partial service?

From: abba...@gmail.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Date: Tue, 15 Sep 2015 15:11:33 +0100

Thanks Ryan,  We have a dozen of other users who can login to them phones 
without any issues. Even I can do it on my CIPC  But these new 3 agents cant.  
From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 15:05
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a 
new/upgrade UCCX and it isn't a JTAPI compatibility issue with CCM; assuming it 
was working and now it is not.

I would check these items;

Agent/phone checks;Make sure the RmCM user/password hasn't changed from what 
UCCX has recorded. NODoes the agent have the Standard CTI Role? YesDoes the 
agent have IPCCX defined in their profile? YesDoes the agent have CTI control 
of the phone? YesDoes the agent have control of the EM. profile? YesIs the 
physical phone associated to the RMCM user? Yes ( currently to log them in I am 
using my soft phone which is associated with RMCM user )

Server checks; yes as many other agent can login and are talking calls. DNS ... 
(are forward and reverse lookups working correctly) NTP ... NTP still working?

Thanks,


Ryan

From: abba...@gmail.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Date: Tue, 15 Sep 2015 14:53:54 +0100That is the case, the DN is exclusive only 
to the profile – its not used on any phy. Phone.  does anyone know, if  want to 
get traces from RTMT which option should I use i.e. Cisco Call Manager will 
suffice ? thanks  From: Ryan Huff [mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 14:35
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Since you mention using 
extension mobility 

When the Agent logs in with their Ex. mobility profile, does the DN happen to 
be on another IP phone? The only way to "share" an ACD extension between 
multiple devices is to assign it to a Device Profile exclusively and then login 
using Extension Mobility to whatever device they wish to use.


Thanks,

RyanFrom: abba...@gmail.com
To: cisco-voip@puck.nether.net
Date: Tue, 15 Sep 2015 13:54:13 +0100
Subject: [cisco-voip] UCCX 9 EM agents cant loginHi all,  Urgent issue here. 
Ext Mob. Enabled agents cant login. Getting “Login failed due to a  
configuration error with your phone and JTAPI or UCM. Contact your admin..” The 
users profile is in the controlled list for RM application user.  The phone 
they are loggin in – is used by other agents with their profiles and they can 
get through.  Not sure what else I can check.  Please help !! Thanks 
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Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility issue 
with CCM; assuming it was working and now it is not.

I would check these items;

Agent/phone checks;

Make sure the RmCM user/password hasn't changed from what UCCX has recorded.
Does the agent have the Standard CTI Role?Does the agent have IPCCX defined in 
their profile?Does the agent have CTI control of the phone?Does the agent have 
control of the EM. profile?Is the physical phone associated to the RMCM user?


Server checks;

DNS ... (are forward and reverse lookups working correctly)NTP ... NTP still 
working?

Thanks,


Ryan


From: abba...@gmail.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login
Date: Tue, 15 Sep 2015 14:53:54 +0100

That is the case, the DN is exclusive only to the profile – its not used on any 
phy. Phone.  does anyone know, if  want to get traces from RTMT which option 
should I use i.e. Cisco Call Manager will suffice ? thanks  From: Ryan Huff 
[mailto:ryanh...@outlook.com] 
Sent: 15 September 2015 14:35
To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Since you mention using 
extension mobility 

When the Agent logs in with their Ex. mobility profile, does the DN happen to 
be on another IP phone? The only way to "share" an ACD extension between 
multiple devices is to assign it to a Device Profile exclusively and then login 
using Extension Mobility to whatever device they wish to use.


Thanks,

RyanFrom: abba...@gmail.com
To: cisco-voip@puck.nether.net
Date: Tue, 15 Sep 2015 13:54:13 +0100
Subject: [cisco-voip] UCCX 9 EM agents cant loginHi all,  Urgent issue here. 
Ext Mob. Enabled agents cant login. Getting “Login failed due to a  
configuration error with your phone and JTAPI or UCM. Contact your admin..” The 
users profile is in the controlled list for RM application user.  The phone 
they are loggin in – is used by other agents with their profiles and they can 
get through.  Not sure what else I can check.  Please help !! Thanks 
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Re: [cisco-voip] UCCX 9 EM agents cant login

[Ryan Huff]

Since you mention using extension mobility 

When the Agent logs in with their Ex. mobility profile, does the DN happen to 
be on another IP phone? The only way to "share" an ACD extension between 
multiple devices is to 
assign it to a Device Profile exclusively and then login using Extension
 Mobility to whatever device they wish to use.


Thanks,

Ryan

From: abba...@gmail.com
To: cisco-voip@puck.nether.net
Date: Tue, 15 Sep 2015 13:54:13 +0100
Subject: [cisco-voip] UCCX 9 EM agents cant login

Hi all,  Urgent issue here. Ext Mob. Enabled agents cant login. Getting “Login 
failed due to a  configuration error with your phone and JTAPI or UCM. Contact 
your admin..” The users profile is in the controlled list for RM application 
user.  The phone they are loggin in – is used by other agents with their 
profiles and they can get through.  Not sure what else I can check.  Please 
help !! Thanks 
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Re: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection

[Ryan Huff]

I'll hav to sift through my logs and see if that is what my issue was. Thanks 
for the follow through Brian.

Thanks,

Ryan

Date: Tue, 15 Sep 2015 10:40:24 -0400
Subject: Re: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection
From: bmead...@vt.edu
To: kev...@advancedtsg.com
CC: ryanh...@outlook.com; cisco-voip@puck.nether.net

We're actually on 8.6.1.
I dug through the logs a bit more and found the same user also had an 8800 
series phone logged in via MRA.  Doing some further searching, I found someone 
who had the same issue logging into Jabber with an 8841 already logged in via 
MRA.
I had the user unplug their 8841 and they were able to login to Jabber fine 
after this.
It looks like I'll be reaching out to the feature preview folks to make sure 
they know about this issue.
Brian
On Tue, Sep 15, 2015 at 8:20 AM, Kevin Przybylowski <kev...@advancedtsg.com> 
wrote:








I almost upgraded our VCS servers to 8.6 last week and noticed a couple reviews 
on CCO so I stuck with 8.5.3.  I’ll give 8.6.1 a try in a few days.
 

 
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff

Sent: Monday, September 14, 2015 4:00 PM

To: bmead...@vt.edu; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection
 
Brian  I had this issue this weekend in 8.6.  My original issue was the "no 
home uds cluster" but I had issues with the proxy protocol violation.

Tac's response was go to 8.6.1 (released 9/11/15 ... yikes) or roll back to 8.5

Thanks,

Ryan





 Original Message 

From: Brian Meade <bmead...@vt.edu>

Sent: Monday, September 14, 2015 03:49 PM

To: cisco-voip@puck.nether.net

Subject: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection

Is anyone else having issues with the "HTTP proxy protocol violation" automated 
detection feature or Expressway?

 


I've got over 10,000 hits on this built-in rule and it seems to be blocking 
some legitimate logins via Jabber.


 


It looks like this in the event log:



2015-09-11T21:05:09-04:00   sh[1195]: Event="Intrusion Protection" 
Src-ip="X.X.X.X" Detail="Collaboration Edge HTTP Intrusion Protection blocking 
X.X.X.X" Level="INFO" UTCTime="2015/09/12-01:05:09"


2015-09-11T21:05:09-04:00   traffic_server[24581]: Event="Sending HTTP error 
response" Status="429" Reason="Unknown Status" Dst-ip="X.X.X.X" 
Dst-port="52940" UTCTime="2015-09-12 01:05:09,151" 



 


It looks like this in the Jabber log:



2015-09-11 17:09:15,746 INFO  [0x0dc0] 
[ls\src\http\BasicHttpClientImpl.cpp(399)] [csf.httpclient] 
[csf::http::executeImpl] - *-* HTTP response code 0 for request #2 to

https://myexpressway.client.com:8443/bG9naWNub3cuY29t/get_edge_config?service_name=_cisco-uds_name=_cuplogin


2015-09-11 17:09:15,746 ERROR [0x0dc0] 
[ls\src\http\BasicHttpClientImpl.cpp(404)] [csf.httpclient] 
[csf::http::executeImpl] - There was an issue performing the call to 
curl_easy_perform for request #2: CONNECTION_TIMEOUT_ERROR



 


It looks like this in the detailed expressway logging:



2015-09-11T11:12:06-04:00 atlitexpe1 UTCTime="2015-09-11 15:12:06,146" 
Event="System Configuration Changed" Node="clusterdb@127.0.0.1" 
PID="<0.3251.0>" Detail="xconfiguration fail2banJailStatus uuid
 12f52e25-4df6-4fd3-9697-621d9de3a796 jail: http-ce-intrusion total_fails - 
changed from: 202411 to: 202416"



 


 


Anyone else seeing issues like this?  This particular user also has an 8841 at 
home.  Is there a limit to number of MRA connections behind a single public IP?


 


Thanks,


Brian Meade






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Re: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection

[Ryan Huff]

Brian  I had this issue this weekend in 8.6.  My original issue was the "no 
home uds cluster" but I had issues with the proxy protocol violation.

Tac's response was go to 8.6.1 (released 9/11/15 ... yikes) or roll back to 8.5

Thanks,

Ryan

 Original Message 
From: Brian Meade 
Sent: Monday, September 14, 2015 03:49 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection

>Is anyone else having issues with the "HTTP proxy protocol violation"
>automated detection feature or Expressway?
>
>I've got over 10,000 hits on this built-in rule and it seems to be blocking
>some legitimate logins via Jabber.
>
>It looks like this in the event log:
>2015-09-11T21:05:09-04:00 sh[1195]: Event="Intrusion Protection"
>Src-ip="X.X.X.X" Detail="Collaboration Edge HTTP Intrusion Protection
>blocking X.X.X.X" Level="INFO" UTCTime="2015/09/12-01:05:09"
>2015-09-11T21:05:09-04:00 traffic_server[24581]: Event="Sending HTTP error
>response" Status="429" Reason="Unknown Status" Dst-ip="X.X.X.X"
>Dst-port="52940" UTCTime="2015-09-12 01:05:09,151"
>
>It looks like this in the Jabber log:
>2015-09-11 17:09:15,746 INFO  [0x0dc0]
>[ls\src\http\BasicHttpClientImpl.cpp(399)] [csf.httpclient]
>[csf::http::executeImpl] - *-* HTTP response code 0 for request #2 to
>https://myexpressway.client.com:8443/bG9naWNub3cuY29t/get_edge_config?service_name=_cisco-uds_name=_cuplogin
>2015-09-11 17:09:15,746 ERROR [0x0dc0]
>[ls\src\http\BasicHttpClientImpl.cpp(404)] [csf.httpclient]
>[csf::http::executeImpl] - There was an issue performing the call to
>curl_easy_perform for request #2: CONNECTION_TIMEOUT_ERROR
>
>It looks like this in the detailed expressway logging:
>2015-09-11T11:12:06-04:00 atlitexpe1 UTCTime="2015-09-11 15:12:06,146"
>Event="System Configuration Changed" Node="clusterdb@127.0.0.1"
>PID="<0.3251.0>" Detail="xconfiguration fail2banJailStatus uuid
>12f52e25-4df6-4fd3-9697-621d9de3a796 jail: http-ce-intrusion total_fails -
>changed from: 202411 to: 202416"
>
>
>Anyone else seeing issues like this?  This particular user also has an 8841
>at home.  Is there a limit to number of MRA connections behind a single
>public IP?
>
>Thanks,
>Brian Meade
>
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[cisco-voip] Expressway mra with webex connect config but not used

[Ryan Huff]

A word on using the mobile jabber clients with mra when webex connect is 
configured on the domain (but isn't being used and can't be turned off for one 
reason or another) ...

The only way I was able to get the provisioning URL to work for the mobile 
clients (so the clients will exclude WebEx service lookup) was to delete the 
app on the device, reinstall the app, then use the URL first after 
reinstallation. Using the URL on the app as-is had no impact, had to reinstall 
the app.

Best way is to host the provisoning URL on an html page and hit it from a 
browser on the device (only way it works on Android).

VCS 8.6 (8.6.1 just came out on CCO  9/11/15)
Jabber iPad 11.0.3
Jabber iPhone 11.0.3
Jabber Android 11.0.1

Thanks,

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Re: [cisco-voip] Error when SSHing into CUCM 10.5

[Ryan Huff]

So this is less a CUCM error as it is actually an issue with RedHat (OS). 
Basically, your platform administrator account is roached (doesn't have the 
right permissions) and when you try to log in -SELinux is spanking you. 

When you installed the patch, did you install with out a switch-version or with 
a switch-version? When it failed; di you reboot the cluster after the failure? 
Was this the publisher that is failed on?

Thanks,

Ryan


Date: Thu, 3 Sep 2015 13:34:21 -0500
From: jonv...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Error when SSHing into CUCM 10.5

Unable to get valid context for administrator
Last login: Thu Sep  3 11:09:07 2015 from 10.10.55.33
Had a failed patch from 10.5.2.11900 to 12901... switched back and everything 
appears to work, but can't SSH into the host.
Google is devoid of results...


Jonathan


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Re: [cisco-voip] Error when SSHing into CUCM 10.5

[Ryan Huff]

Also, if you are able to login into OS Administration (GUI) with the platform 
administrator account, but just can't SSH then you may be missing the 
/etc/selinux/targeted/contexts/users/sshout_u file. If you can get into the 
root account (had it previously enabled before the failed patch), you can 
create the sshout_u file and base it off of 
etc/selinux/targeted/contexts/users/guest_u

You may have to recover the OS though, if you can't get into root. I'd raise a 
TAC case and let them do it if it is a production system.

*** DISCLAIMER ***

I am not endorsing or recommending you to do anything I have described in this 
message. This is for educational purposes only.

From: ryanh...@outlook.com
To: jonv...@gmail.com; cisco-voip@puck.nether.net
Date: Thu, 3 Sep 2015 14:45:55 -0400
Subject: Re: [cisco-voip] Error when SSHing into CUCM 10.5




So this is less a CUCM error as it is actually an issue with RedHat (OS). 
Basically, your platform administrator account is roached (doesn't have the 
right permissions) and when you try to log in -SELinux is spanking you. 

When you installed the patch, did you install with out a switch-version or with 
a switch-version? When it failed; di you reboot the cluster after the failure? 
Was this the publisher that is failed on?

Thanks,

Ryan


Date: Thu, 3 Sep 2015 13:34:21 -0500
From: jonv...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Error when SSHing into CUCM 10.5

Unable to get valid context for administrator
Last login: Thu Sep  3 11:09:07 2015 from 10.10.55.33
Had a failed patch from 10.5.2.11900 to 12901... switched back and everything 
appears to work, but can't SSH into the host.
Google is devoid of results...


Jonathan


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[cisco-voip] IM DRS Restore Question

[Ryan Huff]

I have an 8.4 HA Presence cluster that I am doing a DRS restore on. The 
question I have is regarding contact lists. Will I recover all the user contact 
lists and data by restoring just the primary node or do I need to also restore 
the failover node (rather than just rebuild a new failover) to recover all the 
contact lists?

Thanks,

Ryan
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Re: [cisco-voip] Service Provider SIP Trunks

[Ryan Huff]

Thanks,

Ryan

 Original Message 
From: Ryan Huff ryanh...@outlook.com
Sent: Friday, August 28, 2015 05:12 PM
To: amichaelba...@hotmail.com
Subject: RE: [cisco-voip] Service Provider SIP Trunks

As long as you didn't change anything on the cpe side, it may be more likely 
your itsp changed more than just the pe.

You should be able to reproduce a failed call and provide your itsp with the 
session id and they should be able trace it; sometimes that encourages them to 
find whatever they broke.

Thanks,

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Re: [cisco-voip] Service Provider SIP Trunks

[Ryan Huff]

Sounds like a codec/media issue. Are you supporting early offer?

Thanks,

Ryan

 Original Message 
From: Aaron Banks amichaelba...@hotmail.com
Sent: Friday, August 28, 2015 03:35 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Service Provider SIP Trunks



I have a strange problem with SIP trunks.  Let me preface this with the 
service provider moved the SIP trunks to a different device and that's what 
certain calls stopped working.  Before this move, everything was tested and 
working for 6 weeks.  Read on, someone might have had the same experience.

Post SIP trunk move, callers inside of the organization cannot call 911 or a 
mobile phone (ANY mobile phone).  When they dial the number, let's use 911 for 
example, the call rings once, the calling line ID is delivered to 911 and then 
the call goes to busy.  So 911 knows that organization called.  The same thing 
happens with mobile phones.  All other call types (local, LD, international) 
work.  If I call forward a phone from inside of the organization to a mobile 
phone and call that forwarded phone (from outside or inside), the call is 
redirected to the mobile, call is answered and then the call drops.  If I 
forward that same phone inside of the organization to an outside land line 
((either local or LD), the call is successful.

For 911 or the mobile calls that fail, the SIP trace reveals a 500 (internal 
server error), a BYE message with reason Q.850; cause=16.  The SIP call 
messages show the state of the call is DEAD.

My question is why would the number make any difference at all?  Has anyone 
ever seen this type of issue before?  The provider says it is CUCM 10.5/Voice 
GW 2901 that is rejecting the call.  I disagree.

Many thanks

Aaron

 
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Re: [cisco-voip] IP Phone 7970 Firmware

[Ryan Huff]

Yeah, I broke down and used my 'phone a friend' lifeline .. TAC


From: kevin.dami...@oneneck.com
To: kevin.dami...@oneneck.com; ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] IP Phone 7970 Firmware
Date: Wed, 26 Aug 2015 14:16:12 +









Correction - it found the page, but you get the “software is no longer 
available” message when you download it.
J
 

Kevin Damisch
CCIE #28188 (Voice / Data Center)
Network Engineer IV
OneNeck®  IT Solutions, a TDS® Company
Phone:  (319) 866-7985
Email: 
kevin.dami...@oneneck.com
Website: 
www.oneneck.com

 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Damisch, Kevin

Sent: Wednesday, August 26, 2015 9:13 AM

To: Ryan Huff ryanh...@outlook.com; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] IP Phone 7970 Firmware


 
Strange.  You can’t browse for it, but Google found it.
 
https://software.cisco.com/download/release.html?mdfid=278436620softwareid=282074288release=9.4(2)SR1
 
Kevin
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff

Sent: Wednesday, August 26, 2015 8:44 AM

To: cisco-voip@puck.nether.net

Subject: [cisco-voip] IP Phone 7970 Firmware


 

Is it just me  not finding 7970 listed on CCO?

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