Re: [cisco-voip] Looking for advice on sRTP using tokenless CTL

[Justin Steinberg]

Ryan,

Could you elaborate a little on this change ?I did an upgrade on my lab
system to 11.5(1)su3a and after the upgrade I see from the output of 'show
itl' that the callmanager.pem is still the signer of the ITL file.

Would the ITLRecovery only be the signer of the ITL on a fresh install of
11.5(1)su3/3a ?

The release notes for su3 have a revision history update on October 23rd
that says "Removed Enhanced CTL and ITL Trust information, which is not
available with this release."I'm not sure if that's related to your
comment or if something has changed.

Justin




On Fri, Oct 20, 2017 at 10:13 AM, Ryan Ratliff (rratliff) <
rratl...@cisco.com> wrote:

> Additionally in 11.5SU3 and 12.0 we now sign the ITL and tokenless CTL
> files with the ITLRecovery certificate instead of CallManager.pem.
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/
> rel_notes/11_5_1/SU3/cucm_b_release-notes-cucm-imp-
> 1151su3/cucm_b_release-notes-cucm-imp-1151su3_chapter_00.html#reference_
> 9C103B26C27EFA3008B593B943A6950C
>
> This should greatly reduce the risk of inadvertent trust list issues
> caused by certificate operations.
>
> -Ryan
>
> On Oct 19, 2017, at 3:08 PM, Brian Meade  wrote:
>
> In 10.x, CTL/ITL are signed by the CallManager.pem if you do tokenless.
> There's a recovery key in the ITL so you can run "utils itl reset localkey"
> to resign the ITL with the recovery key to recover your cluster.  But this
> doesn't help you with devices that don't support SBD like Jabber.
>
> In 11.x, CTL/ITL are signed by the CallManager.pem if you do tokenless but
> there's also a recovery key for the CTL.  You can run "utils ctl reset
> localkey" to resign the CTL with the recovery key to recover your cluster.
>
> So I'd recommend in your case that you upgrade to 11.x first or use
> physical tokens.
>
> The latest 11.5 SU requires you to order a free encryption license through
> PUT as well.
>
>
>
> On Tue, Oct 17, 2017 at 2:01 PM, Ryan Huff  wrote:
>
>> Looking at enabling sRTP on a 10.x cluster (CUCM, EXPRESSWAY, CXN, UCCX).
>> As I have been researching this topic; I’ve found the “riskiest” task to be
>> enabling CTL / Mixed Mode in CUCM. Specifically, if you have devices that
>> do not support Security By Default.
>>
>> It’s my understanding that once the callmanager cert changes, any device
>> that can’t negotiate with the TVS service to establish verification will
>> not be able to download the new CTL, and therefore not be able to
>> re-register to CUCM until their CTL is removed.
>>
>> The device/trunk security profile configurations seems straight forward
>> as do the steps to take on CUBE and Expressway (regarding the trunk
>> security).
>>
>> I haven’t completed my research into the CXN/UCCX requirements for SRTP
>> with CUCM.
>>
>> Are their any other major/general pitfalls I should look out for? Anyone
>> have any horror stories or lessons learned to share?
>>
>> Thanks,
>>
>> Ryan
>>
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Re: [cisco-voip] UDS Searches not sanitizing the Plus Symbol

[Justin Steinberg]

i meant to say it will also find numbers with hypens (the documentation
mentions they are ignored)

On Thu, Apr 13, 2017 at 8:05 AM, Justin Steinberg <jsteinb...@gmail.com>
wrote:

> While Anthony's encoding is right, I actually think the documentation does
> intend to actually mean you can act as though the + was not in the phone
> number value as stored in AD.
>
> I have used the query below
>
> https://{host}:8443/cucm-uds/users?numberlast=3175551234
> <(317)%20555-1234>
>
> To search for phone numbers in AD.  This finds any number with right
> justified digits above.  This should find results for +13175551234
> <(317)%20555-1234>, 13175551234 <(317)%20555-1234>, and 3175551234
> <(317)%20555-1234>.   If there are hypens in the numbers in AD (e.g.
> 317-555-1234 <(317)%20555-1234>).
>
> On Thu, Apr 13, 2017 at 12:16 AM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
>> No worries.  Transitioning to app dev will be a bumpy road for a lot of
>> UC Engineers, we might as well help each other progress forward.
>>
>> On Wed, Apr 12, 2017 at 9:37 PM Nathan Reeves <nathan.a.ree...@gmail.com>
>> wrote:
>>
>>> lol, cheers for that, should have picked that up earlier.  Quick test
>>> shows that works perfectly.
>>>
>>> When the docs mention that it ignores the plus symbol, I was working on
>>> the assumption that it therefore would ignore the plus symbol in the actual
>>> number.  Wrong assumption.
>>>
>>> Thanks again
>>>
>>>
>>> On Wed, Apr 12, 2017 at 11:57 PM, Anthony Holloway <
>>> avholloway+cisco-v...@gmail.com> wrote:
>>>
>>>> Plus signs (+) in URL query parameters (the part after the question
>>>> mark [?]) are treated as spaces.  E.g., ?name=anthony+holloway == "anthony
>>>> holloway"  So, you're effectively asking UDS for " 6140011" (note the
>>>> leading space, and omission of the plus sign [+]).
>>>>
>>>> The work around is to use some sort of URL encoding library, which will
>>>> build your URL with the plus sign (+) encoded with it's percent sign (%)
>>>> equivelent, which happens to be %2B.
>>>>
>>>> So, your submitted UDS request would actually look like:
>>>>
>>>> https://172.20.2.21:8443/cucm-uds/users?number=%2B6140011
>>>>
>>>> Finally, this is not a function of UDS at all, and something you'll
>>>> need to know, now that you are explorely RESTful APIs which rely on URL
>>>> structures to work with data.
>>>>
>>>> On Wed, Apr 12, 2017 at 10:46 AM Nathan Reeves <
>>>> nathan.a.ree...@gmail.com> wrote:
>>>>
>>>>> Taking a look around at what options we have to drive additional
>>>>> directories for our IP phones and decided to take a quick look at using 
>>>>> UDS
>>>>> as the data source, accessing it via the published API.
>>>>>
>>>>> One thing I'm finding (which I can't see any bug report on), is that
>>>>> number searches, where the number in UDS contains a plus, does not return
>>>>> search results based on the query submitted.
>>>>>
>>>>> I have a user configured with a mobile number in PlusE164
>>>>> (+6140011 <+61%20400%20111%20111> for example) which is pulled
>>>>> into the CUCM directory via LDAP sync.
>>>>>
>>>>> The API docs note that brackets, plus symbols etc are all ignored in
>>>>> the search.  When I access the UDS API and construct a query string in a
>>>>> URL along the lines of 'https://172.20.2.21:8443/cucm
>>>>> -uds/users?number=61400', the returned response is 0 results.  If I
>>>>> update the Mobile number to remove just the plus (and resync LDAP), the
>>>>> same search now returns my user with the mobile number correctly searched.
>>>>>
>>>>> Running 11.5(1)SU1 (haven't yet checked this against SU2), attempted
>>>>> to use native UDS but also tried searching while UDS Proxy is enabled.
>>>>> Same results either way.
>>>>>
>>>>> Anyone seen this issue or am I missing something?  I can only assume
>>>>> that the sanitized query doesn't correctly ignore the plus symbol.
>>>>>
>>>>> Cheers
>>>>>
>>>>> Nathan
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>>>>>
>>>>
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Re: [cisco-voip] UDS Searches not sanitizing the Plus Symbol

[Justin Steinberg]

While Anthony's encoding is right, I actually think the documentation does
intend to actually mean you can act as though the + was not in the phone
number value as stored in AD.

I have used the query below

https://{host}:8443/cucm-uds/users?numberlast=3175551234

To search for phone numbers in AD.  This finds any number with right
justified digits above.  This should find results for +13175551234,
13175551234, and 3175551234.   If there are hypens in the numbers in AD
(e.g. 317-555-1234).

On Thu, Apr 13, 2017 at 12:16 AM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> No worries.  Transitioning to app dev will be a bumpy road for a lot of UC
> Engineers, we might as well help each other progress forward.
>
> On Wed, Apr 12, 2017 at 9:37 PM Nathan Reeves 
> wrote:
>
>> lol, cheers for that, should have picked that up earlier.  Quick test
>> shows that works perfectly.
>>
>> When the docs mention that it ignores the plus symbol, I was working on
>> the assumption that it therefore would ignore the plus symbol in the actual
>> number.  Wrong assumption.
>>
>> Thanks again
>>
>>
>> On Wed, Apr 12, 2017 at 11:57 PM, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>>> Plus signs (+) in URL query parameters (the part after the question mark
>>> [?]) are treated as spaces.  E.g., ?name=anthony+holloway == "anthony
>>> holloway"  So, you're effectively asking UDS for " 6140011" (note the
>>> leading space, and omission of the plus sign [+]).
>>>
>>> The work around is to use some sort of URL encoding library, which will
>>> build your URL with the plus sign (+) encoded with it's percent sign (%)
>>> equivelent, which happens to be %2B.
>>>
>>> So, your submitted UDS request would actually look like:
>>>
>>> https://172.20.2.21:8443/cucm-uds/users?number=%2B6140011
>>>
>>> Finally, this is not a function of UDS at all, and something you'll need
>>> to know, now that you are explorely RESTful APIs which rely on URL
>>> structures to work with data.
>>>
>>> On Wed, Apr 12, 2017 at 10:46 AM Nathan Reeves <
>>> nathan.a.ree...@gmail.com> wrote:
>>>
 Taking a look around at what options we have to drive additional
 directories for our IP phones and decided to take a quick look at using UDS
 as the data source, accessing it via the published API.

 One thing I'm finding (which I can't see any bug report on), is that
 number searches, where the number in UDS contains a plus, does not return
 search results based on the query submitted.

 I have a user configured with a mobile number in PlusE164 (+6140011
 <+61%20400%20111%20111> for example) which is pulled into the CUCM
 directory via LDAP sync.

 The API docs note that brackets, plus symbols etc are all ignored in
 the search.  When I access the UDS API and construct a query string in a
 URL along the lines of 'https://172.20.2.21:8443/
 cucm-uds/users?number=61400', the returned response is 0 results.  If
 I update the Mobile number to remove just the plus (and resync LDAP), the
 same search now returns my user with the mobile number correctly searched.

 Running 11.5(1)SU1 (haven't yet checked this against SU2), attempted to
 use native UDS but also tried searching while UDS Proxy is enabled.  Same
 results either way.

 Anyone seen this issue or am I missing something?  I can only assume
 that the sanitized query doesn't correctly ignore the plus symbol.

 Cheers

 Nathan
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>>
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Re: [cisco-voip] Question regarding Rightfax Configuration and "Transmission Error"

[Justin Steinberg]

I use t38 over AT just fine

On Thu, Apr 6, 2017 at 10:37 AM, Brian Meade  wrote:

> I've got SIP with T.38 as well as PRI in different environments that are
> all working pretty well with that setup.
>
> On Thu, Apr 6, 2017 at 10:24 AM, Haas, Neal  wrote:
>
>> Do you use SIP or PRI, we use PRI 99.9% of the time, ATT does not allow
>> T.38 over SIP.
>>
>>
>>
>> I will likely get a SIP trunk from Comcast or Time Warner in the future,
>> they both do T.38 over SIP. Both say they can do Faxes without issue.
>>
>>
>>
>> Thank You,
>>
>>
>>
>> Neal Haas
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Brian Meade
>> *Sent:* Thursday, April 6, 2017 7:09 AM
>> *To:* Frank Arrasmith 
>> *Cc:* cisco-voip@puck.nether.net
>> *Subject:* Re: [cisco-voip] Question regarding Rightfax Configuration
>> and "Transmission Error"
>>
>>
>>
>> I've had the most luck turning ECM off and running RightFax at only
>> 9600.  Faxes are slower but usually higher quality and less failures.
>>
>>
>>
>> On Wed, Apr 5, 2017 at 8:06 PM, Frank Arrasmith <
>> frank.arrasm...@gmail.com> wrote:
>>
>> Calling all Rightfax gurus,
>>
>>I have the following question regarding the Rightfax configuration and
>> transmission errors.
>>
>> Background:
>> My enterprise has CUCM 10.5 with a pretty dialed in Fax/T38 setup over
>> SIP and MGCP gateways.  For the most part, faxing is pretty solid inbound
>> and outbound to and from PSTN GW's, CUBEs, ,analog VG's, and regular fax
>> machines(we have a lot).  We have a SIP trunk connected to Rightfax 10.5
>> server, which is managed by another group where we have limited
>> access/experience with Rightfax configuration settings.  It took us awhile
>> to get the Rightfax servers with the correct t38 setup because they had
>> only run traditional CAS T1 prior to us, so it was new to everyone, but it
>> is up and stable except for the following issue..
>>
>> Symptom :
>>
>> The problem we see with Rightfax is with Transmission errors.  It starts
>> with our internal customers reporting that they do not receive a fax even
>> when the sender receives confirmation.  Upon further review we see that the
>> suspect call is listed as a "Transmission Error" in Rightfax, so the fax
>> never gets delivered to the customers account/mailbox even though the call
>> completes.
>>
>> Analysis:
>>
>> Since we are running T38, we can packet capture at the server, and  we
>> see normal fax protocol exchange, except for the suspect calls where we see
>> "RTN" Messages.  My understanding of the T.30 protocol is that when a RTN
>> message is delivered to the sender, that is an indication for the sender to
>> slow down and resend the last page. We actually see in the messages where
>> the RTN message gets sent, and the sender complies with the notice, and
>> sends again at 12000, then call completes as normal with an EOP message and
>> a DCN message.  In these cases,the Rightfax team actually looks at the fax
>> image and may see a bit of blurriness, but perfectly legible text ,even tho
>> its still marked as transmission failure. We have asked them if there is a
>> setting that can be tuned where the RTN does not cause the service to mark
>> the transmission as failure. To this they reply, "It was working fine
>> before when we were on CAS, so it must be on your end."
>>
>> I understand where there are cases where the RTN message is sent because
>> the call quality is actually terrible, but in those cases, there is usually
>> several RTN messages and the sender will drop down to 4800 or below , and
>> then usually give up and the call will fail.  This type of failure is rare
>> (unless we have a major outage) and in this case the sending fax sees that
>> it failed and will proceed with its normal retries.
>>
>> Question:
>>
>> Is this RTN to transmission failure hard coded, or is this a configurable
>> setting?  If this were a regular fax machine, i think this would be a non
>> issue as the receiver would see the crappy page as well as the good copy of
>> that was sent again in their bundle of received pages.
>>
>> Any insight is greatly appreciated and for anyone just getting into FAX
>> over IP with Cisco, I highly recommend the following book and Cisco Live
>> presentations from these guys from Cisco TAC.
>>
>> Fax, Modem, and Text for IP Telephony [Book]
>> 
>>
>> from Textbooks.com
>>
>> by David Hanes, Gonzalo Salgueiro
>>
>>
>>
>>
>>
>> Thanks,
>>
>>  Frank
>>
>>
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Re: [cisco-voip] Integrating IM only between Jabber & S4B cross-organizations

[Justin Steinberg]

This is possible now on the latest version of expressway x8.9 and IMP
server 11.5(1)su2.

This will only work between two different domain names, which seems to be
what you want to do.

This will provide IM & Presence.   Voice & Video calling would require CMS.

On Feb 23, 2017 4:54 PM, "Ben Amick"  wrote:

> So I’m being asked if it’s possible to integrate S4B on O365 and Jabber –
> I know not natively, as Skype/Lync isn’t an XMPP client, and I know there’s
> the connector for expressway to be able to do full IMP & Video, but that’s
> all internal. Is there any way to integrate messaging between S4B and
> Jabber between two separate organizations, especially since one is on O365?
>
>
>
> *Ben Amick*
>
> Telecom Analyst
>
>
>
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Re: [cisco-voip] Not supported I'm sure..... but what do you think?

[Justin Steinberg]

Keep in mind that if you COBRAS data significantly before the cut and then
use message shuttle that you'll get all messages moved over at cut time but
you'll miss things like users who update their voicemail greetings daily,
or put an out of office in place for vacation, etc.

On Tue, Nov 1, 2016 at 12:39 AM, Dave Goodwin 
wrote:

> Agreed on the Connection Shuttle. I have used it and it's quite fast
> compared to the alternative. I encourage you to read the tool help at the
> link provided.
>
> One important point is that the machine running Connection Shuttle
> requires live connectivity to the source and target CUC servers. If you
> have a "live" CUC server, and a restored copy of it on a dead network, I'm
> not sure if Shuttle can talk to both. Maybe if your Windows machine that's
> running it has 2 NICs, maybe if there's a NAT (so your dead network copy
> can keep using the native IP of the CUC it's copying). But, I've never
> tried this.
>
> On Tue, Nov 1, 2016 at 12:37 AM, Erick Wellnitz 
> wrote:
>
>> http://ciscounitytools.com/Applications/CxN/MessageShuttle/
>> MessageShuttle.html
>>
>> TAC supported even.
>>
>> On Oct 31, 2016 9:12 PM, "Lelio Fulgenzi"  wrote:
>>
>>> What is this message shuttle magic you soak of?
>>>
>>> Sent from my iPhone
>>>
>>> On Oct 31, 2016, at 5:37 PM, Ryan Huff  wrote:
>>>
>>> Plus +1 to Anthony
>>>
>>> Message shuttle is silly fast too as it is just doing file copies and
>>> not going through the COBRAS overhead.
>>>
>>> 1.) COBRAS (or DRS) initially to get the subscriber / handler schema
>>> moved over (keeping in mind what COBRAS will not replicate, i.e.
>>> restriction tables ... etc)
>>>
>>> 2.) Message shuttle right before the switch to move content.
>>>
>>> This also saves backing up messages initially  if using COBRAS (assuming
>>> COBRAS is being used in a migration / upgrade capacity and not just a
>>> backup), which makes that whole process a ton faster too.
>>>
>>> -Ryan
>>>
>>> On Oct 31, 2016, at 5:28 PM, Anthony Holloway <
>>> avholloway+cisco-v...@gmail.com> wrote:
>>>
>>> Or Message Shuttle?
>>>
>>> On Mon, Oct 31, 2016 at 11:10 AM, Matthew Loraditch <
>>> mloradi...@heliontechnologies.com> wrote:
>>>
 COBRAS would work for that.



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

 Facebook  | Twitter
  | LinkedIn
 
 | G+ 



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Scott Voll
 *Sent:* Monday, October 31, 2016 12:08 PM
 *To:* Peter Slow 
 *Cc:* cisco voip 

 *Subject:* Re: [cisco-voip] Not supported I'm sure. but what do
 you think?



 Our biggest problem in this situation would be VM.  if we restore from
 a backup on Monday in a Dark Network and switch over a weekend, is there a
 way to grab newer Voicemails to put on the "new" (upgraded) UC at the time
 of the switch?



 Scott





 On Sat, Oct 29, 2016 at 2:03 PM, Peter Slow 
 wrote:

 Hey guys. At a Co. I was working at with a megacluster, we took backups
 of the production cluster, and restored to a newly installed cluster on a
 "dark" net, and I NAT'd all the VMs so they could run and be upgraded
 separately and rebooted and tested with a few phones on a private network.
 Then I used some VMware Power CLI scripts to cut over each datacenter in a
 few seconds... ECATS was involved as well - there is a little bit of a
 precedent, though I recognize that we tailored that specifically for the
 customer's environment, and it might not work as well in some others.

 It should also be noted that the NEW (and subsequently upgraded, and
 switched into production) VMs were built using the actual TAC supported
 install process, as opposed to being cloned. The fact that the megacluster
 had 21 servers in basically equated to "not enough time to roll back before
 monday" for a number of reasons, an issue this upgrade & cutover method
 alleviated. I think it was an 8.6 to 9 upgrade. Because of some block level
 stuff, sine VMware wasn't supported (IIRC) in the starting version, after
 the upgrade of the 2nd set (referred to as "new" earlier)  we took another
 back up and restored to *another* cluster, a set of newly installed 9.x VMs
 in *another* "dark" net.



 BE CAREFUL IF YOU DO THIS

 I did 5 production clusters this way - Having two to three sets of
 identical servers, with identical IPs 

Re: [cisco-voip] Not supported I'm sure..... but what do you think?

[Justin Steinberg]

The upgrades take too long is part of it.  Especially if the upgrade is an
RU upgrade, because it means that both the patch install/upgrade and the
switch needs to take place in a maintenance mode.

The missing piece in my opinion is a way to put CUCM into a maintenance
mode where it continues to service active calls (on ccm, cti, ipvms
processes, etc) but forces new calls/registrations to another cucm.

Unity Connection does have a nice implementation of this (i.e. 'stop taking
calls') which makes for completely transparent reboots, etc.

If CUCM had a maintenance mode like feature, we would be able to do these
things during the day without causing problems.

Justin

On Thu, Oct 27, 2016 at 1:22 PM, Ryan Ratliff (rratliff)  wrote:

> Honest question, what exactly is it about the current implementation that
> fails to deliver on this?
>
> Is it something in the design of the upgrade process?
>
> Is it that the upgrade takes too long to be done during any reasonable
> maintenance window?
>
> Is it that you have to test the new version before you roll it into
> production?
>
> Is it >
>
> -Ryan
>
> On Oct 27, 2016, at 12:02 PM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> If only there was an upgrade process wherein you install the new version
> to an inactive partition, and then could switch to the new version when
> you're ready.  /sarcasm
>
> But seriously though, everyone in this thread is essentially coming up
> with their own clever way of replicating the promise Cisco failed to
> deliver on, which is performing your upgrades during production on the
> inactive partition and then switching versions in a maintenance window.  If
> they would have only held themselves to a higher standard, we wouldn't need
> this complex of an alternate solution.
>
> On Tue, Oct 25, 2016 at 2:45 PM, Ryan Huff  wrote:
>
>> Matthew is correct, copying is listed as "Supported with Caveats" at:
>> http://docwiki.cisco.com/wiki/Unified_Communications_VMware_Requirements;
>> The caveat being found at http://docwiki.cisco.com/wi
>> ki/Unified_Communications_VMware_Requirements#Copy_Virtual_Machine
>>
>>
>> The VM needs to be powered down first and the resulting VM will have a
>> different MAC address (unless it was originally manually specified); so
>> you'll need to rehost the PLM if it is co-res to any VM that you copy.
>>
>>
>> Where I have seen folks get into trouble with this is where a subscriber
>> is copied, and the user mistakenly thinks that by changing the IP and
>> hostname it becomes unique and can be added to the cluster as a new
>> subscriber. I have also seen users make a copy of a publisher and change
>> the network details of the copy, thinking it makes a unique cluster and
>> then wonders why things like ILS wont work between the two clusters (and it
>> isn't just because the cluster IDs are the same).
>>
>>
>> Having said all of that, I would NEVER do this in production ... maybe
>> that is just me being cautious or old school, but that is just me. Even
>> without changing network details on the copy, I have seen this cause issues
>> with Affinity. At the very least, if you travel this path I would make sure
>> that the copy runs on the same host and even in the same datastore.
>>
>>
>> === An alternative path ===
>>
>>
>> Admittedly, this path is longer and there is a little more work involve
>> but is the safer path, IMO and is what I would trust for a production
>> scenario.
>>
>>
>> 1.) Create a private port group on the host. If the cluster is on
>> multiple hosts, span the port group through a connecting network to the
>> other hosts but DO NOT create an SVI anywhere in the the topology for that
>> DOT1Q tag (remembering to add a DOT1Q tag on any networking devices between
>> the two hosts and allowing on any trunks between the two hosts).
>>
>>
>> 2.) Upload Cisco's CSR1000V to the host. If you're not familiar with the
>> product it is at the core and unlicensed, a virtual router with three
>> interfaces by default. Out of the box, it is more than enough to replicate
>> DNS/NTP on your private network which is all you'll need. Assign the
>> private port group to the network adapters and configure DNS and NTP
>> (master 2) on this virtual router.
>>
>>
>> 3.) Build out a replica of your production UC cluster on the private
>> network.
>>
>>
>> 4.) Take a DRS of the production UC apps and then put your SFTP server on
>> the private network and do a DRS restore to the private UC apps.
>>
>>
>> 5.) Upgrade the private UC apps and switch your port group labels on the
>> production/private UC apps during a maintenance window.
>>
>>
>> Thanks,
>>
>>
>> Ryan
>>
>>
>>
>>
>> --
>> *From:* cisco-voip  on behalf of
>> Matthew Loraditch 
>> *Sent:* Tuesday, October 25, 2016 3:01 PM
>> *To:* Tommy Schlotterer; Scott Voll; cisco-voip@puck.nether.net
>>
>> 

Re: [cisco-voip] Video Conference Auto attendant?

[Justin Steinberg]

is this a joke?  how can CUWL pro just not exist anymore for people who
bought it for conferencing functions.

On Wed, Aug 3, 2016 at 3:08 PM, Matthew Loraditch <
mloradi...@heliontechnologies.com> wrote:

> Yeah, CUWL Pro no longer exists and is now CUWL Meetings and is $100 more.
> You can of course upgrade your cuwl pro licenses to CUWL meetings for that
> $100. Your SWSS doesn’t get you a thing. The PMP is now called PMP+
>
> Prices are list so expect your regular discounts.
>
>
>
> So basically we are operating on a 4th iteration of Conferencing
> licensing in two years.  Similar changes exist with the SMP licensing, now
> called SMP+.
>
>
>
> Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
> Network Engineer
> Direct Voice: 443.541.1518
>
> Facebook  | Twitter
>  | LinkedIn
>  |
> G+ 
>
>
>
> *From:* Scott Voll [mailto:svoll.v...@gmail.com]
> *Sent:* Wednesday, August 3, 2016 3:00 PM
> *To:* Lelio Fulgenzi 
> *Cc:* Matthew Loraditch ; Erick
> Wellnitz ; cisco-voip@puck.nether.net
>
> *Subject:* Re: [cisco-voip] Video Conference Auto attendant?
>
>
>
> anyone hear anything revealed?
>
>
>
> Scott
>
>
>
> On Mon, Jul 25, 2016 at 3:52 PM, Lelio Fulgenzi  wrote:
>
>
>
> There was quite a bit of talk at Live about CMS/Acano, and CUWL pro and
> the [s|p]mp ELA and it sounded very much like there would be a true up of
> sorts to get the CMS licenses. All will be revealed after aug 1.
>
> Sent from my iPhone
>
>
> On Jul 25, 2016, at 5:18 PM, Matthew Loraditch <
> mloradi...@heliontechnologies.com> wrote:
>
> CMS will be included in CUWL pro just like TPS is effective 8/1 with the
> new FY. It will also integrate with TMS for scheduling w/ 15.3 in august.
> Hurry up and get somebody to order the free upgrade to CUWL pro from STD
> before Friday and all you have to do is pay for the upgraded SWSS and get a
> new UCS if you  don’t have hardware available.
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
> ] *On Behalf Of *Scott Voll
> *Sent:* Monday, July 25, 2016 2:34 PM
> *To:* Erick Wellnitz 
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Video Conference Auto attendant?
>
>
>
> Great.. I get to spend more money...
>
>
>
> On Mon, Jul 25, 2016 at 9:48 AM, Erick Wellnitz 
> wrote:
>
> Take a look at the Acano/CMS product.  It has IVR capability and can
> integrate with pretty much everything.
>
>
>
> On Jul 25, 2016 9:54 AM, "Scott Voll"  wrote:
>
> What are others doing for video conferences?
>
>
>
> we have a Polycom RMX that we have an extension that everyone dials into.
> Then it asks what ID you would like to use.  when everyone dials into that
> Conference ID, they are all apart of the same Video conference.
>
>
>
> With the Telepresence Server, how do you get that same functionality?
>
>
>
> Thanks
>
>
>
> Scott
>
>
>
>
>
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>
>
>
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>
>
>
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Re: [cisco-voip] DTMF interworking on CUBE - asymmetric payloads

[Justin Steinberg]

interesting - i wonder why that is not supported when it works.  doc error
or some legit technical issue ?

On Tue, Jul 19, 2016 at 3:44 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I do it to, but did you know that RTP-NTE to SIP-KPML is not supported on
> CUBE as of yet?
>
>
> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html#concept_264617919921874995299551391601561__table_16E37E2F33CE4E0B836D2E5A809E7252
>
> On Mon, Jul 18, 2016 at 8:21 PM, Justin Steinberg <jsteinb...@gmail.com>
> wrote:
>
>> yes, CUBE can do RFC2833/NTP to a Telco and SIP-KPML to CUCM.   I do this
>> for calls that terminate on CCX IVR since CCX does not support RFC2833.
>> With only rtp-nte on the dialpeer from CUBE to CUCM, CUCM will invoke a
>> MTP.   Adding sip-kpml to the dial-peer will allow RTP directly from CUBE
>> to CCX without any MTP in the middle.
>>
>> On Mon, Jul 18, 2016 at 5:08 PM, Ed Leatherman <ealeather...@gmail.com>
>> wrote:
>>
>>> Thanks Daniel, that helps a lot in understanding the feature. I'm
>>> curious if CUBE will also translate digits to KPML in this case if the leg
>>> to CUCM has that negotiated. Wish I had a lab built out for this :)
>>>
>>>
>>>
>>> On Mon, Jul 18, 2016 at 4:22 PM, Daniel Pagan <dpa...@fidelus.com>
>>> wrote:
>>>
>>>> Ed:
>>>>
>>>>
>>>>
>>>> I specifically worked with the dynamic payload option for a few cases
>>>> that came my way. Based on my findings, when a dynamic payload type (such
>>>> as 100/101/etc.) is received by CUBE, it will check if the next-hop
>>>> dial-peer has the asymmetric payload feature enabled and, if it is, will
>>>> pass the received payload type through to the next call-leg. Take a look at
>>>> my screen shot below. This was taken from some old notes where AT was the
>>>> customer’s carrier.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> The call flow above shows two call-legs, and *the arrows represent an
>>>> offer/answer exchange*.
>>>>
>>>>
>>>>
>>>> With asymmetric payload enabled on both call legs, the 100 offer from
>>>> ATT is passed to CUCM despite 101 being the default PT for NTE. In the SDP
>>>> answer from CUCM, we’re getting PT 101 -- since asymmetry is enabled on the
>>>> DP to ATT in this call flow, we pass the 101 through to ATT despite having
>>>> received PT 100.
>>>>
>>>>
>>>>
>>>> This results in asymmetry on our negotiated PT for each call-leg.
>>>>
>>>>
>>>>
>>>> *Let’s change it up a bit… A second example.*
>>>>
>>>> If asymmetry was disabled on the dial-peer to CUCM but enabled to ATT,
>>>> we would receive 100 PT from ATT, send 101 to CUCM, receive 101 from CUCM,
>>>> and send 101 to ATT. The resulting PTs would be symmetrical between CUBE
>>>> and CUCM, but asymmetrical between CUBE and ATT.
>>>>
>>>>
>>>>
>>>> See screenshot below for a third example:
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> This example shows asymmetric payload disabled on both call-legs using
>>>> the same call flow. CUBE receives PT of 100 from ATT -- the outbound
>>>> dialpeer has asymmetry disabled, so it transmits the PT specified for that
>>>> dial-peer (default 101 or any hardcoded dynamic PT) to CUCM. We then
>>>> receive 101 from CUCM and, since our inbound dial-peer has asymmetry
>>>> disabled, CUBE sends 100 to match the original PT it received. Asymmetry is
>>>> disabled so CUBE is not passing the received dynamic PT through to the
>>>> next-hop dial-peer - we have symmetry on both call legs for our NTE PT.
>>>>
>>>>
>>>>
>>>> Note that CUBE has no issues receiving one dynamic PT for NTE and
>>>> sending another (ex: receiving PT 100 and transmitting 101 for RTP-NTE) on
>>>> the same call leg.
>>>>
>>>>
>>>>
>>>> Hope this helps
>>>>
>>>>
>>>>
>>>> - Dan
>>>>
>>>> end attach-
>>>>
>>>>
>>>>
>>>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>>>> Behalf Of *Ed Leatherman
>>>>

Re: [cisco-voip] DTMF interworking on CUBE - asymmetric payloads

[Justin Steinberg]

yes, CUBE can do RFC2833/NTP to a Telco and SIP-KPML to CUCM.   I do this
for calls that terminate on CCX IVR since CCX does not support RFC2833.
With only rtp-nte on the dialpeer from CUBE to CUCM, CUCM will invoke a
MTP.   Adding sip-kpml to the dial-peer will allow RTP directly from CUBE
to CCX without any MTP in the middle.

On Mon, Jul 18, 2016 at 5:08 PM, Ed Leatherman 
wrote:

> Thanks Daniel, that helps a lot in understanding the feature. I'm curious
> if CUBE will also translate digits to KPML in this case if the leg to CUCM
> has that negotiated. Wish I had a lab built out for this :)
>
>
>
> On Mon, Jul 18, 2016 at 4:22 PM, Daniel Pagan  wrote:
>
>> Ed:
>>
>>
>>
>> I specifically worked with the dynamic payload option for a few cases
>> that came my way. Based on my findings, when a dynamic payload type (such
>> as 100/101/etc.) is received by CUBE, it will check if the next-hop
>> dial-peer has the asymmetric payload feature enabled and, if it is, will
>> pass the received payload type through to the next call-leg. Take a look at
>> my screen shot below. This was taken from some old notes where AT was the
>> customer’s carrier.
>>
>>
>>
>>
>>
>> The call flow above shows two call-legs, and *the arrows represent an
>> offer/answer exchange*.
>>
>>
>>
>> With asymmetric payload enabled on both call legs, the 100 offer from ATT
>> is passed to CUCM despite 101 being the default PT for NTE. In the SDP
>> answer from CUCM, we’re getting PT 101 -- since asymmetry is enabled on the
>> DP to ATT in this call flow, we pass the 101 through to ATT despite having
>> received PT 100.
>>
>>
>>
>> This results in asymmetry on our negotiated PT for each call-leg.
>>
>>
>>
>> *Let’s change it up a bit… A second example.*
>>
>> If asymmetry was disabled on the dial-peer to CUCM but enabled to ATT, we
>> would receive 100 PT from ATT, send 101 to CUCM, receive 101 from CUCM, and
>> send 101 to ATT. The resulting PTs would be symmetrical between CUBE and
>> CUCM, but asymmetrical between CUBE and ATT.
>>
>>
>>
>> See screenshot below for a third example:
>>
>>
>>
>>
>>
>> This example shows asymmetric payload disabled on both call-legs using
>> the same call flow. CUBE receives PT of 100 from ATT -- the outbound
>> dialpeer has asymmetry disabled, so it transmits the PT specified for that
>> dial-peer (default 101 or any hardcoded dynamic PT) to CUCM. We then
>> receive 101 from CUCM and, since our inbound dial-peer has asymmetry
>> disabled, CUBE sends 100 to match the original PT it received. Asymmetry is
>> disabled so CUBE is not passing the received dynamic PT through to the
>> next-hop dial-peer - we have symmetry on both call legs for our NTE PT.
>>
>>
>>
>> Note that CUBE has no issues receiving one dynamic PT for NTE and sending
>> another (ex: receiving PT 100 and transmitting 101 for RTP-NTE) on the same
>> call leg.
>>
>>
>>
>> Hope this helps
>>
>>
>>
>> - Dan
>>
>> end attach-
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Ed Leatherman
>> *Sent:* Monday, July 18, 2016 3:10 PM
>> *To:* Cisco VOIP 
>> *Subject:* [cisco-voip] DTMF interworking on CUBE - asymmetric payloads
>>
>>
>>
>> I'm trying to get my head wrapped around some DTMF interworking
>>  features...
>>
>>
>>
>> I have this setup:
>>
>>
>>
>> UCM -- CUBE --- 3rd party system
>>
>>
>>
>> For both call legs through CUBE I'm advertising kpml and rtp-nte for
>> dtmf-relay
>>
>>
>>
>> The 3rd party sometimes sends me rtp payload type 101 for nte's, and no
>> kpml, and things work (as a bonus I observed CUBE correctly interworking
>> the nte's from the pbx into KPML, so uccx didn't break).
>>
>> Sometimes they send type 98 and no kpml, and things don't work.
>>
>>
>>
>> I'm trying to understand what is happening and what feature should fix it
>> (without breaking other things)
>>
>>
>>
>> Assumption:
>>
>> "dtmf-relay rtp-nte kpml" is telling CUBE to offer/accept rtp type 101
>> only for nte. I observe that CUBE negotiates KPML only for the associated
>> call leg back to UCM and doesn't bother with rtp-nte, so its just like any
>> other codec that CUBE doesn't care about.
>>
>>
>>
>> So.. if third party system ONLY sent me dtmf-relay with payload type 98,
>> could I just set the rtp payload type for this to 98 on the inbound dial
>> peer? would CUBE then correctly switch these up to 101 headed back to UCM?
>>
>>
>>
>> How can I (or can I at all) make this work in my particular case were I
>> could receive both?
>>
>> I see "asymmetric payload dtmf" thrown about as a possible solution, but
>> I'm having trouble understanding what it actually does. It sounds like it
>> passes these payload types through CUBE, so UCM could be getting rtp
>> payload type 98 - it knows what to do with it? It seems like then CUBE
>> wouldn't be able to translate things to KPML this way...
>>
>>
>>
>> I'm reading
>> 

Re: [cisco-voip] UCCX 11.0.1 bug watch

[Justin Steinberg]

Why is there still not a COP file for CSCux33949?   This has been a known
issue for so long it should not be left up to customers to try the 11.0,
run into a bug, and then open a TAC case.   The BU has known about this for
so long, please release a COP file or SU to fix this so this doesn't keep
happening to customers.

On Wed, Jun 22, 2016 at 10:00 PM, Abhiram Kramadhati (akramadh) <
akram...@cisco.com> wrote:

> Hi Ryan,
>
> For CSCux33949: this is always encountered with enhanced license, but if
> you have premium then the trigger for this is not very common. The original
> design for LD was such that the SubsystemRouteAndQueue is responsible for
> sending heartbeats to the JMS topics (where the LD streams are published).
> However, with Enhanced license this subsystem is disabled; and hence this
> issue is triggered every time.
>
> The change in design is that the heartbeat responsibility is now moved to
> the RmCm subsystem which is present in all license packages. This fix is
> now available in 11.0(1).ES5, so the recommendation would be to install
> that so that the latest bug fixes and the design changes are covered. If
> you have any other defects that are affecting your deployment, feel free to
> unicast and I can take a look. Thanks.
>
> Regards,
> Abhiram Kramadhati
> Technical Solutions Manager, CCBU
> CCIE Collaboration # 40065
>
> From: cisco-voip  on behalf of Ryan
> Huff 
> Date: Thursday, 23 June 2016 at 11:44 AM
> To: cisco voip 
> Subject: [cisco-voip] UCCX 11.0.1 bug watch
>
> Hello CC brothers and sisters,
>
> A bit back I emailed in regarding CSCup71611
>  which is a bug
> that in some instances, prevents creating an initial Call Control group
> after a new install / upgrade. It has been proven that this bug is not a
> 100% hit every time but does exist in the 11.0.1 code base (I hit it in
> 11.0.1), although previously thought to have been corrected in the 10.6.x
> base.
>
> Yesterday, I also hit CSCux33949
>  in the 11.0.1
> code base which impacts the Socket.IO service and creates intermittent
> session timeouts in the LiveData service. I believe, if I'm not mistaken,
> our own Anthony Holloway has mentioned this bug before. The TAC agent I
> worked with stated that CSCux33949
>  is a 100% hit
> every install (but I am not completely sure that is correct).
>
> In both cases, the latter being the more involved of the two, it took
> rooting the box and either messing with PKIDs or uploading jar files to fix
> code (I also fully realize that I signed my own death warrant by deploying
> 11.0.1 over the much more mature and stable 10.6(1)SU2).
>
> Just thought I would bump this back to the top, in case anyone is
> considering a UCCX11.0.1 deployment anytime soon.
>
> Thanks,
>
> Ryan
>
>
>
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Re: [cisco-voip] TSP client after CM cert added?

[Justin Steinberg]

i've seen CTI manager need a restart after new callmanager.pem certs even
for non-secure clusters

On Tue, Jun 21, 2016 at 1:24 PM, Dave Goodwin 
wrote:

> Scott, is the UCM cluster running in mixed mode? Do you know whether the
> security settings were configured during the TSP installation the first
> time? Here is some reading that explains the process, if you feel this may
> apply to you.
> *http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/10_0_1/secugd/CUCM_BK_C68276B4_00_cucm-security-guide-100/CUCM_BK_C68276B4_00_cucm-security-guide-100_chapter_010111.html#CUCM_RF_A85B4283_00
> *
>
> I do know that for other endpoints, after a CallManager.pem certificate
> gets changed, it's required to update the CTL file and then restart Cisco
> TFTP so the endpoints can verify if the new certificate is to be trusted.
>
> -Dave
>
> On Tue, Jun 21, 2016 at 12:17 PM, Scott Voll  wrote:
>
>> I installed CA certs on my Voice cluster last night.  All went pretty
>> good.
>>
>> But today I have an app that runs the TSP plugin, and it's giving me an
>> authentication issue.  Any idea how the TSP plugin would react to the new
>> Certs on CM?
>>
>> TIA
>>
>> Scott
>>
>> CM 10.5.2
>>
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Re: [cisco-voip] Voice Dialing Misdirects

[Justin Steinberg]

make sure the codec is G711 and not G729

On Mon, Jun 6, 2016 at 5:23 PM, Bill Talley  wrote:

> Isn't there a setting to play all matches? I don't recall for sure.  At
> least the caller could pick on or the other.
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> > On Jun 6, 2016, at 3:38 PM, David Zhars  wrote:
> >
> > Correct, it is.  And neither use has any alternate names or extensions
> in their Unity setup.
> >
> > I am truly stumped!!!
> >
> > Sent from my iPad
> >
> >> On Jun 6, 2016, at 2:52 PM, Bill Talley  wrote:
> >>
> >> I believe this is probably a speech enabled directory handler.
> >>
> >> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
> >>
> >>> On Jun 6, 2016, at 1:13 PM, David Zhars  wrote:
> >>>
> >>> We have a greeting when people call the main number, and if you press
> 9, you can speak the name of the person you want to connect to.
> >>>
> >>> This oddity is happening:
> >>>
> >>> I speak "John Smith" who is extension 1305, and the system waits, then
> says "transferring to Jane Doe" who is extension 1306.
> >>>
> >>> I can't seem to figure out where this is off...
> >>> Unity and UCM 8
> >>>
> >>> Thanks
> >>>
> >>> Dave
> >>> ___
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Re: [cisco-voip] LDAP, Sync, Filters and CUCM

[Justin Steinberg]

CUCM doesn't delete the users when they are marked inactive.   you can fix
the LDAP agreement, resync and get them back if you get it fixed before the
garbage cycle clears them out.

this is really an issue for the UCCX team, they should handle user changes
in CUCM more gracefully.  This is also the case for situations where the
username changes.  If the account changes in the LDAP directory, CUCM will
usually see the change and update the username.  however, UCCX doesn't
track the same way and will delete the agent's old account and create a new
account for the new username.  the new UCCX account will have no skills, no
team, no associated historical data, etc.I haven't tested this with
UCCX 11, but this is how it has been on UCCX when I last tested it.

On Tue, May 3, 2016 at 3:04 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> This is related to my post I just made on UCCX and LDAP via CUCM.
>
> I also just found out that a CUCM with an already synced user database
> behaves in the following two ways:
>
>1. If you modify the filter such that it matches 0 records, the sync
>doesn't happen at all.  No users are marked as Inactive, no users are
>pulled in, and no users are updated
>
>You will see this in the DirSync log
>Dirsync synched zero users. Please verify the custom LDAP filter
>configured for this agreement
>
>2. However, if you modify the filter such that it matches a single
>record, the sync does happen.  All of the non-matched users will become
>Inactive.
>
>You will see this in the DirSync log (the value 1660 will vary by
>scenario)
>DSDBInterface.setUserInactive Found 1660 users in database needing
>update
>
> For #1, it seems like this might be a protection mechanism, preventing you
> from destroying your entire corporate directory.  Because, recall that EM,
> Jabber, Finesse, etc., all require your account to be Active Synced in
> order to authenticate you; therefore, making 1660 people go Inactive will
> have a large impact.  Or perhaps it was a coding error, and they should
> have made all users go Inactive?
>
> For #2, if we're thinking #1 could be a protective mode, then wouldn't
> 100% user loss be just as bad as 99%?  Perhaps the protection mechanism
> should look for a smaller percentage drop in Active users and prohibit an
> LDAP update at that time and display a warning on the page (I.e., Like the
> last known good backup now shows up on the About page).
>
> What do you think?  Have you seen this before?  Has it bit you?  Am I
> missing something obvious?  Let me know.  Thanks.
>
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Re: [cisco-voip] UCCX 10.6(1) SU1 & SU2 CAD Upgrade Warning

[Justin Steinberg]

Looks like the upgrade from 10.6(1)SU1 to 10.6(1)SU2 also requires a CAD
upgrade even though the compatibility matrix again says it doesn't.

Per comparability matrix:

10.6(1)su1 CAD version is 10.6.1.1057
10.6(1)su2 CAD version is 10.6.1.1057

My lab testing:

10.6(1)su1 CAD version is 10.6.1.1057
10.6(1)su2 CAD version is 10.6.1.2011

Justin

On Fri, Oct 2, 2015 at 1:22 AM, Abhiram Kramadhati (akramadh) <
akram...@cisco.com> wrote:

> Closing the loop on this.
>
> The CAD version for 10.6(1) is 10.6.1.95
> The CAD version for 10.6(1)SU1 is 10.6.1.1057
>
> The compatibility matrix for 10.6(1)SU1 has wrongly mentioned it as
> 10.6.1.95 and CSCuw51442 has been filed to correct this.
>
> As a side note, it would be best to plan for a client side upgrade for any
> server upgrade done. Please let me know if there are any questions, thank
> you.
>
> Regards,
> Abhiram Kramadhati
> Technical Solutions Manager, CBABU
> CCIE Collaboration # 40065
>
> From: Justin Steinberg <jsteinb...@gmail.com>
> Date: Friday, 25 September 2015 2:41 am
> To: Cisco VOIP <cisco-voip@puck.nether.net>
> Subject: [cisco-voip] UCCX 10.6(1) SU1 CAD Upgrade Warning
>
> The current UCCX compatibility matrix lists CAD 10.6.1.95 for both UCCX
> 10.6(1) and 10.6(1)SU1, implying there is no CAD upgrade forced after the
> SU1 upgrade.
>
> This isn't the case.   Upgrading 10.6(1) to 10.6(1)su1 does force a CAD
> upgrade.
>
> Hopefully this helps someone else.
>
> Justin
>
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Re: [cisco-voip] Additional historical reports available on CCO

[Justin Steinberg]

those reports have the temp tables issue.   they need to be imported using
the uccxhrc user.

Which makes me really question using these reports in production since the
QA on these must have been really minimal to document that they should be
imported with uccxhruser.

On Fri, Apr 8, 2016 at 11:14 AM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Update: I rebooted my UCCX to see if that helped, and now I cannot run
> the Reason Code Report by Agent Grouping report, as it fails with the same
> database error.  So, it got worse.
>
> On Fri, Apr 8, 2016 at 9:44 AM, Ed Leatherman 
> wrote:
>
>> Hmm i'll try the other ones - both reason code reports resulted in the
>> same errors. I tried the second one using CUIC as the data source just to
>> see if it made a difference. Thats the same type of error that i got
>> though. Trying to find out if TAC will support these or not also.
>>
>> On Fri, Apr 8, 2016 at 10:42 AM, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>>> Update:  After importing all reports, I can only run two of the five:
>>>
>>>1. All Agent Fields Report - GOOD
>>>2. Contact Service Queue Activity by Window Duration - Database
>>>status Failed
>>>3. CSQ All Fields Report - Database status Failed
>>>4. Reason Code Report by Agent Grouping - GOOD (This was
>>>coincidentally the first one I imported)
>>>5. Reason Code Report by Reason Code Grouping -  - Database status
>>>Failed
>>>
>>> As one example of the error output from a failed run is:
>>>
>>> *Error information:*
>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDbException:
>>> DbException: CuicDataSourceServiceManagerImpl.getDataSet() { Nested
>>> SQLException; SQLState: IX000 Vendor code: -313 Message: Not owner of
>>> table. } at
>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDataSourceServiceManagerImpl.getDataSetBuilder(CuicDataSourceServiceManagerImpl.java:952)
>>> at
>>> com.cisco.ccbu.cuic.businesslogic.engine.CuicReportEngineWorker.executeQueryUsingDatasetBuilder(CuicReportEngineWorker.java:80)
>>> at
>>> com.cisco.ccbu.cuic.businesslogic.engine.CuicReportEngineWorker.runReport(CuicReportEngineWorker.java:37)
>>> at com.cisco.ccbu.cuic.businesslogic.enginebase.Worker.run(Worker.java:329)
>>> at
>>> com.cisco.ccbu.infra.threads.InstrumentedRunnable.run(InstrumentedRunnable.java:92)
>>> at
>>> java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1145)
>>> at
>>> java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:615)
>>> at java.lang.Thread.run(Thread.java:724) at
>>> com.cisco.ccbu.infra.threads.ThreadPoolThread.run(ThreadPoolThread.java:164)
>>> Caused by: java.sql.SQLException: Not owner of table. at
>>> com.informix.jdbc.IfxSqli.a(IfxSqli.java:3643) at
>>> com.informix.jdbc.IfxSqli.E(IfxSqli.java:3974) at
>>> com.informix.jdbc.IfxSqli.dispatchMsg(IfxSqli.java:2695) at
>>> com.informix.jdbc.IfxSqli.receiveMessage(IfxSqli.java:2611) at
>>> com.informix.jdbc.IfxSqli.a(IfxSqli.java:1830) at
>>> com.informix.jdbc.IfxSqli.executeStatementQuery(IfxSqli.java:1768) at
>>> com.informix.jdbc.IfxSqli.executeStatementQuery(IfxSqli.java:1699) at
>>> com.informix.jdbc.IfxResultSet.a(IfxResultSet.java:210) at
>>> com.informix.jdbc.IfxStatement.executeQueryImpl(IfxStatement.java:1237) at
>>> com.informix.jdbc.IfxPreparedStatement.executeQuery(IfxPreparedStatement.java:401)
>>> at
>>> com.informix.jdbc.IfxCallableStatement.executeQuery(IfxCallableStatement.java:241)
>>> at
>>> com.mchange.v2.c3p0.impl.NewProxyCallableStatement.executeQuery(NewProxyCallableStatement.java:2131)
>>> at
>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDataSourceServiceManagerImpl.executeAndLoadDataset(CuicDataSourceServiceManagerImpl.java:1072)
>>> at
>>> com.cisco.ccbu.cuic.businesslogic.datasource.CuicDataSourceServiceManagerImpl.getDataSetBuilder(CuicDataSourceServiceManagerImpl.java:938)
>>> ... 8 more Caused by: java.sql.SQLException at
>>> com.informix.util.IfxErrMsg.getSQLException(IfxErrMsg.java:413) at
>>> com.informix.jdbc.IfxSqli.E(IfxSqli.java:3979) ... 20 more
>>>
>>> On Fri, Apr 8, 2016 at 9:30 AM, Anthony Holloway <
>>> avholloway+cisco-v...@gmail.com> wrote:
>>>
 I'm only just seeing this now for the first time, and so, I just tried
 to import one report, and so far it worked for me.  I'll try the rest in a
 moment, but for now...

 Here's what I have, and what I did.

 My UCCX is 10.6.1.11001-31

1. Downloaded CCX_NEWREPORTS.zip from http://www.cisco.com/
2. Extracted CCX_NEWREPORTS.zip to folder CCX_NEWREPORTS
3. Opened the Import Guide MS Word document from within the ZIP
archive and began reading the instructions to import the first report
4. Downloaded AGS Server Studio from
http://www.serverstudio.com/downloads/
5. Followed AGS instructions to install and register the product
6. Logged into UCCX App Admin 

Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting

[Justin Steinberg]

i haven't experienced this issue before.Care to share your stored
procedure ?

On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley  wrote:

> Hi all,
>
> Has anyone imported custom stored procedures into the db_cra for use in
> CUIC?  I'm having an issue getting a new stored proc loaded using either
> RazorSQL or AGS Server Studio.  From what I've read, I need to execute the
> stored proc; however, when I do that I get an error that uccxhruser doesn't
> have permission to execute the stored proc.  It seems I can't apply
> permissions to the uccxhruser account on the stored proc until the stored
> proc is loaded.   Would anyone have any tips on how I can go about getting
> the stored proc loaded to the database?
>
> Hopefully this makes sense to someone ;-)
>
> TIA,
> Bill
>
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Re: [cisco-voip] increasing VM virtual memory?

[Justin Steinberg]

I've increased the UCM/CUC 10.x VMs to 6 GB to prevent memory issues and
haven't had any problems.I keep the bug below folded on some paper
tucked away in my wallet just in case.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuo38244

*increase the threshold for LowAvailableVirtualMemory from 75% to 85%*
CSCuq75767

Description
*Symptom:*
A CUCM Publisher running version 10.5(1) experiencing
LowAvailableVirtualMemory alerts.

*Conditions:*
This is most commonly seen on Publisher nodes running version 10.5(1) with
the 1k or 2.5k OVA Templates.

*Workaround:*
Increase the RTMT LowAvailableVirtualMemory threshold from 75% to 85%.

or

Shut down the virtual machine, increase the virtual machine memory capacity
by 2GB. This increase will be required for CUCM 11.0(1)

*Further Problem Description:*
This is not a leak or cause for concern if the VM Utilization should
stabilizes under 85%.

If Virtual Memory utilization increases above 90% there is most likely a
problem which should be investigated further

On Thu, Mar 10, 2016 at 9:53 AM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Nick,
>
> It's actually pretty common, when doing upgrades, that minimum RAM
> requirements go up, and you have to increase the allocated RAM and
> reservation.
>
> Example:
>
> CUCM 10x 2,500 OVA requires 4GB RAM
> 
> CUCM 11x 2,500 OVA requires 6GB RAM
> 
>
> Therefore, an upgrade on a 2,500 OVA from 10x to 11x causes a RAM increase
>
> And it's stated as having changed in the OVA for v11x Read Me
> 
>
> The following are changes from the OVA in the previous (CUCM 10.x) release:
>
> 1) vRAM size has been increased by 2GB for 11.0(1) and above releases.
>
> The procedure is also thinly covered in the OVA Read Me as well:
>
> Upgrades from previous release:
>
>   1) Power off the VM.
>
>   2) Modify the vRAM, CPU, reservations,and OS based on the changes 
> listed above via the vmware-viclient. Note, do not attempt to change the 
> number of disks or disk size.
>
>   3) Modify the Network Adapter.  See “Modifying the Network Adapter” 
> section.
>
> 4) Modify the VM Version if previously using VM Version 7. See 
> "Instructions to upgrade the virtual hardware version" section.
>
>   5) Save the changes
>
>   6) Power back on the VM
>
> I say all of that, not because you indicated that you are performing an
> upgrade, but simply to illustrate that there are some cases where you MUST
> change the RAM, and therefore it makes sense that changing RAM is trivial
> and supported.  Mostly...
>
> Because, a negative issue I've seen come of this, is a UCCX system where
> it was originally deployed for 100 Agents, and someone increased the RAM,
> and then the system started complaining that it was a 300 Agent system with
> insufficient resources (everything except RAM).  The system was working
> fine, mind you, it was just the annoying warning banner in AppAdmin.  So,
> get the warning to go away, that person just dropped the RAM back down to
> the original value.
>
> On Thu, Mar 10, 2016 at 8:35 AM, Barnett, Nick <
> nick.barn...@countryfinancial.com> wrote:
>
>> Are there any negative issues that arise from increasing the RAM on a
>> CUCM 10.0 or CUC 10.5 VM?  I haven’t ever seen anyone talk about it. I
>> assume that is because there is typically an underlying issue that should
>> be resolved rather than just increasing RAM. Does anyone have any input on
>> this?
>>
>>
>>
>> Thanks,
>>
>> Nick
>>
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>>
>>
>
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Re: [cisco-voip] UCCX 11 Finesse HAoW Island Mode

[Justin Steinberg]

This isn't the full answer you're looking for, but I'll still throw it out
there...

I know LDAP enabled agents can login to Finesse when the UCM publisher is
down as that happened to me last week.  The UCM LDAP auth component doesn't
rely on the Dirsync service, so the UCM LDAP auth runs on all UCM nodes.

I had a UCS blade failure that took down the UCM pub, but the UCCX pub and
all the primary AD servers were still online for the UCM subs to
authenticate.

On Fri, Feb 5, 2016 at 4:17 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> UCCXers,
>
> I'm trying to avoid spinning up an entire lab to answer a simple question
> that the SRND is glossing over.  "Can Agents login to Finesse on the Island
> Mode side opposite the CUCM Publisher if using LDAP Authentication?"
>
> What the SRND has to say about failover and Island Mode:
>
>
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_11_0/design/guide/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11_appendix_0100.html#UCCX_RF_W5EB2ACC_00
>
> A little further down in the SRND it talks about Finesse in Island Mode,
> and it states that Agents can work on both sides, but it does not state, if
> that is: A) for only already logged in Agents, or B) for CUCM local
> authentication or LDAP authentication or otherwise.
>
>
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_11_0/design/guide/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11_appendix_0100.html#UCCX_RF_F3A11E07_00
>
> This is a very shallow description on what I consider to be a very deep
> topic, so I'm asking here for real world experience.
>
> Assume that we have two Data Centers: DC-A and DC-B.
>
> *DC-A Contains:*
>
>- LDAP Server A
>- CUCM Publisher
>- UCCX Publisher (Currently Engine Master)
>- Agents
>
>
> *DC-B Contains*
>
>- LDAP Server B
>- CUCM Subscriber
>- UCCX Subscriber (Currently Engine Slave)
>- Agents
>
>
> *Assumed Config*
>
>- Call flows are internal, no voice gateways to worry about
>- CUCM LDAP Auth config is pointing at LDAP Server A first and LDAP
>Server B second
>- UCCX Publisher AXL/JTAPI config is pointing at CUCM Pub first and
>CUCM Sub second
>- UCCX Subscriber AXL/JTAPI config is pointing at CUCM Sub first and
>CUCM Pub second
>- UCCX CTI Route Points have Device Pool with CMG pointing at CUCM Pub
>first and CUCM Sub second
>- UCCX Publisher CTI Ports have Device Pool with CMG pointing at CUCM
>Pub first and CUCM Sub second
>- UCCX Subscriber CTI Ports have Device Pool with CMG pointing at CUCM
>Sub first and CUCM Pub second
>
>
> *Question*
>
>1. Can an Agent in DC-B, who was not logged in before Island Mode
>happened, now log in, while in Island mode?  Does CUCM's authentication
>method change the answer?  E.g., LDAP integrated user versus local user.
>
> Thank you.
>
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Re: [cisco-voip] UCCX Bug

[Justin Steinberg]

are you running 10.6(1)su1 ?I have seen this on 10.6 base on two
seperate installs.  it's really ugly, all Finesse Admin team specific data
is lost.  the biggest problem is if you customize XML and then lose it and
have to try and figure out how to rebuild it.   Since i had this the first
time I make it a point to save the XML desktop layout on my PC for each
team in the event I have to restore.

The first time I had this problem was probably close to a year ago i'd
guess, but i personally haven't seen it on 10.(6)1su1.

On Tue, Dec 29, 2015 at 9:45 AM, Ed Leatherman 
wrote:

> Just hit this fun bug, thought I would pass it along. CSCuv24965
>
> Running latest uccx 10.6, HA, enhanced.
>
> TAC said it currently had a lot of attention with the developers, no fix
> yet and no idea yet what triggers it. Workaround is restarting tomcat and
> re-associating all of your reason codes and other team related settings
> with the correct teams.
>
> Happy Holidays! :)
>
> --
> Ed Leatherman
>
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[cisco-voip] UCM Hunt Group Reporting

[Justin Steinberg]

Has anyone used any CDR package they like with UCM environments with heavy
hunt group usage ?

This is a retail environment with centralized operator group handling
inbound main number calls, and transferring calls to various hunt groups
throughout the retail sites.   Essentially we are using UCM hunt groups
instead of a call center solution; the call flow/inbound caller experience
is fine with the UCM hunt groups we just are lacking on the reporting.

I'm looking for an off the shelf CDR package that has in depth hunt group
reports along with the option to create custom reports in a WYSIWYG style
interface.

Today we have about 180 hunt pilots, some of those hunt pilots are for the
front line operator groups, & some of those are for the back line hunt
groups that take transfers from the operators.  We would like to have a
single report for the hunt groups, but be able to group the operator hunt
groups into a separate table than the backline groups.

We would also like to be able to see a break down of how many calls are
transferred by the operators: to back line hunt groups, to people's
individual extensions, etc.

We've tried some off the shelf packages, and we can get much of the above
data but we are finding that we have to run multiple reports and then
combine them in excel to get the executive report we need to pass to upper
management.

I don't believe a call center solution is going to work in this
environment, primarily because many of the people taking the backline calls
are working in retail locations and mixing their time between walk-in
customers and phone calls.   There are multiple phones throughout the
retail store counters and a single person might answer any given call on
multiple phones depending on where they are in the store at any given time.

If anyone has any CDR packages they've used that could meet some of these
requirements I'd appreciate any recommendation.

Personally I think we may be going down the route of writing a custom CDR
solution based on the UCM CDR CSV files but want to look for any existing
solutions if they can meet the requirements.

Justin
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Re: [cisco-voip] CUCM Native Presence (BLF/SD) populating Directories / Call Lists

[Justin Steinberg]

BLFs don't generate missed call alerts.  only DNs. It might be helpful
to get a third phone (or IP Communicator) so you don't have to 'dial
itself', that might be making things a little confusing to understand the
various behaviors.

On Tue, Dec 1, 2015 at 8:32 PM, Jeffrey Girard  wrote:

> Nate –
>
> Thanks for the reply.
>
>
>
> Yes, see below
>
>
>
> .  Finally, I set the Enterprise Parameter “BLF for Call
> Lists” to enabled.
>
>
>
>
>
> *From:* NateCCIE [mailto:natec...@gmail.com]
> *Sent:* Tuesday, December 1, 2015 8:29 PM
> *To:* Jeffrey Girard; cisco-voip@puck.nether.net
> *Subject:* RE: [cisco-voip] CUCM Native Presence (BLF/SD) populating
> Directories / Call Lists
>
>
>
> Do you have the enterprise parameter “BLF for call lists” enabled?
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
> ] *On Behalf Of *Jeffrey Girard
> *Sent:* Tuesday, December 01, 2015 6:20 PM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] CUCM Native Presence (BLF/SD) populating
> Directories / Call Lists
>
>
>
> All –
>
> I have been lurking on this list for quite a while as I
> continue to hone my collaboration skills.
>
>
>
> CAVEAT:  This is not a production system and is not for
> any customer.  This is my lab as I continue to try to learn.
>
>
>
> CUCM v9
>
>
>
> Desired endstate:  Populate the Missed Calls with CUCM Native Presence /
> Speed Dial information.
>
>
>
> What I have configured:
>
>
>
> 2 phones, both 7962s.  Both have a BLF/SD pointing to the
> other.  One DN is 3001 and the other DN is 3002.  BLF/SDs work fine.
> Created a single partition that holds all the phones PT_Internal and a
> single CSS that holds the partition CSS_Internal.  On each of the 7962s, I
> set the line SUBSCRIBE CSS to CSS_Internal.  Finally, I set the Enterprise
> Parameter “BLF for Call Lists” to enabled.
>
>
>
> However, if I go offhook on 3002 and dial itself, I see
> the BLF light on 3001.  If I don’t answer the call and then I check the
> missed calls for 3001, I expect to see an entry for 3002.  However, nothing
> shows.  If I go offhook on 3002, call 3001, let it ring and do not answer,
> that missed call shows up as it should.
>
>
>
> My Directories are populated with information from when I
> park calls, directed parks, but not the missed call tied to the BLF.
>
>
>
> Is this the expected behavior?  Am I expecting to see
> something (this missed call from 3002) that should not be there by design?
>
>
>
> Thanks,
>
> Jeff
>
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Re: [cisco-voip] DRS Backups 10.5

[Justin Steinberg]

solarwinds does have a lightweight free SFTP client.  It usually works
reliably if the Windows server it runs on is 32 bit and if the Cisco UC
apps are relatively new.  Obviously it still isn't supported by TAC but
when it works it works.

On Thu, Oct 8, 2015 at 12:03 PM, Brian Meade  wrote:

> FreeFTPD historically had a 1GB limitation but that's been long-gone.
> Documentation was just never updated.
>
> On Thu, Oct 8, 2015 at 11:01 AM, Jason Aarons (AM) <
> jason.aar...@dimensiondata.com> 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
>>
>> [image: twitter] 
>>
>> [image: chat][image: Phone] <+13142121814>[image: video]
>>
>> “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 <%28519%29%20741-2200%20x%207000>*
>>
>>
>>
>>
>>
>>
>>
>> *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] Background url

[Justin Steinberg]

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) <
louis.koekem...@dimensiondata.com> 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:
>
>  URL="TFTP:Desktops/800x480x24/Background.png"/>
>
>
>
> 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 –
>
> [image: Description: Description: 21875_DD_NI_CC_Signature_v-1]
> 
>
>
>
>
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
> 
>
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Re: [cisco-voip] Understanding a Defect's Affected Versions

[Justin Steinberg]

I agree with that.  It's too hard to know how to search the bug toolkit for
fixes in a certain version.
On Oct 5, 2015 11:54 PM, "Erick Bergquist"  wrote:

> I'm also not a fan of the newer release notes not including a list of
> the Resolved Bugs, but a link to bug search tool...
>
> That leaves it up to us to find what bugs were fixed or hoping bug
> search tool returns them all, plus not a nice list/summary to glance
> through.
>
> On Mon, Oct 5, 2015 at 10:47 PM, Brian Meade  wrote:
> > 10.5.2.12028-1 is an Engineering Special which uses a different numbering
> > scheme.  I thought the ReadMe used to show what ES the SU was built off
> of
> > but having trouble finding it.
> >
> > SU2/SU2a were most likely built off of older engineering specials than
> > 10.5.2.12028-1.
> >
> > The higher release thing really only works in the case of published
> versions
> > on cisco.com.
> >
> > On Mon, Oct 5, 2015 at 11:34 PM, Erick Bergquist 
> wrote:
> >>
> >> Some bugs, like CSCuu58142 effecting single number reach doesn't seem
> >> to follow higher versions contain the fix methodology.
> >>
> >> Bug toolkit says this is fixed in 10.5.2.12028-1 but 10.5.2 SU2, SU2a
> >> (10.5.2.12900 and 10.5.2.12901) don't contain the bug fix per TAC and
> >> going over the release notes for SU2, SU2a.
> >>
> >> I need to use the 10.5.2.12028-1 ES or latest ES 10.5.2.13039-1.
> >> Currently debating which route I'm going to go or wait out for SU3 or
> >> until we upgrade to 11.x.  This SNR bug is effecting some users about
> >> every 1-2 months.  Workaround is to disable SNR on their remote
> >> destination profile and re-enable it.
> >>
> >>
> >>
> >> On Tue, Sep 29, 2015 at 2:03 PM, Ryan Ratliff (rratliff)
> >>  wrote:
> >> > it's up to the discretion of the bug author.  <--
> >> >
> >> >
> >> > This means it’s accuracy varies greatly by product and even bug
> author.
> >> > For
> >> > UCM you should always assume you are vulnerable if the fixed-in
> version
> >> > is
> >> > higher than what you are currently running unless the bug description
> >> > clearly states otherwise or the feature impacted by the bug doesn’t
> >> > exist in
> >> > your version.
> >> >
> >> > -Ryan
> >> >
> >> > On Sep 29, 2015, at 2:25 PM, Anthony Holloway
> >> >  wrote:
> >> >
> >> > In reference to this defect:
> >> >
> >> > https://tools.cisco.com/bugsearch/bug/CSCuv45722
> >> >
> >> > Can you help me understand what this means as far as all affected
> >> > versions?
> >> >
> >> > On the surface, it would appear that it's only affecting 9.1(2).
> >> > However,
> >> > with a fixed in version being way out in 11.5, that would also
> indicate
> >> > to
> >> > me that an upgrade to 10.5(2)SU2a, as an example, would not fix this
> >> > issue.
> >> >
> >> > Does Cisco imply all versions affected between the listed affected
> >> > versions
> >> > and the fixed in version?  Or, should this defect list all affected
> >> > versions?
> >> >
> >> > I cannot recall what I've heard about this in the past.  I'm almost
> >> > guessing
> >> > there's no exact science to it, and it's up to the discretion of the
> bug
> >> > author.
> >> >
> >> > Thanks for your help.
> >> > ___
> >> > cisco-voip mailing list
> >> > cisco-voip@puck.nether.net
> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >> >
> >> > ___
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> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >> >
> >> ___
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> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
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[cisco-voip] UCCX 10.6(1) SU1 CAD Upgrade Warning

[Justin Steinberg]

The current UCCX compatibility matrix lists CAD 10.6.1.95 for both UCCX
10.6(1) and 10.6(1)SU1, implying there is no CAD upgrade forced after the
SU1 upgrade.

This isn't the case.   Upgrading 10.6(1) to 10.6(1)su1 does force a CAD
upgrade.

Hopefully this helps someone else.

Justin
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Re: [cisco-voip] MRA (Collaboration Edge) Intrusion Protection

[Justin Steinberg]

There are some settings on the Expressway regarding the number of auth
attempts, etc.  have you tried to increase those to see if that makes any
difference ?

On Tue, Sep 15, 2015 at 10:45 AM, Ryan Huff  wrote:

> 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 
> 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] Any way to invoke / force MTP for CSF devices using device mobility?

[Justin Steinberg]

I think they need to reconsider their policy against preventing VPN users
from communicating with each other.  Perhaps they can and an ACL to block
undesired traffic and allow the RTP to pass directly between the VPN
clients.

On Mon, Aug 24, 2015 at 9:49 PM, Tong, Dana dana.t...@team.telstra.com
wrote:

 Hi Brian,



 The customer had advised me that they were reluctant to enable MTP for
 some 6000 endpoints. They’re also geographically spread out across 6
 continents.



 Cheers

 Dana





 *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
 Meade
 *Sent:* Tuesday, 25 August 2015 12:03 AM
 *To:* Tong, Dana dana.t...@team.telstra.com
 *Cc:* Cisco VOIP cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Any way to invoke / force MTP for CSF devices
 using device mobility?



 Is there any harm in just enabling it for all CSF devices?  You can easily
 add it to the Common Device Configuration and do a quick BAT to push out
 the change.



 On Sun, Aug 23, 2015 at 7:58 PM, Tong, Dana dana.t...@team.telstra.com
 wrote:

 Hi there,



 I have a customer who has VPN users but does not appear to have enabled IP
 connectivity between remote access users.

 Hence when they are on the VPN they cannot call each other with Jabber. Is
 there a way to use device mobility to invoke an MTP / TRP at all?



 The CSF device has a checkbox but does not appear to be related to Device
 Pool.



 MRA would probably work for them. But I think they would also have to
 implement some DNS filtering/ and ACL because they’d still be wanting to
 VPN at the same time.



 Cheers

 Dana


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Re: [cisco-voip] jabber directory lookup attributes (UDS)

[Justin Steinberg]

Yes you can just delete the ldap directory  authentication and rebuild it
with the new telephone number mapping.  the users wont disappear (even
though it will warn you they will). that should take care of the
inbound caller ID extension mapping as well.

On Wed, Aug 5, 2015 at 3:46 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

 I can’t remember when this was added, it might be in 9,  but I know in 10+
 you can temporarily convert everyone back to local users and then redo the
 sync. That would make sure no one got deleted. When and how you can do that
 depends on how heavily dependent you are on that ldap information.



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

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



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Lelio Fulgenzi
 *Sent:* Wednesday, August 5, 2015 3:40 PM
 *To:* Justin Steinberg jsteinb...@gmail.com
 *Cc:* cisco-voip@puck.nether.net

 *Subject:* Re: [cisco-voip] jabber directory lookup attributes (UDS)





 Thanks Justin,



 I'm gathering options, so that might definitely be a possibility. Not sure
 how I would handle that though, basically, having to create a new LDAP sync
 without deleting the existing users.



 I'm also trying to resolve how Jabber does a search on the UDS contacts
 with an inbound caller ID (extension). Apparently, that's the only time the
 directory lookup rules are used?



 ugh. this was supposed to be easy. :)



 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: *Justin Steinberg jsteinb...@gmail.com
 *To: *Lelio Fulgenzi le...@uoguelph.ca
 *Cc: *cisco-voip@puck.nether.net
 *Sent: *Wednesday, August 5, 2015 2:29:12 PM
 *Subject: *Re: [cisco-voip] jabber directory lookup attributes (UDS)



 someone correct my if I'm wrong, but I don't think Jabber will search any
 of the custom LDAP attributes as part of cucm dirsync.  I would
 recommend configuring your directory numbers in AD in the ipPhone field in
 the format Y.   then change your cucm dirsync so that cucm imports
 the ipPhone field into CUCM as the CUCM telephone number.that why you
 won't need any dial rules or lookup rules for this setup.



 On Wed, Aug 5, 2015 at 10:54 AM, Lelio Fulgenzi le...@uoguelph.ca wrote:



 Trying to find a way around the way telephone numbers are stored in our
 Active Directory so Jabber users can search by DN.



 Right now, we're using UDS so that way the experience is the same on
 premise and off-premise. CUCM is LDAP integrated with Active Directory.



 Unfortunately, our telephone numbers are stored as follows:
 +1-NPA-NXX- xY. (small x is a literal x, big X and Y are 0-9)



 The dashes and blank are ignored, but the little x is interpreted as a 9,
 so in Jabber, a directory search using the user name shows
 +1NPANXX9Y



 For the most part, I can build dial rules to work around this, dropping a
 whole bunch of digits before dialing so that way only the extension is
 dialed.



 But searching for extensions is gonna be almost impossible since I can't
 write a directory lookup rule to include the spaces, etc.



 I'd like to be able to use the other attributes mentioned here:




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_5/CJAB_BK_D6497E98_00_deployment-installation-guide-ciscojabber/configure_directory_integration.html#CJAB_RF_A590F694_00



 I'm just wondering if I simply use otherTelephone and otherTelephone
 in the custom LDAP synchronization fields. Will it only take the first one?



 Anyone else play with this?



 Lelio





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



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




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Re: [cisco-voip] Cisco Finesse Phone Book

[Justin Steinberg]

No there isn't a way.  Hopefully this is added to the roadmap, since it
would be easy for finesse to utilize the UDS API.

There are likely 3rd party gadgets, but this should be built in.
On Jul 24, 2015 2:11 PM, John J bcch...@gmail.com wrote:

 Anyone know if there's a way to populate Finesse's Phone Book with CUCM or
 LDAP directories to search?

 Thanks.

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Re: [cisco-voip] Digicert Wildcard certificates

[Justin Steinberg]

While we are on the topic of certs, has anyone had issues with certain CAs
not allowing top level domain as a SAN (e.g. cisco.com) ?

GoDaddy would complain in the UI that you shouldn't have a top level domain
as a SAN but would still sign the cert.   I'm having a problem know with
Internet2/Incommon where it won't let me put a top level domain in the cert
as a SAN.  It just won't take the CSR.

Justin

On Tue, Jul 21, 2015 at 8:16 AM, NateCCIE natec...@gmail.com wrote:

 I think it’s 15 SANS plus *.domain.com and domain.com



 Pricing is at https://www.digicert.com/wildcard-ssl-certificates.htm





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Anthony Holloway
 *Sent:* Monday, July 20, 2015 11:49 PM
 *To:* Charles Goldsmith; Ian Anderson
 *Cc:* Cisco VOIP

 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates



 That's great to hear about digicert. I just went through a rough time with
 Comodo trying to get multiserver certs and my CNAMEs in the SAN field. How
 many SAN entries does digicert limit you to and at what price per year?



 On Mon, Jul 20, 2015 at 11:19 AM Charles Goldsmith wo...@justfamily.org
 wrote:

 One thing of note, Digicert works very well with all of our UC apps with
 their UC certificate.  Add all of your server names as SAN's, as well as
 the domain name, and just duplicate the certificate for each app, changing
 the CN.  It works well and also Digicert has great support.



 On Sun, Jul 19, 2015 at 4:27 AM, Ian Anderson i...@andersoi.co.uk wrote:

 Hi Nate,



 I think that the concern of using wildcards generaly comes from the
 security and compliance folks in that if the private key of any of the
 servers was to be compromised then the resulting public and private keys
 could be used to impersonate any subdomain, e.g e-payments.domain.com..



 That said, as long as the customer is aware of the risk then the digicert
 is a fantastic option, although a lot of these issues go away in 10.5.



 The only app I've had it completely throw a wobble on so far is UCCX 9.0
 as this was checking the CN on certificate upload and didn't like * even
 though the server name as in the SAN.



 Cheers



 Ian



 On 16 July 2015 at 02:35, NateCCIE natec...@gmail.com wrote:

 Most of the time wildcard certs mean you have a CSR and a private key
 generated by something, and then you upload the private key and the public
 key to lots of servers.  The application would need to be able to upload a
 private key and not require its own CSR.



 Cucm, unity cxn, uccx, do not support uploading a private key.



 Expressway, I think conductor do allow you to upload a private key.



 But what makes digicert really cool is you can buy the wildcard cert, then
 you keep reissuing a new certificate from that one purchase.



 You can do this from what I understand an unlimited times.



 There may be other CAs that do this.  I saw one the seemed like it was
 going to work, but since the CSR did not include the * as a SAN, they would
 not issue the cert.



 Digicert with the Willard includes the *.domain.com and domain.com SANs
 automatically, and you can specify about 15 other SANs for each CSR/cert.



 So cucm and the other apps are happy because the cert was generated using
 its own CSR.



 Using these certs, I had one TAC case where cucm balked at the cert, but I
 could upload the cluster wide tomcat SAN cert via imp. This turned out to
 be a problem with the domain casing not matching between all of the servers
 and the cert. always use domain.com and not DOMain.com and life is happy.



 I am not affiliated with digicert other than they are here in Utah also.
 It just makes life really easy to tell the customer to buy this one cert
 and O I can make all of the Cisco UC/jabber cert errors go away!



 Ps. Has anyone figured out what to do with conductor wanting IP address in
 the SAN?

 Sent from my iPhone


 On Jul 15, 2015, at 10:42 AM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 I'm a little confused here.  According to this article:
 http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/115957-high-level-view-ca-00.html#wildcard,
 and this defect ID: https://tools.cisco.com/bugsearch/bug/CSCta14114/,
 wild card certs are not supported.  Are we talking about the same thing
 here?



 On Wed, Jul 15, 2015 at 10:08 AM Eric Pedersen peders...@bennettjones.com
 wrote:

 Digicert lets you put your domain and subdomains of any level as SANs.
 It’s great! They even generated a duplicate certificate for me with a
 different root CA that was supported with WebEx enabled Telepresence. We
 use their wildcard certificates on all of our UC servers.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Heim, Dennis
 *Sent:* 15 July 2015 8:28 AM
 *To:* Ian Anderson; NateCCIE; Cisco VOIP


 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates



 I’ve found the 

Re: [cisco-voip] Digicert Wildcard certificates

[Justin Steinberg]

Ya sorry I meant the parent domain.

The issue ended up being that the  Incommon wasn't setup right.   Their 800
tech support fixed it in like 40 seconds which was pretty cool.

I believe the 10.5 systems add the parent domain, or maybe it is just
Multiserver certs.

Justin
Justin,

TLDs are like .com, .net, .org , etc.  I think you meant parent domain.

Also, is that a feature of the multiserver cert, because I don't see CER
for example putting the parent domain in the CSR.

On Tue, Jul 21, 2015 at 10:24 AM Justin Steinberg jsteinb...@gmail.com
wrote:

 While we are on the topic of certs, has anyone had issues with certain CAs
 not allowing top level domain as a SAN (e.g. cisco.com) ?

 GoDaddy would complain in the UI that you shouldn't have a top level
 domain as a SAN but would still sign the cert.   I'm having a problem know
 with Internet2/Incommon where it won't let me put a top level domain in the
 cert as a SAN.  It just won't take the CSR.

 Justin

 On Tue, Jul 21, 2015 at 8:16 AM, NateCCIE natec...@gmail.com wrote:

 I think it’s 15 SANS plus *.domain.com and domain.com



 Pricing is at https://www.digicert.com/wildcard-ssl-certificates.htm





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Anthony Holloway
 *Sent:* Monday, July 20, 2015 11:49 PM
 *To:* Charles Goldsmith; Ian Anderson
 *Cc:* Cisco VOIP

 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates



 That's great to hear about digicert. I just went through a rough time
 with Comodo trying to get multiserver certs and my CNAMEs in the SAN field.
 How many SAN entries does digicert limit you to and at what price per year?



 On Mon, Jul 20, 2015 at 11:19 AM Charles Goldsmith wo...@justfamily.org
 wrote:

 One thing of note, Digicert works very well with all of our UC apps with
 their UC certificate.  Add all of your server names as SAN's, as well as
 the domain name, and just duplicate the certificate for each app, changing
 the CN.  It works well and also Digicert has great support.



 On Sun, Jul 19, 2015 at 4:27 AM, Ian Anderson i...@andersoi.co.uk wrote:

 Hi Nate,



 I think that the concern of using wildcards generaly comes from the
 security and compliance folks in that if the private key of any of the
 servers was to be compromised then the resulting public and private keys
 could be used to impersonate any subdomain, e.g e-payments.domain.com..



 That said, as long as the customer is aware of the risk then the digicert
 is a fantastic option, although a lot of these issues go away in 10.5.



 The only app I've had it completely throw a wobble on so far is UCCX 9.0
 as this was checking the CN on certificate upload and didn't like * even
 though the server name as in the SAN.



 Cheers



 Ian



 On 16 July 2015 at 02:35, NateCCIE natec...@gmail.com wrote:

 Most of the time wildcard certs mean you have a CSR and a private key
 generated by something, and then you upload the private key and the public
 key to lots of servers.  The application would need to be able to upload a
 private key and not require its own CSR.



 Cucm, unity cxn, uccx, do not support uploading a private key.



 Expressway, I think conductor do allow you to upload a private key.



 But what makes digicert really cool is you can buy the wildcard cert,
 then you keep reissuing a new certificate from that one purchase.



 You can do this from what I understand an unlimited times.



 There may be other CAs that do this.  I saw one the seemed like it was
 going to work, but since the CSR did not include the * as a SAN, they would
 not issue the cert.



 Digicert with the Willard includes the *.domain.com and domain.com SANs
 automatically, and you can specify about 15 other SANs for each CSR/cert.



 So cucm and the other apps are happy because the cert was generated using
 its own CSR.



 Using these certs, I had one TAC case where cucm balked at the cert, but
 I could upload the cluster wide tomcat SAN cert via imp. This turned out
 to be a problem with the domain casing not matching between all of the
 servers and the cert. always use domain.com and not DOMain.com and life
 is happy.



 I am not affiliated with digicert other than they are here in Utah also.
 It just makes life really easy to tell the customer to buy this one cert
 and O I can make all of the Cisco UC/jabber cert errors go away!



 Ps. Has anyone figured out what to do with conductor wanting IP address
 in the SAN?

 Sent from my iPhone


 On Jul 15, 2015, at 10:42 AM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 I'm a little confused here.  According to this article:
 http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/115957-high-level-view-ca-00.html#wildcard,
 and this defect ID: https://tools.cisco.com/bugsearch/bug/CSCta14114/,
 wild card certs are not supported.  Are we talking about the same thing
 here?



 On Wed, Jul 15, 2015 at 10:08 AM Eric

Re: [cisco-voip] Digicert Wildcard certificates

[Justin Steinberg]

To Dennis' point you don't have to put DNS=mycollab.com in the SAN.   There
is an alternative to use DNS=collab-edge.mycollab.com

http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-5/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-5.pdf

[image: Inline image 1]

On Wed, Jul 15, 2015 at 2:16 PM, Heim, Dennis dennis.h...@wwt.com wrote:

  If you have not seen the Cisco Live session on collab security I would
 definitely recommend it. It had some good discussion on certificates. Based
 on that Wildcard certs will never be supported on CUCM and the like and are
 frowned upon within the security community.



 *Dennis Heim | Emerging Technology Architect (Collaboration)*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]

 “There is a fine line between Wrong and Visionary. Unfortunately, you have
 to be a visionary to see it. – Sheldon Cooper



 Click here to join me in my Collaboration Meeting Room
 https://wwt.webex.com/meet/dennis.heim



 *From:* Eric Pedersen [mailto:peders...@bennettjones.com]
 *Sent:* Wednesday, July 15, 2015 12:51 PM
 *To:* Anthony Holloway; Heim, Dennis; Ian Anderson; NateCCIE; Cisco VOIP
 *Subject:* RE: [cisco-voip] Digicert Wildcard certificates



 Good point. I spoke too soon: we use wildcard certificates on VCS-E and
 WebEx Meeting Server only. IIRC VCS officially doesn’t support wildcard
 certificates either but everything seems to work provided the hostnames are
 configured as SANs. CUCM might be the same with the multi-server
 certificate but I haven’t tried.



 *From:* Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com
 avholloway+cisco-v...@gmail.com]
 *Sent:* 15 July 2015 10:43 AM
 *To:* Eric Pedersen; Heim, Dennis; Ian Anderson; NateCCIE; Cisco VOIP
 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates



 I'm a little confused here.  According to this article:
 http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/115957-high-level-view-ca-00.html#wildcard,
 and this defect ID: https://tools.cisco.com/bugsearch/bug/CSCta14114/,
 wild card certs are not supported.  Are we talking about the same thing
 here?



 On Wed, Jul 15, 2015 at 10:08 AM Eric Pedersen peders...@bennettjones.com
 wrote:

  Digicert lets you put your domain and subdomains of any level as SANs.
 It’s great! They even generated a duplicate certificate for me with a
 different root CA that was supported with WebEx enabled Telepresence. We
 use their wildcard certificates on all of our UC servers.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Heim, Dennis
 *Sent:* 15 July 2015 8:28 AM
 *To:* Ian Anderson; NateCCIE; Cisco VOIP


 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates



 I’ve found the hardest thing to find a cert providers that likes putting
 the domain as a san such as DNS=mycollab.com. Has anyone found any
 providers that are kosher with that? From one of the Cisco Live sessions, I
 was told this is needed for service discovery to function properly.



 *Dennis Heim | Emerging Technology Architect (Collaboration)*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]

 “There is a fine line between Wrong and Visionary. Unfortunately, you have
 to be a visionary to see it. – Sheldon Cooper



 Click here to join me in my Collaboration Meeting Room
 https://wwt.webex.com/meet/dennis.heim



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Ian Anderson


 *Sent:* Wednesday, July 15, 2015 10:18 AM
 *To:* NateCCIE; Cisco VOIP
 *Subject:* Re: [cisco-voip] Digicert Wildcard certificates





 On 15 July 2015 at 15:02, NateCCIE natec...@gmail.com wrote:

  Did you put all of your SANs in the digicert page?

 z

 I have this working on all of my expressway installs.

  Hi Nate,



 Thanks for the quick response, just for preservation in the archives for
 future posterity and confirmation that digicert seems fine despite the
 warnings in the manuals, it seemed I was running into 2 separate issues.



 1) I had uploaded the intermediate cert, but needed to manually download
 and upload the root CA

 2) That then got me past the TLS error, only to find that I had
 fat-fingered the hostname in the SAN field :-(



 Cheers



 Ian



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 subject matter. If this message has been received in error, please contact
 the sender and delete all copies. Like other forms of communication, e-mail
 communications may be vulnerable to interception by unauthorized parties.
 If you do not wish us to communicate with you by e-mail, please notify us
 at your earliest convenience. In the absence of such notification, your
 

[cisco-voip] Opinions on 10.5(2) SU2 ?

[Justin Steinberg]

Anyone running this version in production ?

I saw a COP file was released this week addressing three SIP bugs that
unregister the phones (CM crash?).

http://www.cisco.com/web/software/282204704/18582/ciscocm.FQDNwithDNS-v1.0.k3.readme.pdf

Any of the Cisco guys know whether this is going to cause a SU2 respin ?
 I'm running into BAT issues wiht 10.5(2) SU1 and would hate to apply
SU2+COP if there is a SU2a coming soon.

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

Re: [cisco-voip] Opinions on 10.5(2) SU2 ?

[Justin Steinberg]

Thanks Chris, I was suspecting that would be the case.  Any guess at an ETA
?

On Fri, Jul 10, 2015 at 4:24 PM, Chris Ward (chrward) chrw...@cisco.com
wrote:

  There is an SU2a en route. I would standby for it.



 +Chris

 TME - MediaSense and Unity Connection



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Justin Steinberg
 *Sent:* Friday, July 10, 2015 3:56 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Opinions on 10.5(2) SU2 ?



 Anyone running this version in production ?



 I saw a COP file was released this week addressing three SIP bugs that
 unregister the phones (CM crash?).




 http://www.cisco.com/web/software/282204704/18582/ciscocm.FQDNwithDNS-v1.0.k3.readme.pdf



 Any of the Cisco guys know whether this is going to cause a SU2 respin ?
  I'm running into BAT issues wiht 10.5(2) SU1 and would hate to apply
 SU2+COP if there is a SU2a coming soon.



 Justin

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

Re: [cisco-voip] Making a message appear anonymous/unknown via single inbox

[Justin Steinberg]

i'm not sure if this is the right way to handle this, but I wrote a SIP
normalization script to remove the info on the CUCM side before the call is
sent over to CUC and in initial testing this seems to do the job.

for the archives...

M = {}
 function M.outbound_INVITE(msg)
if msg:getHeader(Diversion)
then
local privacy = msg:getHeaderValueParameter(Remote-Party-ID, privacy)
if string.find(privacy, full)
then
msg:applyNumberMask(Remote-Party-ID, 0)
local rpid = msg:getHeader(Remote-Party-ID)
local rpiduri = string.match(rpid, (.+))
msg:modifyHeader(Remote-Party-ID, rpiduri)
 msg:applyNumberMask(P-Asserted-Identity, 0)
local pai = msg:getHeader(P-Asserted-Identity)
local paiuri = string.match(pai, (.+))
msg:modifyHeader(P-Asserted-Identity, paiuri)
end
if string.find(privacy, name)
then
local rpid = msg:getHeader(Remote-Party-ID)
local rpiduri = string.match(rpid, (.+))
msg:modifyHeader(Remote-Party-ID, rpiduri)
 local pai = msg:getHeader(P-Asserted-Identity)
local paiuri = string.match(pai, (.+))
msg:modifyHeader(P-Asserted-Identity, paiuri)
end
if string.find(privacy, uri)
then
msg:applyNumberMask(Remote-Party-ID, 0)
msg:applyNumberMask(P-Asserted-Identity, 0)
end
end
 end
return M

On Tue, Jun 23, 2015 at 11:47 AM, Justin Steinberg jsteinb...@gmail.com
wrote:

 in addition to the PSTN, they also want to have caller id blocked on calls
 on-net to other Cisco phones.   So we have certain CSS on Cisco phones that
 route calls through translation patterns to set the restricted CLID flag
 before the call rings the other Cisco phones.When we do this, the Cisco
 phone called party does see 'Private' but if they don't answer the call and
 it goes into voicemail then the voicemail message will still show the
 calling party number.

 On Tue, Jun 23, 2015 at 11:12 AM, Ryan Huff ryanh...@outlook.com wrote:

 Justin,

 For clarification, you have a user(s) that makes an outbound call from
 call manager to the pstn (via a sip trunk to an itsp?) And some of them
 want to block their Caller ID or mask it to anon?

 Thanks,

 Ryan


  Original Message 
 From: Justin Steinberg jsteinb...@gmail.com
 Sent: Tuesday, June 23, 2015 11:06 AM
 To: Ted Nugent tednugen...@gmail.com
 Subject: Re: [cisco-voip] Making a message appear anonymous/unknown via
 single inbox
 CC: Cisco VoIPoE List cisco-voip@puck.nether.net

 Apologies for resurrecting an old thread here, but I was curious if there
 was fixed in more recent versions of Unity Connection.   I'm trying this
 with 10.5(2)su1 with SIP trunk between UCM and UCXN and having the problem
 where UCXN ignores the restriction settings configured on CUCM.

 I'm thinking maybe I could make a LUA script on the UCM sip trunk to
 UCXN, but looking for an easier solution.

 In my situation, I have certain callers that want to have their CLID
 blocked.   So it is based on the placing the call, not the phone that
 receives the call and I think this prevents me from using the two options
 discussed here earlier.

 Justin

 On Thu, Jan 24, 2013 at 4:28 PM, Ted Nugent tednugen...@gmail.com
 wrote:

 Thanks Chris
 I like option 2 as well since they only have 24 ports as it stands. I'll
 give it a shot and let you know how it turns out. Thanks again!

 On Thu, Jan 24, 2013 at 3:16 PM, Chris Ward (chrward) chrw...@cisco.com
  wrote:

  Hi Ted,



 Seems like you are hitting some known limitations. The settings in CUCM
 for restricting Calling Party Information don’t really eliminate it, it
 seems to just mark it as “Restricted” so the information remains and CUC
 seems hell-bent on finding calling party information.



 I did find 2 solutions for you, both of which use SIP:



 1)  Setup an additional integration between CUCM and CUC that is
 meant only for this anonymous line. In the SIP trunk config, you have to
 uncheck the “Remote-Party-Id” under the “Call Routing Information”
 section and make sure “Calling Line ID Presentation” is set to
 Restricted under the “Outbound Calls” section. This will result in an
 “Unknown CallerID” message in Outlook.

 2)  Setup a Loopback SIP trunk for calls to this specific
 Anonymous DID. You only need one SIP trunk to point to CUCMs own IP
 address, it acts as the outbound and inbound SIP trunk in this scenario.
 Under the “Outbound Calls” section, you would need define a Calling Party
 Transformation CSS that would have access to a transformation pattern that
 matches all Calling parties (remembering that only calls to this anonymous
 DID are affected) and mask the calling party with some obscure mask like
  or you can put a . at the end of the pattern you match and then drop
 all pre-dot if you want it blank. On the route pattern that points to this
 loopback SIP trunk, you would modify the called party so that once the call
 re-enters CUCM, it will be destined for the *7999 number (pulled from you
 example before) and continues through the normal process to be forwarded to
 voicemail.



 Neither solution

Re: [cisco-voip] Making a message appear anonymous/unknown via single inbox

[Justin Steinberg]

Apologies for resurrecting an old thread here, but I was curious if there
was fixed in more recent versions of Unity Connection.   I'm trying this
with 10.5(2)su1 with SIP trunk between UCM and UCXN and having the problem
where UCXN ignores the restriction settings configured on CUCM.

I'm thinking maybe I could make a LUA script on the UCM sip trunk to UCXN,
but looking for an easier solution.

In my situation, I have certain callers that want to have their CLID
blocked.   So it is based on the placing the call, not the phone that
receives the call and I think this prevents me from using the two options
discussed here earlier.

Justin

On Thu, Jan 24, 2013 at 4:28 PM, Ted Nugent tednugen...@gmail.com wrote:

 Thanks Chris
 I like option 2 as well since they only have 24 ports as it stands. I'll
 give it a shot and let you know how it turns out. Thanks again!

 On Thu, Jan 24, 2013 at 3:16 PM, Chris Ward (chrward) chrw...@cisco.com
 wrote:

  Hi Ted,



 Seems like you are hitting some known limitations. The settings in CUCM
 for restricting Calling Party Information don’t really eliminate it, it
 seems to just mark it as “Restricted” so the information remains and CUC
 seems hell-bent on finding calling party information.



 I did find 2 solutions for you, both of which use SIP:



 1)  Setup an additional integration between CUCM and CUC that is
 meant only for this anonymous line. In the SIP trunk config, you have to
 uncheck the “Remote-Party-Id” under the “Call Routing Information”
 section and make sure “Calling Line ID Presentation” is set to
 Restricted under the “Outbound Calls” section. This will result in an
 “Unknown CallerID” message in Outlook.

 2)  Setup a Loopback SIP trunk for calls to this specific Anonymous
 DID. You only need one SIP trunk to point to CUCMs own IP address, it acts
 as the outbound and inbound SIP trunk in this scenario. Under the “Outbound
 Calls” section, you would need define a Calling Party Transformation CSS
 that would have access to a transformation pattern that matches all Calling
 parties (remembering that only calls to this anonymous DID are affected)
 and mask the calling party with some obscure mask like  or you can put
 a . at the end of the pattern you match and then drop all pre-dot if you
 want it blank. On the route pattern that points to this loopback SIP trunk,
 you would modify the called party so that once the call re-enters CUCM, it
 will be destined for the *7999 number (pulled from you example before) and
 continues through the normal process to be forwarded to voicemail.



 Neither solution is super-attractive, but both will work. I personally
 like #2 since it means you won’t have split up your CUC ports for a
 separate SIP integration. Let me know if you have any questions. It is
 probably a little more involved than the description I provided.



 +Chris

 Unity Connection TME



 *From:* Ted Nugent [mailto:tednugen...@gmail.com]
 *Sent:* Wednesday, January 23, 2013 6:47 PM

 *To:* Chris Ward (chrward)
 *Cc:* Cisco VoIPoE List
 *Subject:* RE: [cisco-voip] Making a message appear anonymous/unknown
 via single inbox



 Will do thanks!

 On Jan 23, 2013 5:57 PM, Chris Ward (chrward) chrw...@cisco.com
 wrote:

 Let me try it out in my lab tomorrow and get back to you. Remind me
 Friday morning if you don’t hear from me. J



 +Chris

 Unity Connection TME



 *From:* Ted Nugent [mailto:tednugen...@gmail.com]
 *Sent:* Wednesday, January 23, 2013 5:09 PM
 *To:* Chris Ward (chrward)
 *Cc:* Cisco VoIPoE List
 *Subject:* Re: [cisco-voip] Making a message appear anonymous/unknown
 via single inbox



 Chris thanks for the reply, Yes it is stripping the info washing it
 through a TP, the phone I'm calling shows Private. While using the TP
 method I'm basically using the Direct to voicemail  (*) VM profile
 which is CFWDAll to VM on a CTI-RP.

 For example 7999 is translated to *7999*CTIRPCFwdALL VM



 However using the Huntpilot way, I have a HP going directly to HL/LG
 containing the VM Ports and a Directed routing rule going to the subscriber
 greeting. The HP is configured calling party name/line presention to
 restricted, same as i did with the TP



 As mention both have the same affect?



 This is an SCCP integration with CUC and we get the same results via
 external calls from an MGCP controlled PRI and SIP/SCCP phones



 Any thoughts?

 Thanks

 Ted

 On Wed, Jan 23, 2013 at 1:55 PM, Chris Ward (chrward) chrw...@cisco.com
 wrote:

 Ted,



 Removing the calling party information from the call before it gets to
 CUC is definitely the way to do this. So, I think you are on the right
 track.



 Now, as to why it’s not working, there are a few things you could try. As
 a test, I would setup a translation pattern to strip the calling
 information and then forward it to your desk phone (making sure you don’t
 receive calling party information) so you can make sure that the
 translation pattern is doing what you think it is.

Re: [cisco-voip] Making a message appear anonymous/unknown via single inbox

[Justin Steinberg]

in addition to the PSTN, they also want to have caller id blocked on calls
on-net to other Cisco phones.   So we have certain CSS on Cisco phones that
route calls through translation patterns to set the restricted CLID flag
before the call rings the other Cisco phones.When we do this, the Cisco
phone called party does see 'Private' but if they don't answer the call and
it goes into voicemail then the voicemail message will still show the
calling party number.

On Tue, Jun 23, 2015 at 11:12 AM, Ryan Huff ryanh...@outlook.com wrote:

 Justin,

 For clarification, you have a user(s) that makes an outbound call from
 call manager to the pstn (via a sip trunk to an itsp?) And some of them
 want to block their Caller ID or mask it to anon?

 Thanks,

 Ryan


  Original Message 
 From: Justin Steinberg jsteinb...@gmail.com
 Sent: Tuesday, June 23, 2015 11:06 AM
 To: Ted Nugent tednugen...@gmail.com
 Subject: Re: [cisco-voip] Making a message appear anonymous/unknown via
 single inbox
 CC: Cisco VoIPoE List cisco-voip@puck.nether.net

 Apologies for resurrecting an old thread here, but I was curious if there
 was fixed in more recent versions of Unity Connection.   I'm trying this
 with 10.5(2)su1 with SIP trunk between UCM and UCXN and having the problem
 where UCXN ignores the restriction settings configured on CUCM.

 I'm thinking maybe I could make a LUA script on the UCM sip trunk to UCXN,
 but looking for an easier solution.

 In my situation, I have certain callers that want to have their CLID
 blocked.   So it is based on the placing the call, not the phone that
 receives the call and I think this prevents me from using the two options
 discussed here earlier.

 Justin

 On Thu, Jan 24, 2013 at 4:28 PM, Ted Nugent tednugen...@gmail.com wrote:

 Thanks Chris
 I like option 2 as well since they only have 24 ports as it stands. I'll
 give it a shot and let you know how it turns out. Thanks again!

 On Thu, Jan 24, 2013 at 3:16 PM, Chris Ward (chrward) chrw...@cisco.com
 wrote:

  Hi Ted,



 Seems like you are hitting some known limitations. The settings in CUCM
 for restricting Calling Party Information don’t really eliminate it, it
 seems to just mark it as “Restricted” so the information remains and CUC
 seems hell-bent on finding calling party information.



 I did find 2 solutions for you, both of which use SIP:



 1)  Setup an additional integration between CUCM and CUC that is
 meant only for this anonymous line. In the SIP trunk config, you have to
 uncheck the “Remote-Party-Id” under the “Call Routing Information”
 section and make sure “Calling Line ID Presentation” is set to
 Restricted under the “Outbound Calls” section. This will result in an
 “Unknown CallerID” message in Outlook.

 2)  Setup a Loopback SIP trunk for calls to this specific Anonymous
 DID. You only need one SIP trunk to point to CUCMs own IP address, it acts
 as the outbound and inbound SIP trunk in this scenario. Under the “Outbound
 Calls” section, you would need define a Calling Party Transformation CSS
 that would have access to a transformation pattern that matches all Calling
 parties (remembering that only calls to this anonymous DID are affected)
 and mask the calling party with some obscure mask like  or you can put
 a . at the end of the pattern you match and then drop all pre-dot if you
 want it blank. On the route pattern that points to this loopback SIP trunk,
 you would modify the called party so that once the call re-enters CUCM, it
 will be destined for the *7999 number (pulled from you example before) and
 continues through the normal process to be forwarded to voicemail.



 Neither solution is super-attractive, but both will work. I personally
 like #2 since it means you won’t have split up your CUC ports for a
 separate SIP integration. Let me know if you have any questions. It is
 probably a little more involved than the description I provided.



 +Chris

 Unity Connection TME



 *From:* Ted Nugent [mailto:tednugen...@gmail.com]
 *Sent:* Wednesday, January 23, 2013 6:47 PM

 *To:* Chris Ward (chrward)
 *Cc:* Cisco VoIPoE List
 *Subject:* RE: [cisco-voip] Making a message appear anonymous/unknown
 via single inbox



 Will do thanks!

 On Jan 23, 2013 5:57 PM, Chris Ward (chrward) chrw...@cisco.com
 wrote:

 Let me try it out in my lab tomorrow and get back to you. Remind me
 Friday morning if you don’t hear from me. J



 +Chris

 Unity Connection TME



 *From:* Ted Nugent [mailto:tednugen...@gmail.com]
 *Sent:* Wednesday, January 23, 2013 5:09 PM
 *To:* Chris Ward (chrward)
 *Cc:* Cisco VoIPoE List
 *Subject:* Re: [cisco-voip] Making a message appear anonymous/unknown
 via single inbox



 Chris thanks for the reply, Yes it is stripping the info washing it
 through a TP, the phone I'm calling shows Private. While using the TP
 method I'm basically using the Direct to voicemail  (*) VM profile
 which is CFWDAll to VM on a CTI-RP.

 For example 7999 is translated

Re: [cisco-voip] Consolidate multiple CUCM clusters using Prime Collab Provisioning

[Justin Steinberg]

I would say this would be extremely ambitious.   I would just recommend
using the UCM BAT export/import, along with some excel manipulation.
On Jun 15, 2015 3:49 PM, Rob Dawson rdaw...@force3.com wrote:

  Not much more info than the subject – is it possible to consolidate
 multiple CUCM clusters onto a new virtualized infrastructure utilizing PCP?
 Anyone done it? Pointers?



 Thanks,

 Rob

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Re: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN

[Justin Steinberg]

I should also add this is intermittent.  Sometimes Jabber will connection
to Unity Connection just fine with SSO.  Sometimes it will try to connect
to the DNS servers (Show Connection Status shows the DNS server as
voicemail) and fail.  Sometimes it will try to connect to Unity Connection
but without SSO.

On Fri, Jun 12, 2015 at 9:18 AM, Justin Steinberg jsteinb...@gmail.com
wrote:

 It seems like the problem is that for some reason, Jabber thinks the DNS
 server is the voicemail server.  So it is performing a SSO check with the
 DNS server and that fails.

 Any idea why Jabber would be doing this ?

 On Thu, Jun 11, 2015 at 1:46 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 i've looked through it and i'm not sure where it is going wrong.   It
 doesn't help that the file is 15,000 or so lines long for a three minute
 login :)

 i've opened a case with Webex Messenger team and will probably do the
 same with Jabber/CUC.

 Justin

 On Thu, Jun 11, 2015 at 11:07 AM, Brian Meade bmead...@vt.edu wrote:

 Probably worth pulling a problem report right after sign in.  It should
 show the process of checking if SSO is enabled for Unity Connection.

 On Wed, Jun 10, 2015 at 4:33 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 Thanks Chris, I checked all that out and it looks alright.

 In testing this afternoon, we've gotten the UCXN SSO to work on a
 couple occasions but it doesn't stay.  The Jabber client ends up not
 logging into voicemail and in the Jabber client FileOptionsAccounts, it
 wants the user to enter their credentials.Jabber shouldn't be allowing
 users to enter their credentials in the client with SSO enabled.

 I turned up samltrace to debug on UCXN and pulled the logs.  Jabber
 doesn't even seem to be hitting the UCXN server when it fails, it's like
 Jabber doesn't realize UCXN is SSO enabled.I'm opening a case with
 Webex Messenger support to start, then will see where that goes.

 Justin

 On Wed, Jun 10, 2015 at 1:39 PM, Chris Clouse chris.clo...@cdw.com
 wrote:

  · Verify the following Unity Connection Services are
 started.

 - Connection Jetty
   - Connection REST Service

 · Verify the class of service has Allow Users to Use the Web
 Inbox and RSS Feeds, Allow Users to Access Voice Mail Using an IMAP Client
 and/or Single Inbox, Allow Users to Use Unified Client to Access Voice 
 Mail
 enabled



 · Verify the Unity Connection API Settings are enabled
 (System Settings-Advanced-API Settings)

 - Allow Access to Secure Message Recordings through CUMI
   - Display Message Header Information of Secure Messages through
   CUMI
   - Allow Message Attachments through CUMI

 Also make sure that you don’t have
  
 VoicemailService_UseCredentialsFromphone/VoicemailService_UseCredentialsFrom
 in your jabber-config.xml file as it will need to have a separate login 
 for
 the voicemail server versus CallManager.



 *~Chris*



 *From:* Justin Steinberg [mailto:jsteinb...@gmail.com]
 *Sent:* Wednesday, June 10, 2015 12:30 PM
 *To:* Chris Clouse
 *Cc:* Cisco VOIP
 *Subject:* RE: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Yes I've done that.  The Unity Connection Web Inbox and UCM Self Care
 Portal page both have functioning SSO.   It's just Jabber that won't
 utilize SSO when using CUC.

 Justin

 On Jun 10, 2015 1:16 PM, Chris Clouse chris.clo...@cdw.com wrote:

  In order for the phone services and voicemail to be connected via
 SSO, CUCM and UCXN will also need to be enabled for SAML SSO on their own
 as well as having the WebEx Messenger SSO.  I would recommend that you be
 on 10.5+ even though it states supported with 10.0.




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/SAML_SSO_deployment_guide/10_0_1/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for_chapter_010.html



 *~Chris*



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Justin Steinberg
 *Sent:* Wednesday, June 10, 2015 12:09 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Has anyone setup SSO in the hybrid Jabber deployment model?

 Customer has hosted WebEx Messenger Jabber, on-prem CUCM, and UCXN.
 We have enabled SSO for all three systems.  We can use SSO to login to
 jabber and the UCM and UCXN end user self service web interfaces.All
 that seems OK and SSO is working.

 The problem is that once jabbers logs in to WebEx Messenger, it
 requires the user to go into fileoptions and manually enter their
 voicemail credentials.

 I expect that it should just SSO into WebEx messenger, UCM phone
 services and UCXN voicemail services.

 Any thoughts?

 Justin



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Re: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN

[Justin Steinberg]

i've looked through it and i'm not sure where it is going wrong.   It
doesn't help that the file is 15,000 or so lines long for a three minute
login :)

i've opened a case with Webex Messenger team and will probably do the same
with Jabber/CUC.

Justin

On Thu, Jun 11, 2015 at 11:07 AM, Brian Meade bmead...@vt.edu wrote:

 Probably worth pulling a problem report right after sign in.  It should
 show the process of checking if SSO is enabled for Unity Connection.

 On Wed, Jun 10, 2015 at 4:33 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 Thanks Chris, I checked all that out and it looks alright.

 In testing this afternoon, we've gotten the UCXN SSO to work on a couple
 occasions but it doesn't stay.  The Jabber client ends up not logging into
 voicemail and in the Jabber client FileOptionsAccounts, it wants the user
 to enter their credentials.Jabber shouldn't be allowing users to enter
 their credentials in the client with SSO enabled.

 I turned up samltrace to debug on UCXN and pulled the logs.  Jabber
 doesn't even seem to be hitting the UCXN server when it fails, it's like
 Jabber doesn't realize UCXN is SSO enabled.I'm opening a case with
 Webex Messenger support to start, then will see where that goes.

 Justin

 On Wed, Jun 10, 2015 at 1:39 PM, Chris Clouse chris.clo...@cdw.com
 wrote:

  · Verify the following Unity Connection Services are started.

 - Connection Jetty
   - Connection REST Service

 · Verify the class of service has Allow Users to Use the Web
 Inbox and RSS Feeds, Allow Users to Access Voice Mail Using an IMAP Client
 and/or Single Inbox, Allow Users to Use Unified Client to Access Voice Mail
 enabled



 · Verify the Unity Connection API Settings are enabled (System
 Settings-Advanced-API Settings)

 - Allow Access to Secure Message Recordings through CUMI
   - Display Message Header Information of Secure Messages through
   CUMI
   - Allow Message Attachments through CUMI

 Also make sure that you don’t have
  
 VoicemailService_UseCredentialsFromphone/VoicemailService_UseCredentialsFrom
 in your jabber-config.xml file as it will need to have a separate login for
 the voicemail server versus CallManager.



 *~Chris*



 *From:* Justin Steinberg [mailto:jsteinb...@gmail.com]
 *Sent:* Wednesday, June 10, 2015 12:30 PM
 *To:* Chris Clouse
 *Cc:* Cisco VOIP
 *Subject:* RE: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Yes I've done that.  The Unity Connection Web Inbox and UCM Self Care
 Portal page both have functioning SSO.   It's just Jabber that won't
 utilize SSO when using CUC.

 Justin

 On Jun 10, 2015 1:16 PM, Chris Clouse chris.clo...@cdw.com wrote:

  In order for the phone services and voicemail to be connected via SSO,
 CUCM and UCXN will also need to be enabled for SAML SSO on their own as
 well as having the WebEx Messenger SSO.  I would recommend that you be on
 10.5+ even though it states supported with 10.0.




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/SAML_SSO_deployment_guide/10_0_1/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for_chapter_010.html



 *~Chris*



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Justin Steinberg
 *Sent:* Wednesday, June 10, 2015 12:09 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Has anyone setup SSO in the hybrid Jabber deployment model?

 Customer has hosted WebEx Messenger Jabber, on-prem CUCM, and UCXN.   We
 have enabled SSO for all three systems.  We can use SSO to login to jabber
 and the UCM and UCXN end user self service web interfaces.All that
 seems OK and SSO is working.

 The problem is that once jabbers logs in to WebEx Messenger, it requires
 the user to go into fileoptions and manually enter their voicemail
 credentials.

 I expect that it should just SSO into WebEx messenger, UCM phone
 services and UCXN voicemail services.

 Any thoughts?

 Justin



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



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Re: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN

[Justin Steinberg]

Thanks Chris, I checked all that out and it looks alright.

In testing this afternoon, we've gotten the UCXN SSO to work on a couple
occasions but it doesn't stay.  The Jabber client ends up not logging into
voicemail and in the Jabber client FileOptionsAccounts, it wants the user
to enter their credentials.Jabber shouldn't be allowing users to enter
their credentials in the client with SSO enabled.

I turned up samltrace to debug on UCXN and pulled the logs.  Jabber doesn't
even seem to be hitting the UCXN server when it fails, it's like Jabber
doesn't realize UCXN is SSO enabled.I'm opening a case with Webex
Messenger support to start, then will see where that goes.

Justin

On Wed, Jun 10, 2015 at 1:39 PM, Chris Clouse chris.clo...@cdw.com wrote:

  · Verify the following Unity Connection Services are started.

 - Connection Jetty
   - Connection REST Service

 · Verify the class of service has Allow Users to Use the Web
 Inbox and RSS Feeds, Allow Users to Access Voice Mail Using an IMAP Client
 and/or Single Inbox, Allow Users to Use Unified Client to Access Voice Mail
 enabled



 · Verify the Unity Connection API Settings are enabled (System
 Settings-Advanced-API Settings)

 - Allow Access to Secure Message Recordings through CUMI
   - Display Message Header Information of Secure Messages through
   CUMI
   - Allow Message Attachments through CUMI

 Also make sure that you don’t have
  
 VoicemailService_UseCredentialsFromphone/VoicemailService_UseCredentialsFrom
 in your jabber-config.xml file as it will need to have a separate login for
 the voicemail server versus CallManager.



 *~Chris*



 *From:* Justin Steinberg [mailto:jsteinb...@gmail.com]
 *Sent:* Wednesday, June 10, 2015 12:30 PM
 *To:* Chris Clouse
 *Cc:* Cisco VOIP
 *Subject:* RE: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Yes I've done that.  The Unity Connection Web Inbox and UCM Self Care
 Portal page both have functioning SSO.   It's just Jabber that won't
 utilize SSO when using CUC.

 Justin

 On Jun 10, 2015 1:16 PM, Chris Clouse chris.clo...@cdw.com wrote:

  In order for the phone services and voicemail to be connected via SSO,
 CUCM and UCXN will also need to be enabled for SAML SSO on their own as
 well as having the WebEx Messenger SSO.  I would recommend that you be on
 10.5+ even though it states supported with 10.0.




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/SAML_SSO_deployment_guide/10_0_1/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for_chapter_010.html



 *~Chris*



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Justin Steinberg
 *Sent:* Wednesday, June 10, 2015 12:09 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Has anyone setup SSO in the hybrid Jabber deployment model?

 Customer has hosted WebEx Messenger Jabber, on-prem CUCM, and UCXN.   We
 have enabled SSO for all three systems.  We can use SSO to login to jabber
 and the UCM and UCXN end user self service web interfaces.All that
 seems OK and SSO is working.

 The problem is that once jabbers logs in to WebEx Messenger, it requires
 the user to go into fileoptions and manually enter their voicemail
 credentials.

 I expect that it should just SSO into WebEx messenger, UCM phone services
 and UCXN voicemail services.

 Any thoughts?

 Justin


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[cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN

[Justin Steinberg]

Has anyone setup SSO in the hybrid Jabber deployment model?

Customer has hosted WebEx Messenger Jabber, on-prem CUCM, and UCXN.   We
have enabled SSO for all three systems.  We can use SSO to login to jabber
and the UCM and UCXN end user self service web interfaces.All that
seems OK and SSO is working.

The problem is that once jabbers logs in to WebEx Messenger, it requires
the user to go into fileoptions and manually enter their voicemail
credentials.

I expect that it should just SSO into WebEx messenger, UCM phone services
and UCXN voicemail services.

Any thoughts?

Justin
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Re: [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN

[Justin Steinberg]

Yes I've done that.  The Unity Connection Web Inbox and UCM Self Care
Portal page both have functioning SSO.   It's just Jabber that won't
utilize SSO when using CUC.

Justin
On Jun 10, 2015 1:16 PM, Chris Clouse chris.clo...@cdw.com wrote:

  In order for the phone services and voicemail to be connected via SSO,
 CUCM and UCXN will also need to be enabled for SAML SSO on their own as
 well as having the WebEx Messenger SSO.  I would recommend that you be on
 10.5+ even though it states supported with 10.0.




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/SAML_SSO_deployment_guide/10_0_1/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for/CUCM_BK_SB003832_00_saml-sso-deployment-guide-for_chapter_010.html



 *~Chris*



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Justin Steinberg
 *Sent:* Wednesday, June 10, 2015 12:09 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Hosted Jabber, SSO and onprem UCM and UCXN



 Has anyone setup SSO in the hybrid Jabber deployment model?

 Customer has hosted WebEx Messenger Jabber, on-prem CUCM, and UCXN.   We
 have enabled SSO for all three systems.  We can use SSO to login to jabber
 and the UCM and UCXN end user self service web interfaces.All that
 seems OK and SSO is working.

 The problem is that once jabbers logs in to WebEx Messenger, it requires
 the user to go into fileoptions and manually enter their voicemail
 credentials.

 I expect that it should just SSO into WebEx messenger, UCM phone services
 and UCXN voicemail services.

 Any thoughts?

 Justin

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Re: [cisco-voip] UCCX 10.6 | CUCM 10.5.2 | CAD Extension out of service

[Justin Steinberg]

Are you on UCM 10.5.2su1 ?   I haven't seen that issue at all.

Is your VMware LRO disabled ?

Do your UCMs have one or two vCPU ?
On Jun 9, 2015 12:42 AM, Ryan Huff ryanh...@outlook.com wrote:

 Has anyone been noticing that in this environment (UCCX 10.6 | CUCM 10.5.2
 | CAD); extensions will sometimes report as OutOfService (and subsequently
 cause CAD login issues) when in fact, the phone is registered, associated
 to the jTAPI/RMCMSubsys user, has CTI Control enabled (and is setup
 correctly in all other respects)?

 What seems to resolve this issue for me; is to super copy the phone to a
 fake mac address, delete the original and then re-mac the newly created
 device. Doing a cluster reboot seems to address this as well.

 This is a new 2 cluster build. Everything about the phone seems to
 function fine within call manager when CAD is having a login issue. I've
 changed the addressing mode on the phone (and CTI ports) to ipv4 only,
 turned JAL off on the phone. Just really odd and not very easy to define.

 I know CAD is near death and Finesse is the future, but I have to support
 CAD in this scenario.

 Thanks,

 Ryan

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Re: [cisco-voip] UCCX 10.6 | CUCM 10.5.2 | CAD Extension out of service

[Justin Steinberg]

I always turn off LRO.  I've had weird issues that are next to impossible
to troubleshoot at the application layer and disabling LRO is the solution.
   I would also recommend that if you are using the 2500 user template that
you make sure your callmanagers are using 2 vCPUs when using CCX.  I have
found the system to be much more reliable and stable with 2 vCPU.

Justin

On Tue, Jun 9, 2015 at 8:40 AM, Ryan Huff ryanh...@outlook.com wrote:

 Hi Justin, thank you for your reply.

 CUCM 10.5.2su1 is in the active partition on all nodes (10.5.2.11900-3).

 LRO is interesting; my elastic sky version is 5.5 and according to the
 DocWiki (http://docwiki.cisco.com/wiki/Disable_LRO) it seems to only be
 an issue with 4.1 and more specifically, with CUCM  8.6 with ESXi 4.1.
 I've also seen several UC evangelical sites proclaim that LRO needs to be
 disabled where ever Virtual Cisco UC runs, versions be damned.

 I'm in pre-flight, so now is definitely the time to power down and knock
 LRO off.

 I guess I am more interested in the *why* at this point (plight of an
 engineer). I suppose this issue (if it is even an issue) will self-heal
 itself in v11 when CAD is no longer an option, but still plenty of need to
 support existing CAD.

 This is the first time I used this version mix (UCCX 10.6 | CUCM 10.5.2su1
 | CAD), and I haven't ran into this issue before.

 Thanks for any thoughts you put into this.

 -Ryan


 --
 Date: Tue, 9 Jun 2015 07:41:46 -0400
 Subject: Re: [cisco-voip] UCCX 10.6 | CUCM 10.5.2 | CAD Extension out of
 service
 From: jsteinb...@gmail.com
 To: ryanh...@outlook.com
 CC: cisco-voip@puck.nether.net


 Are you on UCM 10.5.2su1 ?   I haven't seen that issue at all.

 Is your VMware LRO disabled ?

 Do your UCMs have one or two vCPU ?
 On Jun 9, 2015 12:42 AM, Ryan Huff ryanh...@outlook.com wrote:

 Has anyone been noticing that in this environment (UCCX 10.6 | CUCM 10.5.2
 | CAD); extensions will sometimes report as OutOfService (and subsequently
 cause CAD login issues) when in fact, the phone is registered, associated
 to the jTAPI/RMCMSubsys user, has CTI Control enabled (and is setup
 correctly in all other respects)?

 What seems to resolve this issue for me; is to super copy the phone to a
 fake mac address, delete the original and then re-mac the newly created
 device. Doing a cluster reboot seems to address this as well.

 This is a new 2 cluster build. Everything about the phone seems to
 function fine within call manager when CAD is having a login issue. I've
 changed the addressing mode on the phone (and CTI ports) to ipv4 only,
 turned JAL off on the phone. Just really odd and not very easy to define.

 I know CAD is near death and Finesse is the future, but I have to support
 CAD in this scenario.

 Thanks,

 Ryan

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Re: [cisco-voip] Priority of Calls To Hunt Pilot

[Justin Steinberg]

first come first serve

On Mon, Jun 8, 2015 at 10:40 AM, Gary Parker g.j.par...@lboro.ac.uk wrote:

 Afternoon all, I wonder if anyone could shed some light on a query I have
 regarding my CUCM 8.6.2 system.

 I have a hunt pilot that is feeding into a small callcentre operation (6
 DNs, one line group, longest idle distribution). The vast majority of calls
 to the hunt pilot will be coming via SIP from off-net via an MGCP connected
 CUBE. I will also be seeing a significantly smaller number of calls from
 on-net to the same hunt pilot.

 Assuming that all operators are busy and the hunt pilot is presenting an
 engaged tone, does CUCM perform any prioritisation over what calls get
 connected to the hunt group (on-net or off-net) first and is that
 configurable? I.e. can I prioritise internal calls getting connected to the
 hunt group over external?

 ---
 /-Gary Parker--f--\
 | Unified Communications Service Manager  |
 n  Loughborough University, IT Services   |
 | tel:+441509635635 im:cc...@lboro.ac.uk  o
 | http://delphium.lboro.ac.uk/pubkey.txt  |
 \r--d-/


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Re: [cisco-voip] collab edge dns/SSL cert

[Justin Steinberg]

Click through the error.  Dont modify the CSR or take out SANs.   The fqdn
should be in the CN and SAN.  I dont know why godaddy complains about that
but I just ignore it and things are fine.

Justin
On Jun 1, 2015 1:49 PM, Ed Leatherman ealeather...@gmail.com wrote:

 Matt had it right with his suggestion of dumping the CSR into the decoder,
 although I wouldn't have recognized it as a problem.

 When expressway generates the CSR it is adding a SAN entry that is
 identical to the CN. So it doesn't seem like having my root domain in there
 was the problem to begin with. According to the GoDaddy support person that
 was what was kicking the error - and apparently if you just click through
 the error it will generate the cert anyway, i'm assuming it will just leave
 out that offending SAN entry.

 I'll circle around once we have the verifications done and have a chance
 to upload it.

 On Mon, Jun 1, 2015 at 10:32 AM, Ed Leatherman ealeather...@gmail.com
 wrote:

 I tried a different CSR with alternate names collab-edge.domain.edu and
 expe.telecom.domain.edu , without the generic domain.edu, still same
 error. I'll see what godaddy support tells me.

 On Mon, Jun 1, 2015 at 10:03 AM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  It could be depending on what exactly was ordered, but I know godaddy
 supports having the domain as a SAN. I have it on certs I’ve bought in the
 past month for expressway and it’s actually supposed to be there:




 http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-5/Cisco-Expressway-Certificate-Creation-and-Use-Deployment-Guide-X8-5-1.pdf



 See page 8 and 9. You can prefix collab-edge to the domain if you like,
 but if you are doing XMPP federation you need it anyway.









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

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



 *From:* Chris Ward (chrward) [mailto:chrw...@cisco.com]
 *Sent:* Monday, June 1, 2015 9:52 AM
 *To:* Matthew Loraditch; Ed Leatherman; Cisco VOIP
 *Subject:* RE: [cisco-voip] collab edge dns/SSL cert



 I think the problem is requesting your root domain. Some issuers won’t
 issue root domain certs and the ones that do call them wildcard certs as
 they cover an entire domain (support for wildcard certs are somewhat
 limited).



 For example, if you were to go to https://cisco.com/ rather than
 https://www.cisco.com/ you would find that the first has an invalid SSL
 cert as cisco doesn’t have a root domain cert.



 For the very security savvy, it is considered to be inappropriate to use
 domain-level certs.



 Go with just the hostname of the Expressway and potentially an actual
 alternate hostname if you ever needed to provide an alternate DNS entry to
 reach the same Expressway. In either case, drop domain.edu. You don’t
 need it and I suspect that’s that GoDaddy is complaining about.



 +Chris

 TME - MediaSense and Unity Connection



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Matthew Loraditch
 *Sent:* Monday, June 01, 2015 9:44 AM
 *To:* Ed Leatherman; Cisco VOIP
 *Subject:* Re: [cisco-voip] collab edge dns/SSL cert



 https://www.sslshopper.com/csr-decoder.html



 Try dumping the csr in there and see if you see something unexpected.



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

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



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Ed Leatherman
 *Sent:* Monday, June 1, 2015 9:41 AM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] collab edge dns/SSL cert



 Hello everyone!



 I'm getting an error kicked back from GoDaddy trying to sign my
 expressway-e cert, looking for a sanity check here.



 I'm setting up the external side as a cluster (of 1 currently), I'd like
 for my users to be able to sign in as usern...@domain.edu for MRA.



 dns:

 expressway-e is expe-cluster1-node1.domain.edu

 srv = _collab-edge._tls.domain.edu , sips._tcp.domain.edu both point to
 the expe-cluster1-node1



 exp-e cluster name is domain.edu



 on my CSR i have it set to generate a SAN for FQDN of expressway cluster
 plus FQDN of this peer, so:

 DNS:expe-cluster1-node1.domain.edu

 DNS:domain.edu



 GoDaddy kicks back an error saying You can not add a SAN that is the
 same as the domain you are already using.



 Is my dns/SAN configuration incorrect or is this a deficiency with
 godaddy (standard UCC cert)? Or did I 

Re: [cisco-voip] Changing DNS entries in Call Manager 9.1.2.10000-28

[Justin Steinberg]

are you sure that all your CM subscribers have the proper primary and
secondary DNS ?Since your outbound is the only think that failed, maybe
you have a specific CM sub routing the outbound calls and it didn't have
proper redundant DNS.   Or maybe you had a MGCP gateway for outbound that
didn't have secondary DNS so it unregistered from the CMs when the primary
went down.or maybe you hit a bug with regards to DNS failover like you
suspect, but it is also possible that there is just a component in the
solution that doesn't have redundant DNS and it only affected outbound
calls.

On Wed, May 27, 2015 at 10:17 AM, Gyrion, Larry larry.gyr...@deancare.com
wrote:

  We do have a secondary DNS in place,  after further investigation he
 primary DNS never went fully down, it went unresponsive during a back-up
 procedure.

 Is it that since the DNS never went ‘fully’ down the Cisco voice side (the
 SIP trunks) never knew to switch to the secondary DNS (not as smart as the
 Microsoft workstations/servers).





 Thank you



 *From:* Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com]
 *Sent:* Tuesday, May 26, 2015 8:03 PM
 *To:* Gyrion, Larry; Cisco-voip (cisco-voip@puck.nether.net)
 *Subject:* RE: Changing DNS entries in Call Manager 9.1.2.1-28



 Everything is hostnames so https works without complaining.  Certificates
 with ip addresses give warnings.  443/TLS/PKI is the future J



 You can change CUCM back to ip address but applications and websites,
 clients like Jabber, will give warnings/errors.  I think your DNS should be
 rock solid, maybe you need secondary/tertiary dns entries.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Gyrion, Larry
 *Sent:* Tuesday, May 26, 2015 5:20 PM
 *To:* Cisco-voip (cisco-voip@puck.nether.net)
 *Subject:* [cisco-voip] Changing DNS entries in Call Manager
 9.1.2.1-28





 We had an issue where we lost outbound calling ability when out primary
 DNS experiencing an unscheduled outage.

 Our DNS entries are by host-name, not IP address.  (it never failed over
 to the secondary DNS server, other items like computers did and internal
 and incoming traffic was working fine)



 We also use UCCE 9



 I’m not sure why it was configured by host name rather than IP address
 when it was configured a long time ago.



 So my questions are:

 Is there a valid reason why we use host-names instead of ip addresses?



 How can we change from host-name to IP address?

 Will this affect the licensing (ELM)? (The below is reference to pre 9.0
 CUCM)



 *From:* avhollo...@gmail.com [mailto:avhollo...@gmail.com
 avhollo...@gmail.com] *On Behalf Of *Anthony Holloway
 *Sent:* Monday, January 26, 2015 8:13 PM
 *To:* Gyrion, Larry; Cisco-voip (cisco-voip@puck.nether.net)
 *Subject:* Re: [cisco-voip] Changing DNS entries in Call Manager 8.6.2



 The easiest way to view the license MAC, is to SSH to the server, and
 issue the show status command.



 Also, http://cisco.com/go/license enables you to rehost your own license
 files without opening a case.  Of course, I don't guarantee you'll be
 successful, but it's nice to know this option exists.



 [image: Inline image 1]



 Another thing to note, you will get 30 days to rehost your license before
 anything bad happens to your servers, but if you're in a pinch, and you're
 like on day 28 and you need like 10 more days, you can revert your change,
 then make the same change again, to restart the 30 day period.



 If that was confusing, let me use this example.  If my primary DNS was
 1.1.1.1, and I changed it to 2.2.2.2, I would have 30 days to rehost my
 licenses.  On day 28, I set the primary DNS back to 1.1.1.1, then
 immediately back to 2.2.2.2, and the 30 days starts over.



 Last, buy certainly not least, if you are changing DNS settings, it would
 be imperative for you to consider what might happen if you changed your DNS
 suffix.  I cannot speak to your environment exactly, but suffice it to say,
 certificates are based on names, and names sometimes contain DNS suffixes.
 You might start a chain reaction of changes, and as such you should plan
 that piece out more carefully.  If you're only changing DNS server
 addresses, then you can ignore this last paragraph.



 Good luck.



 On Mon Jan 26 2015 at 4:43:19 PM Gyrion, Larry larry.gyr...@deancare.com
 wrote:

 Looking for some guidance on updating the DNS entries on our CUCM
 cluster.  A colleague went through the process, but upon entering the
 command received a warning stating that the change would invalidate our
 licenses.  Has anybody come across this before, and if so, what was the
 proper course of action to ensure license preservation?

 CUCM 8.6.2





 Thank you,

 *Larry Gyrion** | Telecommunications Analyst | Information Technology*
 Dean Clinic - Corporate offices
 1800 W. Beltline Hwy
 Madison WI. 53713
 Phone 608.294.6201 | 5406201| Fax 608.280.6852
 larry.gyr...@deancare.com | 

Re: [cisco-voip] UCCX Upgrade question

[Justin Steinberg]

that's not possible afaik

On Tue, May 19, 2015 at 9:25 PM, Ryan Huff ryanh...@outlook.com wrote:

 I have CCX 8.5.1; I am going to 10.6.1.

 I am NOT doing an in-place upgrade. I'm having issues sourcing the .ISO
 for their current version, so I can restore/jump upgrade it to 10.6.1 and
 get a same version DRS to import.

 Is it possible to extract the historical reporting data *only*, and move
 that into CUIC (CCX 10.6.1). I could be asking something ridiculous; just
 not that versed in this area of CCX.

 Any other ideas? (in-place upgrade just isn't possible due to other
 factors not-relevant to this question).

 Thanks,

 ryan

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Re: [cisco-voip] Remove ELM/PLM Option

[Justin Steinberg]

it immediately reboots the server (after you confirm the change) and after
the reboot the link for PLM is still there on the main webpage.   However,
once you click the link it says PLM has been removed by your
administrator.

I know that behavior isn't what any of us want, but atleast it will prevent
confusion with someone logging in and wondering why the licenses are lost.

On Thu, May 21, 2015 at 12:11 PM, Heim, Dennis dennis.h...@wwt.com wrote:

  I have not had a chance yet,



 *Dennis Heim | Emerging Technology Architect (Collaboration)*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]

 Innovation happens on project squared -- http://www.projectsquared.com



 *Click here to join me in my Collaboration Meeting Room
 https://wwt.webex.com/meet/dennis.heim*







 *From:* Erick [mailto:erick...@gmail.com]
 *Sent:* Tuesday, May 19, 2015 9:28 PM
 *To:* Anthony Holloway
 *Cc:* Heim, Dennis; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Remove ELM/PLM Option



 Did you try the command to remove this?



 Any side effects or gotchas? Just curious as I'd like to get rid of the
 link on some subscriber nodes and such.



 Sent from my iPhone


 On May 15, 2015, at 3:39 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

  Search for:

 Remove Cisco Prime License Manager



 In:




 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/plm/10_5_1/userguide/CPLM_BK_U9B156B7_00_user-guide-rel-1052.pdf



 On Fri, May 15, 2015 at 3:15 PM Heim, Dennis dennis.h...@wwt.com wrote:

  Is there a command to remove the ELM option from the webpage of CUCM? I
 have a dedicated ELM, and I thought there was a way to do this.



 *Dennis Heim | Emerging Technology Architect (Collaboration)*

 World Wide Technology, Inc. | +1 314-212-1814

 image001.png https://twitter.com/CollabSensei

 image002.pngimage003.png +13142121814image004.png

 Innovation happens on project squared -- http://www.projectsquared.com



 *Click here to join me in my Collaboration Meeting Room
 https://wwt.webex.com/meet/dennis.heim*







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[cisco-voip] CUCM Immersive Video Device List

[Justin Steinberg]

Below is a list of all the devices on CUCM 10.5(2)SU1 that are configured
to use the Immersive Video region bandwidth setting.



Does anyone know if there is a way to modify this list? I would like to add
the new DX series into the list.   Seems like an oversight since the EX are
in the list.



Product

Protocol

Feature

Parameters

Cisco TelePresence

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 1000

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 1100

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 1300-47

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 1300-65

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 200

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 3000

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 3200

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 400

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 500-32

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence 500-37

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Codec C40

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Codec C60

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Codec C90

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence EX60

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence EX90

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence IX5000

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX200

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX200 G2

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX300

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX300 G2

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX700

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence MX800

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 42 (C20)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 42 (C40)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 42 (C60)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 52 (C40)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 52 (C60)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 52 Dual (C60)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 65 (C60)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Profile 65 Dual (C90)

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence Quick Set C20

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence SX10

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence SX20

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence SX80

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence TX1310-65

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence TX9000

SIP

Immersive Video Support for TelePresence Devices

Cisco TelePresence TX9200

SIP

Immersive Video Support for TelePresence Devices

Generic Multiple Screen Room System

SIP

Immersive Video Support for TelePresence Devices

Generic Single Screen Room System

SIP

Immersive Video Support for TelePresence Devices
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Re: [cisco-voip] RTMT??? or some other way to monitor a phone

[Justin Steinberg]

RTMT has a device search that let's you filter by device pool or subnet.
On May 10, 2015 7:38 PM, Abhiram Kramadhati (akramadh) akram...@cisco.com
wrote:

  This might be helpful: https://www.singlewire.com/free-xml-apps.html


   From: Scott Voll svoll.v...@gmail.com
 Date: Saturday, 9 May 2015 12:24 am
 To: Brian Meade bmead...@vt.edu
 Cc: cisco-voip@puck.nether.net cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone

   I was really hoping not to pull SDL traces (more work) and I was hoping
 just to look at a small segment of phones not everything.

  Guess that answers my question.

  Scott



 On Thu, May 7, 2015 at 5:19 PM, Brian Meade bmead...@vt.edu wrote:

 Using the Syslog Viewer in RTMT, open the AlternateSyslog for the primary
 subscriber the phones are registered to.  That will show all phone
 registrations/unregistrations.

  On Thu, May 7, 2015 at 5:42 PM, Scott Voll svoll.v...@gmail.com wrote:

  we have 8861 phones that are just below an AP that are registering and
 unregistering (but working well enough that we have not moved them back to
 wired) so we are trying to monitor how much it's happening.  Already have a
 TAC case open on the wireless side.

  Scott


 On Thu, May 7, 2015 at 2:38 PM, Ryan Huff ryanh...@outlook.com wrote:

 Might be easier to start the other way around.

 What is actually wrong?

 RTMT does work with registered phones, and presumably,  your wireless
 phones are registered?

 Are the wireless phones not registering to call manager?

 Thanks,

 Ryan


  Original Message 
 From: Scott Voll svoll.v...@gmail.com
 Sent: Thursday, May 7, 2015 05:34 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] RTMT??? or some other way to monitor a phone

 We are troubleshooting a wireless deployment of 8861 phones.

  is there any way to monitor just a phone / phone model / device pool
 / etc so I can just see what I'm looking for?

  Looks like RTMT is just all registered phones.

  TIA

  Scott



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Re: [cisco-voip] PCD 10.5(3) and IMP 9x to 10x

[Justin Steinberg]

i've done it and it is mostly straightforward.  In my case, I did a network
migration to change the hostname of IM/P.   However, I used the same IP
address as the old server (this was a MCS to UCS 8 to 10 migration).

I only did a network migration because I wanted to change the hostname to
align with naming conventions.The first go around, I didn't update the
DNS with the new hostname A/PTR for the new IMP name.   The PCD migration
task for the IMP server failed.  It needed to have the DNS server updated
with the new IM/P 10.x server's A/PTR in DNS and the old MCS v8 CUPS server
removed from DNS (since it was using the same IP)..

Make sure that all the VMs in the new v10 cluster have perfect DNS
resolution, A  PTR before the PCD.   If there are any issues, this will
mess up the migration.

Also, open the console of the VM in VMware during the PCD tasks to monitor
the status of the tasks.   PCD doesn't provide alot of progress update, but
if you watch the console of the VM in VMware you can follow the familiar
CLI install process.

On Mon, May 4, 2015 at 10:57 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Has anyone experience with PCD and performing a *Network Migration* on
 IMP?  If not, how about performing a *Network Migration* on CUCM, and
 then how did you handle IMP?

 Thanks.

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Re: [cisco-voip] PCD 10.5(3) and IMP 9x to 10x

[Justin Steinberg]

I'd recommend to install PCD, discover your esxi hosts and then your old UC
cluster(s), then you can go through the various wizards to see if PCD will
let you do what you are trying to do.

If I recall with PCD 10.5.2, I could change the hostname but not the domain
name.  That's my recollection, though it may not be right.  But again, if
you setup PCD you can walk through the tutorial and checkout the options.
On May 4, 2015 11:30 PM, Anthony Holloway avholloway+cisco-v...@gmail.com
wrote:

 That's a relief to hear you've done it. The documentation isn't very clear
 if a Network Migration is possible, let alone a Simple Migration. Just take
 a look at the PCD 10.5(3) doc under supported releases. Thanks for
 replying.
 On Mon, May 4, 2015 at 10:27 PM Justin Steinberg jsteinb...@gmail.com
 wrote:

 i've done it and it is mostly straightforward.  In my case, I did a
 network migration to change the hostname of IM/P.   However, I used the
 same IP address as the old server (this was a MCS to UCS 8 to 10 migration).

 I only did a network migration because I wanted to change the hostname to
 align with naming conventions.The first go around, I didn't update the
 DNS with the new hostname A/PTR for the new IMP name.   The PCD migration
 task for the IMP server failed.  It needed to have the DNS server updated
 with the new IM/P 10.x server's A/PTR in DNS and the old MCS v8 CUPS server
 removed from DNS (since it was using the same IP)..

 Make sure that all the VMs in the new v10 cluster have perfect DNS
 resolution, A  PTR before the PCD.   If there are any issues, this will
 mess up the migration.

 Also, open the console of the VM in VMware during the PCD tasks to
 monitor the status of the tasks.   PCD doesn't provide alot of progress
 update, but if you watch the console of the VM in VMware you can follow the
 familiar CLI install process.

 On Mon, May 4, 2015 at 10:57 PM, Anthony Holloway 
 avholloway+cisco-v...@gmail.com wrote:

 Has anyone experience with PCD and performing a *Network Migration* on
 IMP?  If not, how about performing a *Network Migration* on CUCM, and
 then how did you handle IMP?

 Thanks.

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Re: [cisco-voip] dx80 and MRA

[Justin Steinberg]

i noticed that Expressway X8.5.2 was just released and was surprised when
the DX/7800/8800 was still listed as Preview support.

I suspect the Preview is due to the items below:
1) Primary line support only  (multiple lines won't work on MRA)
2) Hunt group/pilot/list aren't supported
3) Shared line limitations (status of calls on other appearances of the
shared line aren't reliable)

I hope those items are on the roadmap to be resolved before full MRA
support for these devices. The Primary line feature for sure, since
this will be a big issue in call centers.

On Thu, Apr 30, 2015 at 12:10 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.com wrote:

 EFT is ongoing but it’s not an open one like Jabber does.

 Last target I heard was end of May.

 You’ll see the 78/88XX release much sooner than that but with caveats.
 The release notes for 10.3.1 will have more details when they post.

 -Ryan

 On Apr 29, 2015, at 3:02 PM, Charles Goldsmith wo...@justfamily.org
 wrote:

 I know there have been other discussions about this, and an update to
 the dx80 code was slated for q2y15.

 Can we have an update from someone in the know?

 Is there a beta/trial program available for the code?  If so, we could
 probably assist in testing if needed and also I work for a partner, if
 that helps.  Contact me off list for details.

 Thanks!
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Re: [cisco-voip] RTMT on Windows 8.1

[Justin Steinberg]

works fine for me.  what error are you getting ?

On Tue, Apr 21, 2015 at 4:27 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  Hmm need to try with local admin account. Same but on Domain. Argh..



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

  Facebook https://www.facebook.com/heliontech?ref=hl | Twitter
 https://twitter.com/HelionTech | LinkedIn
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 G+ https://plus.google.com/+Heliontechnologies/posts



 *From:* Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com]
 *Sent:* Tuesday, April 21, 2015 4:04 PM
 *To:* Matthew Loraditch; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] RTMT on Windows 8.1



 Yes, I do.



 Windows is: 6.3 build 9600 (aka Win 8.1)

 CUCM is: 10.5.2.11900-3 (aka 10.5(2)SU1)

 RTMT is: 10.5.0.0



 Note that I am not a member of a domain, I turn off UAC, and I am the
 administrator.  Anything else you want to know about my setup?



 On Tue, Apr 21, 2015 at 2:56 PM Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  Anyone have this beast working? Compatibility mode has me nowhere…  This
 is latest RTMT from 10.5.2



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 Network Engineer
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Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5

[Justin Steinberg]

Hi Adam,

Can you elaborate on this yet ?  :)I never see any PCD software posted
to cisco.com, it always seems to require E-delivery download. I'm not
sure why that is, since all the other Prime Collaboration (Provisioning 
Assurance) can be downloaded from cisco.com

Thanks,

Justin

On Mon, Apr 13, 2015 at 11:54 AM, Adam Frankel (afrankel) 
afran...@cisco.com wrote:

  “Maybe an ability to leverage ISOs on VMWare datastores instead of from
 the PCD server. “



 …check cisco.com later this week or early next week J



 --

 Adam



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Heim, Dennis
 *Sent:* Monday, March 30, 2015 9:16 AM
 *To:* Matthew Collins; Haas, Neal; 'Andrew Grech'; Nick

 *Cc:* Cisco VoIP Group
 *Subject:* Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5



 PCD is a great tool for single-site installs or upgrades in a lab
 environment. Biggest limitation of PCD when multisites are involved, since
 it pushes all ISOs from the PCD server. Maybe an ability to leverage ISOs
 on VMWare datastores instead of from the PCD server.



 *Dennis Heim | Emerging Technology Architect (Collaboration)*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]

 Innovation happens on project squared -- http://www.projectsquared.com



 *Click here to join me in my Collaboration Meeting Room
 https://wwt.webex.com/meet/dennis.heim*







 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Matthew Collins
 *Sent:* Monday, March 30, 2015 6:08 AM
 *To:* Haas, Neal; 'Andrew Grech'; Nick
 *Cc:* Cisco VoIP Group
 *Subject:* Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5



 Just a note on this.



 I have just completed a direct Hardware 8.5 to VM 10.5 with network
 changes using Prime collaboration deployment. No Cop files need to be
 installed, No jump upgrades, No restoring from backup. Really was a simple
 process.



 If you haven’t used PCD I would strongly suggest taking a look next time
 you upgrade or install a new build.



 Regards



 Matthew Collins





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Haas, Neal
 *Sent:* 27 March 2015 14:41
 *To:* 'Andrew Grech'; Nick
 *Cc:* Cisco VoIP Group
 *Subject:* Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5



 10.5 is VM only, 8.6.1 usually was MCS hardware.



 To upgrade you will need to 8.6.1 upgrade to 9.1 on MCS. Then 9.1 MCS to
 VM, the upgrade 9.1 to 10.5(2)SU1.





 Neal Haas



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Andrew Grech
 *Sent:* Friday, March 27, 2015 7:30 AM
 *To:* Nick
 *Cc:* Cisco VoIP Group
 *Subject:* Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5



 Supported vs the system will upgrade are two different things.

 On 27/03/2015 1:51 AM, Nick csv...@googlemail.com wrote:

 Hi All



 Just checking through documentation for CUCM 10.5 for an upgrade, the
 compatibility guide states that a direct upgrade is from 8.6.1 onwards as
 shown below.


   Upgrade Paths for Cisco Unified Communications Manager Release
 10.5(2)SU1



 *Note*


  --

 If your release is not listed in the following table, find the upgrade
 path from your current release to a listed release in Cisco Unified
 Communications Manager Software Compatibility Matrix for Release 9.X and
 Earlier at
 http:/​/​www.cisco.com/​en/​US/​docs/​voice_ip_comm/​cucm/​compat/​ccmcompmatr1.pdf
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr1.pdf.

  --





 Table 15 Export Restricted Supported Cisco Unified Communications Manager
 Upgrades for Release 10.5(2)SU1

 *10.5(2)SU1 *

 10.5.2.11900-3

 Active

 February 24, 2015

 Direct Upgrade:

 10.5(2), 10.5(1)SU1a, 10.5(1)SU1, 10.5(1), 10.0(1)SU2, 10.0(1)SU1,
 10.0(1), 9.1(2)SU2a, 9.1(2)SU2, 9.1(2)SU1, 9.1(2), 9.1(1a), 9.1(1), 9.0(1),

 8.6(2a)SU5, 8.6(2a)SU4a, 8.6(2a)SU4, 8.6(2a)SU3, 8.6(2a)SU2, 8.6(2a)SU1,
 8.6(2a),

 8.6(2), 8.6(1a), 8.6(1)

 Supported: (Consult the Cisco Unified Communications Manager Upgrade Guide
 for details)

 8.5.(1)SU7, 8.5.(1)SU6, 8.5(1)SU5, 8.5(1)SU4, 8.5(1)SU3, 8.5(1)SU2,

 8.5(1)SU1, 8.5(1), 8.0(3a)SU3, 8.0(3a)SU2, 8.0(3a)SU1, 8.0(3a),

 8.0(3), 8.0(2c)SU1, 8.0(2c), 8.0(2b), 8.0(2a), 8.0(2), 8.0(1),

 7.1(5b)SU6(restricted), 7.1(5b)SU5(restricted), 7.1(5b)SU4(restricted),

 7.1(5b)SU3(restricted), 7.1(5b)SU2(restricted), 7.1(5b)(restricted),

 7.1(5a)(restricted), 7.1(5)SU1a(restricted), 7.1(5)SU1(restricted),

 7.1(5)(restricted), 7.1(3b)SU2, 7.1(3b)SU1, 7.1(3b), 7.1(3a)SU1a,

 7.1(3a)SU1, 7.1(3a), 7.1(3), 6.1(5)SU3, 6.1(5)SU2, 6.1(5)SU1,

 6.1(5), 6.1(4a)SU2, 6.1(4a), 6.1(4)SU1, 6.1(4)



 However in the Read Me notes for 10.5.2 Su1 it states the following



  *Version and Description*

 This SU is a 

Re: [cisco-voip] CIPC via MRA? Anyone done it?

[Justin Steinberg]

CIPC isn't MRA aware

On Fri, Apr 17, 2015 at 4:36 PM, Erick Wellnitz ewellnitzv...@gmail.com
wrote:

 Had a question posed to me and I'm not 100% on the answer.  I think the
 answer is no but I thought I'd see if anyone had done this successfully.

 Anyone been successful at getting CIPC to register via MRA?

 I'm thinking that because the connection to Expressway-E is TLS it isn't
 going to work.

 ___
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Re: [cisco-voip] Finesse customisations - feedback

[Justin Steinberg]

Abhiram,

Thanks for soliciting the feedback.

I've found a few items on my list that I'll share.  All my experience is
from UCCX by the way.


   - UCCX Finesse API doesn't support passing call variables in the
   MAKE_CALL request.   This has been a huge problem for me, since we are
   using this API to place calls from a CRM.  We want to pass the account
   numbers into Finesse but it won't let us.   This is documented in the
   Finesse dev guide as a limitation/not supported.  But I wish the BU would
   reconsider this to provide more capabilities into UCCX/Finesse API.
   - UCCX Finesse API for 'Update Call Variable' data doesn't allow an
   admin user to submit requests on behalf of agents.   We tried to use this
   to have the CRM update the call variable after the fact (due to #1 above)
   but the CRM wasn't able to do it because the admin credentials on Finesse
   don't have access to this API.  It has to be the agent themselves who
   update their own call, which is a problem since the CRM doesn't have the
   agent's credentials.
   - Finesse gadgets that are placed in a 'tab' in the desktop layout have
   frames added around them.  There is no way to prevent Finesse from adding
   this frame/border.However, if you add the gadget to the global 'page'
   inside Finesse, then it does not add a frame.I think that Finesse
   should allow the admin to choose whether or not a frame/border is added to
   a gadget, regardless of whether the gadget is added to 'page' or 'tab'.
   - Finesse default Team Performance gadget.   It would be nice to have
   the system reason codes added into this gadget.  Today, if the user is 'not
   ready' due to a custom code, it is displayed in the gadget.  However, if
   they are not ready due to an inbound non-acd call or outbound call, the
   gadget just shows as 'not ready'.   This has let to supervisors not being
   able to determine whether an agent is slacking off, or on an outbound call.
   - Functionality to alert/remind a user that they are in 'not ready'
   state.   I've had problems in small call centers where an agent will forgot
   their in not ready, or will RONA a call and not realize.  In any case, they
   end up not ready and don't realize it.   If Finesse could have an option to
   remind users they are not ready, this would help in certain environments
This should definitely be an option that could be enabled, since some
   environments would not want to have this feature.
   - Access to source code for the callcontrol  teamperformance gadget (or
   ability to modify those gadgets)
   - 'Finesse Lite' that can be added into a Jabber custom tab.  There are
   some third party solutions for this today, but should be something that
   could be included by default.

Justin

On Tue, Apr 14, 2015 at 4:15 AM, Abhiram Kramadhati (akramadh) 
akram...@cisco.com wrote:

  Hi Erik,

  This is actually a follow up of that and also a follow-up of the
 Ask-the-Expert session I did on Finesse.

  I noticed that there are quite a few partners on this group who might
 not have had a chance to provide feedback on those two events and hence the
 question.

  Cheers,
 Abhiram Kramadhati

 Sent from my iPhone

 On 14 Apr 2015, at 5:46 pm, Erik Goppel egop...@gmail.com wrote:

   Abhiram,

  Please have a look at the discussion on the community.
 https://communities.cisco.com/thread/51226
  this list was set up ate the UCCE Tech Summit in Amsterdam, and
  has been discussed with Ted Phipps with the whole group there.

  Also please ask your question in that thread or the community, as that
 is the place where ATP Partners, will look, as not all of them are
 subscribed to a list like this.


  Thanks,

  Erik Goppel
 Technical Consultant Unified Communications
 Dimension Data Netherlands


 On Tue, Apr 14, 2015 at 4:57 AM, Abhiram Kramadhati (akramadh) 
 akram...@cisco.com wrote:

  Hi all,

  Have received good amount of feedback about Finesse and some of the
 features. We are currently working on this and in that regard, would
 appreciate some feedback on this from your experience from the field:

  *Which are the most common customisations you have been asked from
 customers when you are trying to deploy Finesse? - a customisation need not
 be a new gadget integrating to another server. It could just be modifying
 out-of-the-box Finesse behaviour. *

  If you have got some time and if you have something to share about
 this, please drop me a line at *akram...@cisco.com akram...@cisco.com*

  Thanks in advance!

   *Abhiram Kramadhati*
 Technical Solutions Manager
 Customer Solutions Success team, CBABU
 akram...@cisco.com
 Phone: *+61 2 8446 6257 %2B61%202%208446%206257*

 CCIE Voice - 40065

 *Cisco Systems Australia Pty Limited*
 The Forum
 201 Pacific Highway
 2065
 St Leonards
 Australia
 Cisco.com http://www.cisco.com/web/AU/

 http://wwwin.cisco.com/marketing/corporate/brand/intelbrand/brandstrat/signature/Insert%20your%20LinkedIn%20link
  

Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

[Justin Steinberg]

Sure, thanks Ryan.   SR 634454765

On Mon, Apr 13, 2015 at 3:04 PM, Ryan LaFountain (rlafount) 
rlafo...@cisco.com wrote:

   Hi Justin,

  Can you send me the TAC case please? I'm curious to know why they told
 you it requires a CAD upgrade...

   Thank you,

 Ryan LaFountain
 Unified Contact Center
 Cisco Services
 Direct: +1 919 392 9898
 Hours: M - F 9:00am - 5:00pm Eastern Time

   From: Justin Steinberg
 Date: Monday, April 13, 2015 at 3:02 PM
 To: Abhiram Kramadhati (akramadh)
 Cc: voip puck

 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   I opened up a TAC case to get early access to the COP file and found
 that the COP file makes modifications to CAD, which requires a CAD upgrade
 on all the clients.   That's really disappointing because the entire goal
 here is to move off of CAD.  Having to get all the desktop teams (4 desktop
 teams in this environment) to upgrade CAD another time, just to
 decommission CAD is really alot of wasted time and frustration.

  TAC is checking with BU to see if there is anyway to avoid the CAD
 upgrade.

  Justin

 On Thu, Apr 9, 2015 at 8:46 PM, Abhiram Kramadhati (akramadh) 
 akram...@cisco.com wrote:

  Hi Justin/all,

  The support for mixed mode is almost ready and you should see it on CCO
 in no more than 2 weeks. Will update if there are any changes.

   Regards,
 Abhiram Kramadhati
 Technical Solutions Manager, CBABU
 CCIE Voice # 40065

   From: akramadh akram...@cisco.com
 Date: Tuesday, 7 April 2015 8:52 am
 To: Justin Steinberg jsteinb...@gmail.com

 Cc: Josh Warcop j...@warcop.com, Lelio Fulgenzi le...@uoguelph.ca,
 voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Hi Justin,

  Allow me till Wednesday, I will have some confirmation on the exact
 dates although I can confirm that it right around the corner. BTW, I can
 help you get the cop file for your customer immediately because it is being
 given out on a per-customer basis – so the support can be given for your
 customer now if needed and you don’t have to wait for the announcement of
 the support.

   Regards,
 Abhiram Kramadhati
 Technical Solutions Manager, CBABU
 CCIE Voice # 40065

   From: Justin Steinberg jsteinb...@gmail.com
 Date: Monday, 6 April 2015 6:27 am
 To: akramadh akram...@cisco.com
 Cc: Josh Warcop j...@warcop.com, Lelio Fulgenzi le...@uoguelph.ca,
 voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

Abhiram,

  Is there any update on the COP file for official support for Finesse and
 CAD on the same CCX cluster?The design guide still says not supported
 for running both CAD  Finesse.

  Justin

 On Mon, Jan 19, 2015 at 5:29 PM, Abhiram Kramadhati (akramadh) 
 akram...@cisco.com wrote:

  Hi all,

  The support for mixed-mode is expected to be announced in the coming
 ways. It will be starting with 10.6 and the support can be claimed by
 installing a COP file. The details will come out soon, but in summary it
 will be in 10.6. Also, there will be a white-paper released regarding the
 migration plan and best practices. Some of the things that come to my mind
 straight away are:

- Voice is supported on mixed mode – you can have some teams on
either CAD or Finesse handling calls
- Chat and email agents have to be migrated completely onto Finesse
- CAD supervisors only to monitor CAD teams and so on for Finesse
- 1 team = 1 type of agent desktop

 There are some limitations too and they will be in the document which
 will be published. However, if you can think about any scenario you would
 like to discuss and get some feedback – please feel free to reach out to
 me. Or you can put it out here in the forum, Ryan or myself can brainstorm
 with you.

   *Abhiram Kramadhati*
 Technical Solutions Manager
 Customer Solutions Success team, CBABU
 akram...@cisco.com

 CCIE Voice - 40065

 *Cisco Systems Australia Pty Limited*
 The Forum
 201 Pacific Highway
 2065
 St Leonards
 Australia
 Cisco.com http://www.cisco.com/web/AU/
Think before you print.

 This email may contain confidential and privileged material for the sole
 use of the intended recipient. Any review, use, distribution or disclosure
 by others is strictly prohibited. If you are not the intended recipient (or
 authorized to receive for the recipient), please contact the sender by
 reply email and delete all copies of this message.

 Please click here
 http://www.cisco.com/web/about/doing_business/legal/cri/index.html
 for Company Registration Information.

   From: Josh Warcop j...@warcop.com
 Date: Tuesday, 20 January 2015 6:19 am
 To: Lelio Fulgenzi le...@uoguelph.ca, voip puck 
 cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Both can be activated but it is a requirement that a team be on CAD
 or Finesse. Essentially giving a 'Team' based migration approach from CAD
 to Finesse.
   --
 From: Lelio Fulgenzi

Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

[Justin Steinberg]

I opened up a TAC case to get early access to the COP file and found that
the COP file makes modifications to CAD, which requires a CAD upgrade on
all the clients.   That's really disappointing because the entire goal here
is to move off of CAD.  Having to get all the desktop teams (4 desktop
teams in this environment) to upgrade CAD another time, just to
decommission CAD is really alot of wasted time and frustration.

TAC is checking with BU to see if there is anyway to avoid the CAD upgrade.


Justin

On Thu, Apr 9, 2015 at 8:46 PM, Abhiram Kramadhati (akramadh) 
akram...@cisco.com wrote:

  Hi Justin/all,

  The support for mixed mode is almost ready and you should see it on CCO
 in no more than 2 weeks. Will update if there are any changes.

   Regards,
 Abhiram Kramadhati
 Technical Solutions Manager, CBABU
 CCIE Voice # 40065

   From: akramadh akram...@cisco.com
 Date: Tuesday, 7 April 2015 8:52 am
 To: Justin Steinberg jsteinb...@gmail.com

 Cc: Josh Warcop j...@warcop.com, Lelio Fulgenzi le...@uoguelph.ca,
 voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Hi Justin,

  Allow me till Wednesday, I will have some confirmation on the exact
 dates although I can confirm that it right around the corner. BTW, I can
 help you get the cop file for your customer immediately because it is being
 given out on a per-customer basis – so the support can be given for your
 customer now if needed and you don’t have to wait for the announcement of
 the support.

   Regards,
 Abhiram Kramadhati
 Technical Solutions Manager, CBABU
 CCIE Voice # 40065

   From: Justin Steinberg jsteinb...@gmail.com
 Date: Monday, 6 April 2015 6:27 am
 To: akramadh akram...@cisco.com
 Cc: Josh Warcop j...@warcop.com, Lelio Fulgenzi le...@uoguelph.ca,
 voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

Abhiram,

  Is there any update on the COP file for official support for Finesse and
 CAD on the same CCX cluster?The design guide still says not supported
 for running both CAD  Finesse.

  Justin

 On Mon, Jan 19, 2015 at 5:29 PM, Abhiram Kramadhati (akramadh) 
 akram...@cisco.com wrote:

  Hi all,

  The support for mixed-mode is expected to be announced in the coming
 ways. It will be starting with 10.6 and the support can be claimed by
 installing a COP file. The details will come out soon, but in summary it
 will be in 10.6. Also, there will be a white-paper released regarding the
 migration plan and best practices. Some of the things that come to my mind
 straight away are:

- Voice is supported on mixed mode – you can have some teams on
either CAD or Finesse handling calls
- Chat and email agents have to be migrated completely onto Finesse
- CAD supervisors only to monitor CAD teams and so on for Finesse
- 1 team = 1 type of agent desktop

 There are some limitations too and they will be in the document which
 will be published. However, if you can think about any scenario you would
 like to discuss and get some feedback – please feel free to reach out to
 me. Or you can put it out here in the forum, Ryan or myself can brainstorm
 with you.

   *Abhiram Kramadhati*
 Technical Solutions Manager
 Customer Solutions Success team, CBABU
 akram...@cisco.com

 CCIE Voice - 40065

 *Cisco Systems Australia Pty Limited*
 The Forum
 201 Pacific Highway
 2065
 St Leonards
 Australia
 Cisco.com http://www.cisco.com/web/AU/
Think before you print.

 This email may contain confidential and privileged material for the sole
 use of the intended recipient. Any review, use, distribution or disclosure
 by others is strictly prohibited. If you are not the intended recipient (or
 authorized to receive for the recipient), please contact the sender by
 reply email and delete all copies of this message.

 Please click here
 http://www.cisco.com/web/about/doing_business/legal/cri/index.html for
 Company Registration Information.

   From: Josh Warcop j...@warcop.com
 Date: Tuesday, 20 January 2015 6:19 am
 To: Lelio Fulgenzi le...@uoguelph.ca, voip puck 
 cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Both can be activated but it is a requirement that a team be on CAD or
 Finesse. Essentially giving a 'Team' based migration approach from CAD to
 Finesse.
   --
 From: Lelio Fulgenzi le...@uoguelph.ca
 Sent: 1/19/2015 2:04 PM
 To: voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Thanks for everyone's help on this, especially the Cisco folk who are
 going back to check for docs and publishing dates, etc.

 I'm glad I posted this and started the conversation.


  ---
 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

Re: [cisco-voip] Cisco Unified Border Element Session Licensing

[Justin Steinberg]

Cant cube licenses be bought ala carte ?  Do you even have to provide a
router serial ?
On Apr 11, 2015 9:39 AM, Mark Holloway m...@markholloway.com wrote:

 I don’t think CUBE licensing is perpetual. You should call Cisco to get a
 definite answer but the folks I know had to rip and replace.

 On Apr 10, 2015, at 4:28 PM, Countryman, Edward 
 edward.country...@presencehealth.org wrote:

  Does anyone know if I would be allowed to move SUBE session licensing
 from my 3925 router to my new 3945E router?



 We are upgrading to the larger router to support more concurrent calls
 from our provider, but I don’t want to abandon the $30,000 plus investment
 I already have in session licensing on the 3925.


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



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


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Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

[Justin Steinberg]

Thanks for that link Anthony.   It just seems like the support for mixed
mode is really taking a long time.  This was a feature in all of the
marketing slides for 10.6 and it is now April with no publicly released COP.

On Mon, Apr 6, 2015 at 10:31 AM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 *Thanks for your question. The support for co-residence of CAD and
 Finesse will be available in UCCX 10.6. We are currently in the
 controlled-release phase - what this means is that the support for the
 mixed mode is only being given out to a few customers whose migration plan
 is reviewed by internal teams and a COP file is provided for that customer
 only. This is to ensure that any feedback is incorporated before the
 official support is announced. The supported will be announced soon and the
 COP file which provides the support will be available for all customers in
 version 10.6.*

 Source:
 https://supportforums.cisco.com/discussion/12443556/ask-expert-finesse-uccx-next-generation-agent-desktop-and-transitioning-cad#comment-10335836

 On Mon, Apr 6, 2015 at 9:21 AM Brian Meade bmead...@vt.edu wrote:

 I've got it running for one customer.  Not sure where we got the file
 from though.  File name is ciscouccx.1061.MixedMode.7.cop.sgn.  I'm
 guessing they haven't released it publicly yet as I don't see it on CCO.

 On Sun, Apr 5, 2015 at 3:27 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 Abhiram,

 Is there any update on the COP file for official support for Finesse and
 CAD on the same CCX cluster?The design guide still says not supported
 for running both CAD  Finesse.

 Justin

 On Mon, Jan 19, 2015 at 5:29 PM, Abhiram Kramadhati (akramadh) 
 akram...@cisco.com wrote:

  Hi all,

  The support for mixed-mode is expected to be announced in the coming
 ways. It will be starting with 10.6 and the support can be claimed by
 installing a COP file. The details will come out soon, but in summary it
 will be in 10.6. Also, there will be a white-paper released regarding the
 migration plan and best practices. Some of the things that come to my mind
 straight away are:

- Voice is supported on mixed mode – you can have some teams on
either CAD or Finesse handling calls
- Chat and email agents have to be migrated completely onto Finesse
- CAD supervisors only to monitor CAD teams and so on for Finesse
- 1 team = 1 type of agent desktop

 There are some limitations too and they will be in the document which
 will be published. However, if you can think about any scenario you would
 like to discuss and get some feedback – please feel free to reach out to
 me. Or you can put it out here in the forum, Ryan or myself can brainstorm
 with you.

   *Abhiram Kramadhati*
 Technical Solutions Manager
 Customer Solutions Success team, CBABU
 akram...@cisco.com

 CCIE Voice - 40065

 *Cisco Systems Australia Pty Limited*
 The Forum
 201 Pacific Highway
 2065
 St Leonards
 Australia
 Cisco.com http://www.cisco.com/web/AU/
Think before you print.

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 sole use of the intended recipient. Any review, use, distribution or
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   From: Josh Warcop j...@warcop.com
 Date: Tuesday, 20 January 2015 6:19 am
 To: Lelio Fulgenzi le...@uoguelph.ca, voip puck 
 cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Both can be activated but it is a requirement that a team be on CAD
 or Finesse. Essentially giving a 'Team' based migration approach from CAD
 to Finesse.
  --
 From: Lelio Fulgenzi le...@uoguelph.ca
 Sent: 1/19/2015 2:04 PM
 To: voip puck cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] FYI: UCCx v9.0 EOL announced...

   Thanks for everyone's help on this, especially the Cisco folk who
 are going back to check for docs and publishing dates, etc.

 I'm glad I posted this and started the conversation.


 ---
 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: *Mike mik...@msn.com
 *To: *Charles Goldsmith wo...@justfamily.org
 *Cc: *Anthony Holloway avholloway+cisco-v...@gmail.com, Lelio
 Fulgenzi le...@uoguelph.ca, Kevin Przybylowski 
 kev...@advancedtsg.com, voip puck cisco-voip@puck.nether.net
 *Sent: *Monday, January 19, 2015 1:25:41 PM
 *Subject: *RE: [cisco-voip] FYI: UCCx v9.0 EOL announced...

  I can find out tomorrow. I was told it was supported

Re: [cisco-voip] (no subject)

[Justin Steinberg]

So we all agree that expanding an existing vdisk isn't the same as
installing from a larger OVA template.

That being said, the documentation about multiple vdisk systems seems to no
longer apply.  Starting with the 9.1 OVA templates, even the 10k user
template is a single 1x110GB disk. 9.0 was the last CM OVA template
version to have 2x80GB vdisk config for the 7500 and 10k user templates.

Does anyone know whether there are any other settings such as DB
performance settings that are different between OVA templates, or is it
just simply vCPU, vRAM, vDISK size. If it is just the vDisk size, I'm
leading towards staying on my current expanded vdisk template until a time
in the future where the active/user partition needs to be larger to
accommodate a newer software version.

Justin


On Mon, Mar 23, 2015 at 2:17 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Ryan, you replied to Tim, but I think Justin is the one who really needs
 to read your reply.  His method to move from one OVA to the next may not
 have resulted in what he expected.


 On Mon, Mar 23, 2015 at 12:45 PM Ryan Ratliff (rratliff) 
 rratl...@cisco.com wrote:

  It’s expected that only the common partition grows if you expand the
 vDisk.  The feature is there because some versions of UCM require more
 space available in that partition than others.


 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/10_0_1/CUCM_BK_U4214F9D_00_upgrade-guide-cucm-100/CUCM_BK_U4214F9D_00_upgrade-guide-cucm-100_chapter_011.html#CUCM_TK_C9AFC8CC_00

   You need to change the disk size changes only if you need additional
 space to complete the upgrade. The disk space requirements are specified in
 the Readme file for the OVA template.

 Expanding the disk size to add space to the common partition will not
 increase the user capacity of your system. If you need to extend the user
 capacity of your system, you must migrate from a single-disk to a
 multi-disk virtual machine

 So if you really need to go from a 2500 user deployment to a 7500 user
 one you should DRS, rebuild, restore so your active and inactive partitions
 get sized accordingly.

  -Ryan

  On Mar 23, 2015, at 10:28 AM, Tim Frazee tfra...@gmail.com wrote:

  unknown. installer still does a hardware check.

  while testing my moh problem, I used an 8.6 ova to deploy the old 10k
 template, made the one OS tweak to the image, then installed 10.5(2) on it.
 successful install.

 On Sun, Mar 22, 2015 at 9:43 AM, NateCCIE . natec...@gmail.com wrote:

 I wonder what the difference is between expanding and installing on 120G
 from the beginning, since there is the report of it only growing the common
 partition.

 On Sat, Mar 21, 2015 at 9:03 AM, Tim Frazee tfra...@gmail.com wrote:

  the resize cop file is for 9x only, 10 has it built in. I'm running
 around with a tac case to address a stock 9x or 10x 7.5 to 10k user build
 that results with a 110G disk. doesn't leave much space for those 500 moh
 sources.

  if you shut the image down and increase from 110 to at least 112G, the
 boot process grows the common partition out.

  as a standard, any of our 10x installs for large clients, we are
 growing the disk out to 120G just to be safe.

 On Fri, Mar 20, 2015 at 3:50 PM, Erick erick...@gmail.com wrote:

  The VMware disk reallocation worked for us also going from 80gig  to
 110gig for 10.5. Were on. 9.1 prior.

  The readme in the download link Is pretty good but doesn't say
 outright what to increase it to.

  High level steps ,

  Make sure you have a good backup

  Install the cop file
 Shutdown the vm
 Change virtual disk from 80G to 110G
 Save /OK settings
 Power on VM

  It will reboot a few times while extending disk then come up fine /
 normal .


 Sent from my iPhone

 On Mar 20, 2015, at 9:34 AM, Justin Steinberg jsteinb...@gmail.com
 wrote:

   So in the CUCM 10.5 download section for the Utilities, it seems to
 have combined the common cleanup COP file and the VMware Disk Size
 Reallocation.

  There is a COP file title '*VMware Disk Size Reallocation COP file' *but
 the actual file is *ciscocm.free_common_space_v1.3.k3.cop.sgn*

  The actual reallocation cop file isn't part of the CUCM 10.5
 download, I had to go back into an older version to file that COP.  So 
 that
 is why I was thinking in 10.5 all you would need to do is change the size
 of the vDisk in VMware and restart CUCM 10.5.

  Is there an official document on the process to follow for this
 change ?

  Justin



 On Fri, Mar 20, 2015 at 10:12 AM, Roger Wiklund 
 roger.wikl...@gmail.com wrote:

 I have.

 Went from 2500 to 7500 on CUCM 10.5(1).

 You need to download the VMware Disk Size Reallocation COP file for
 10.5. Worked like a charm.


 http://www.cisco.com/web/software/282204704/18582/CleanupCommonCOPfilev1.3.pdf

 http://www.cisco.com/web/software/282204704/18582/ciscocm.vmware_disk_size_reallocation_v1.0.pdf

 On Fri, Mar 20, 2015 at 2:28 PM, Justin Steinberg 
 jsteinb

Re: [cisco-voip] (no subject)

[Justin Steinberg]

I pulled the trigger on this last night with CM 10.5.2, migrating two UCMs
changing from the 2500 template to the 7500 template.  There was really
nothing to it, other than shutting down the VM and increasing the vDisk
from 80 GB to 110 GB.

Upon startup the software automatically detected the vdisk change and ran
an 'Expanding Disk' script.   The new disk space was given to the common
partition.

I did compare a new 10.5 7500 template with the expanded 2500 to 7500
template.   The difference I noticed was that on a true fresh install of
7500 user template the active and inactive software partitions have 20GB
allocated.   The 2500 had about 14 GB to these partitions.  Expanding the
2500 user template to 7500 user only increased the common partition, the
software partitions remained at 14GB.

Justin
On Mar 21, 2015 11:03 AM, Tim Frazee tfra...@gmail.com wrote:

 the resize cop file is for 9x only, 10 has it built in. I'm running around
 with a tac case to address a stock 9x or 10x 7.5 to 10k user build that
 results with a 110G disk. doesn't leave much space for those 500 moh
 sources.

 if you shut the image down and increase from 110 to at least 112G, the
 boot process grows the common partition out.

 as a standard, any of our 10x installs for large clients, we are growing
 the disk out to 120G just to be safe.

 On Fri, Mar 20, 2015 at 3:50 PM, Erick erick...@gmail.com wrote:

 The VMware disk reallocation worked for us also going from 80gig  to
 110gig for 10.5. Were on. 9.1 prior.

 The readme in the download link Is pretty good but doesn't say outright
 what to increase it to.

 High level steps ,

 Make sure you have a good backup

 Install the cop file
 Shutdown the vm
 Change virtual disk from 80G to 110G
 Save /OK settings
 Power on VM

 It will reboot a few times while extending disk then come up fine /
 normal .


 Sent from my iPhone

 On Mar 20, 2015, at 9:34 AM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 So in the CUCM 10.5 download section for the Utilities, it seems to have
 combined the common cleanup COP file and the VMware Disk Size Reallocation.

 There is a COP file title '*VMware Disk Size Reallocation COP file' *but
 the actual file is *ciscocm.free_common_space_v1.3.k3.cop.sgn*

 The actual reallocation cop file isn't part of the CUCM 10.5 download, I
 had to go back into an older version to file that COP.  So that is why I
 was thinking in 10.5 all you would need to do is change the size of the
 vDisk in VMware and restart CUCM 10.5.

 Is there an official document on the process to follow for this change ?

 Justin



 On Fri, Mar 20, 2015 at 10:12 AM, Roger Wiklund roger.wikl...@gmail.com
 wrote:

 I have.

 Went from 2500 to 7500 on CUCM 10.5(1).

 You need to download the VMware Disk Size Reallocation COP file for
 10.5. Worked like a charm.


 http://www.cisco.com/web/software/282204704/18582/CleanupCommonCOPfilev1.3.pdf

 http://www.cisco.com/web/software/282204704/18582/ciscocm.vmware_disk_size_reallocation_v1.0.pdf

 On Fri, Mar 20, 2015 at 2:28 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:
  Has anyone successfully expanded the virtual disk size of CUCM VMs
 without
  rebuild/DRS?
 
 
 
  I have an install where CM 10.5 is using the 2500 user template and we
 want
  to increase to 7500 users.  The 2500 OVA is 1 vCPU, 4GB, 1x80GB.
 The 7500
  OVA is 2vCPU, 6 GB, 1x110GB.In the past, the older 7500 user CM
 versions
  had two virtual 80 GB disks, however since 9.1 the 7500 user is a
 single 110
  GB disk.   It seems like with a single virtual disk it would be easier
 to
  expand an existing VM without rebuild.
 
 
 
  There are several bugs on the topic:
 
  https://tools.cisco.com/bugsearch/bug/CSCug63058
 
  https://tools.cisco.com/bugsearch/bug/CSCuc58936
 
 
 
  In older CM versions there was a COP file to assist with allowing the
 VM to
  use more disk when the vdisk was increased.  However, now I believe
 that it
  is just built in to CM to use more disk on reboots if it detects a
 vdisk
  change instead of needing to run the OVA.
 
 
 
  There is still conflicting documentation on the topic, so I will
 probably
  open a TAC case but curious if anyone has dealt with this before?
 
 
  Justin
 
 
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  https://puck.nether.net/mailman/listinfo/cisco-voip
 


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[cisco-voip] (no subject)

[Justin Steinberg]

Has anyone successfully expanded the virtual disk size of CUCM VMs without
rebuild/DRS?



I have an install where CM 10.5 is using the 2500 user template and we want
to increase to 7500 users.  The 2500 OVA is 1 vCPU, 4GB, 1x80GB.The
7500 OVA is 2vCPU, 6 GB, 1x110GB.In the past, the older 7500 user CM
versions had two virtual 80 GB disks, however since 9.1 the 7500 user is a
single 110 GB disk.   It seems like with a single virtual disk it would be
easier to expand an existing VM without rebuild.



There are several bugs on the topic:

https://tools.cisco.com/bugsearch/bug/CSCug63058

https://tools.cisco.com/bugsearch/bug/CSCuc58936



In older CM versions there was a COP file to assist with allowing the VM to
use more disk when the vdisk was increased.  However, now I believe that it
is just built in to CM to use more disk on reboots if it detects a vdisk
change instead of needing to run the OVA.



There is still conflicting documentation on the topic, so I will probably
open a TAC case but curious if anyone has dealt with this before?


Justin
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Re: [cisco-voip] USB Tokens

[Justin Steinberg]

on this subject, is it possible to take a mixed mode v8 or v9 (signed by
USB tokens) and upgrade to v10 and then use the new CLI command to update
the CTL and move away from the need for the tokens?

Is there anything to do other than run the CLI command to update the CTL ?
  Do I then no longer need the tokens?

On Thu, Mar 12, 2015 at 11:22 AM, Jason Burns burns.ja...@gmail.com wrote:

 You can use the same USB tokens for the two systems.

 I would recommend this because it makes it a bit easier to move phones
 around between the two systems.

 If both systems are under the same domain of control, and same level of
 trust, there is no problem using the tokens again.

 If one system has phones that should NEVER register to the other system
 for security reasons - then I would recommend using separate tokens.

 --
 Jason Burns

 On Thu, Mar 12, 2015 at 11:09 AM, Reto Gassmann v...@mrga.ch wrote:

 hello Group

 We run a cucm 9.1(2) and have the Phones authenticated. To do that we
 used the two USB Tokens and the CTL Client.

 Now we set up a new Testserver and want to do some tests with certificat
 renwal and upgrades to 10.5. can I use the same two USB Tokens I used for
 the production system or do I have to order two new USB Tokens for the
 Testserver?

 Thanks for helping out
 Regards Reto

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Re: [cisco-voip] Jabber, MRA, and Free Public WiFi

[Justin Steinberg]

good write  up.

I wonder what would happen if the _collab-edge._tls SRV returned port 443
with an internet firewall in front of Expressway translating 443 to 8443.
I wonder whether the Jabber clients read the port from the SRV or whether
they have 8443 hardcoded.

I'll try to test that on my next deployment.

On Fri, Feb 27, 2015 at 2:02 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 All,

 Just a heads up to my fellow techs, I am at Caribou Coffee today and my
 Jabber will not sign in.

 The user experience is as follows: Jabber discovers MRA successfully, but
 when trying to authenticate it sends an auth request to:

 https://collab-edge.company.com:8443/oauthcb

 The logs show that an HTTP timeout occurs: (Found
 in C:\Users\you\AppData\Local\Cisco\Unified
 Communications\Jabber\CSF\Logs\csf-unified.log)

 2015-02-27 09:14:40,081 INFO  [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1163)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - *-* Making HTTP request
 to: https://collab-edge.company.com:8443/oauthcb [3]
 2015-02-27 09:14:40,081 INFO  [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1738)] [csf.httpclient]
 [http::CurlHeaders::CurlHeaders] - Number of Request Headers : 1
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1345)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - Checking for proxy
 information for request [3] ...
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [ts\csf-netutils\src\http\Request.cpp(83)] [csf.httpclient]
 [http::Request::getProxy] - No Proxy will be used per configuration of this
 request
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1429)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - No proxy information
 available [3].
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1502)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - Setting connect timeout value
 in milliseconds to : 1
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1511)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - Setting transfer timeout
 value in milliseconds to : 3
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1514)] [csf.httpclient]
 [http::CurlHttpUtils::configureEasyRequest] - HTTP Request Configured.
 2015-02-27 09:14:40,081 DEBUG [0x0af0]
 [ls\src\http\BasicHttpClientImpl.cpp(482)] [csf.httpclient]
 [http::performCurlRequest] - About to perform curl connection request...
 2015-02-27 09:14:40,096 DEBUG [0x0af0]
 [netutils\src\http\CurlHttpUtils.cpp(307)] [csf.httpclient]
 [http::CurlHttpUtils::logPhaseData] - Pre connect phase. Resolved IP:
 23.23.23.23
 2015-02-27 09:14:50,079 DEBUG [0x0af0]
 [etutils\src\http\CurlHttpUtils.cpp(1679)] [csf.httpclient]
 [http::CurlHttpUtils::logOperationTiming] - Network IO timestamps: [name
 lookup = 0.016 ; connect = 0 ; ssl connect = 0 ; pre-transfer = 0 ;
 start-transfer = 0 ; total = 10 ; redirect = 0]
 2015-02-27 09:14:50,079 INFO  [0x0af0]
 [ls\src\http\CurlAnswerEvaluator.cpp(117)] [csf.httpclient]
 [http::CurlAnswerEvaluator::curlCodeToResult] - curlCode=[28] error
 message=[Connection timed out after 1 milliseconds]
 result=[CONNECTION_TIMEOUT_ERROR] fips enabled=[false]
 2015-02-27 09:14:50,079 INFO  [0x0af0]
 [ls\src\http\BasicHttpClientImpl.cpp(410)] [csf.httpclient]
 [http::executeImpl] - *-* HTTP response from:
 https://collab-edge.company.com:8443/oauthcb [3] - 0.
 2015-02-27 09:14:50,079 ERROR [0x0af0]
 [ls\src\http\BasicHttpClientImpl.cpp(414)] [csf.httpclient]
 [http::executeImpl] - There was an issue performing the call to
 curl_easy_perform: CONNECTION_TIMEOUT_ERROR
 2015-02-27 09:14:50,079 DEBUG [0x0af0]
 [etutils\src\http\HttpRequestData.cpp(90)] [csf.httpclient]
 [http::HttpRequestData::returnEasyCURLConnection] - Returning borrowed
 EasyCURLConnection from request : 3
 2015-02-27 09:14:50,079 DEBUG [0x0af0]
 [utils\adapters\EdgeUtilsAdapter.cpp(255)] [csf.netutils.adapters]
 [netutils::adapters::EdgeUtilsAdapter::isRequestTransformed] -
 isRequestTransformed: result:0. originalPath: '/oauthcb' pathFromUrlUsed:
 '/oauthcb'.
 2015-02-27 09:14:50,079 DEBUG [0x0af0]
 [tutils\src\http\HttpRequestData.cpp(105)] [csf.httpclient]
 [http::HttpRequestData::~HttpRequestData] - Destroying instance of Request
 data, with request: 3

 And then I get the message in Jabber which says Cannot Communicate with
 the Server

 [image: Inline image 1]

 It turns out that if I try to telnet to collab-edge.company.com on port
 8443, it fails:

 [image: Inline image 2]

 And a Wireshark reveals that the TCP three way handshake never happens,
 with two TCP SYN re-transmits, before finally timing out.

 [image: Inline image 3]

 Interestingly, this free WiFi network does not prevent me from accessing
 the standard HTTPS port of 443, and I can actually 

Re: [cisco-voip] UCCX Virtual 9.0(2) to 10.6(1) Upgrade

[Justin Steinberg]

I just went through a CUC 8.5(1)su7 to 10.5(2) upgrade last night.  I tried
to upgrade before the vRAM increase from 4 GB to 6GB and the upgrade failed
about 5 minutes in due to 'unsupported hardware'.   Shutdown the VMs,
increased the vRAM to 6GB, booted it back up and the upgrade was fine.

On Fri, Feb 20, 2015 at 10:29 AM, Ryan LaFountain (rlafount) 
rlafo...@cisco.com wrote:

   Hi All,

  We really should have this better documented as we don’t follow CUCM 1:1
 as Anthony pointed out in the areas of underlying VOS versions and testing.
 I’ve field CSCut04158 for the doc-wiki to be updated. It should be visible
 soon.

   Thank you,

 Ryan LaFountain
 Unified Contact Center
 Cisco Services
 Direct: +1 919 392 9898
 Hours: M - F 9:00am - 5:00pm Eastern Time

   From: Anthony Holloway
 Date: Friday, February 20, 2015 at 8:49 AM
 To: Andrew Grech, Daniel Pagan, Cisco VoIP Group
 Subject: Re: [cisco-voip] UCCX Virtual 9.0(2) to 10.6(1) Upgrade

   I opened a PDI ticket on this, because CUCM and CUC differ on when you
 should change the VM settings.  Here's what I wrote to PDI:

  I am planning an upgrade from 8.6(2a)SU2 to 10.5(1)SU1.

  I am looking at the pre-upgrade tasks of this document:


 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/10_0_1/CUCM_BK_U4214F9D_00_upgrade-guide-cucm-100.html


  It states that I need to modify the VM guest settings, to include the OS
 version, before I do the upgrade.

  However, the CUC upgrade guide (which should largely be the same as CUCM
 considering the shared ISO image), states to do the same but Post-Upgrade,
 not before. That guide is here:


 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/upgrade/guide/10xcucrugx/10xcucrug010.html#16383


  I'm wondering if it really matters when I change these settings on the
 VM: pre or post upgrade, just as long as I do it before VMTools is updated.
 Or might there be a problem with the documentation?

  And their reply was simply, follow the documentation, because each team
 tests their upgrades differently, and how they documented it is how it
 worked for them.

  That also means that for apps like UCCX, if you don't see anywhere,
 where it says to change the OS from 32 to 64 bit, then you shouldn't.
 You'd be making an assumption that the UCCX team moved their OS version,
 just because the CUCM team did.

  Lastly, for RAM and CPU increases, I did those pre-upgrade, with the
 thoughts that: A) It's not unsupported to have more resources available to
 the VM, and B) It might make the upgrade process faster.

  Those are my thoughts on the subject.

 On Fri Feb 20 2015 at 7:10:57 AM Andrew Grech agrec...@gmail.com wrote:

 Thanks Dan I just checked the OVAs and I do need to change the vNIC I
 remember doing this for another product but it wasn't documented unlike ccx
 unless I'm missing it.

 My other concern is my signed certificates but I have access to the CA so
 its not a game changer if I need to reissue.

 I have thought about running sandbox servers up in my lab loading a drs
 then upgrading. I could then disconnect the NICs of the current vms and
 move the sandbox servers into production. This would provide an easy
 rollback. Anyone have experience with this? Note I won't be cloning the
 sandbox servers from production but rather running them up from scratch.

 On 20/02/2015 10:28 PM, Daniel Pagan dpa...@fidelus.com wrote:
 
  Based on the UCCX tech note I found, I’d say the shutdown and final
 modifications should be made after a full switch version is performed and
 you’re running the upgrade release on your active partitions.
 
 
 
  This differs from the official word from Cisco on CUCM, which is to
 adjust the virtual machine resources to match the OVA template
 specifications before the upgrade is performed. However, also according to
 official documentation, changing the vNIC to VMXNET3 through PowerCLI
 should be done post-upgrade.
 
 
 
  Personally I’ve seen all these modifications being performed after the
 upgrade is completed and version switched with no negative impact
 whatsoever, but official documentation and TAC’s stance says the virtual
 machine settings should be modified first, followed by the upgrade, then
 wrapped up with the vNIC change to VMXNET3. I can provide documentation if
 needed - it seems that UCCX documentation differs.
 
 
 
  Also don’t forget to set the OS security settings to “permissive” if
 you wish to auto-update the VMware Tools on the virtual machines.
 
 
 
  - Dan
 
 
 
  From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf
 Of Andrew Grech
  Sent: Thursday, February 19, 2015 11:43 PM
  To: Cisco VoIP Group
  Subject: [cisco-voip] UCCX Virtual 9.0(2) to 10.6(1) Upgrade
 
 
 
  Hi,
 
 
 
  I'm preparing an upgrade of uccx 9.0(2) to 10.6(1)
 
 
 
  When would be be the best time to adjust the OVA file?
 
 
 
  I need to change the RAM and Guest OS to RHEL6 (although Guest OS
 change step is not documented)
 
 
 
 
 
  

Re: [cisco-voip] Informix Drivers for CCX

[Justin Steinberg]

what wasn't working about the drivers ?  I've used them before without any
issues.   I suspect maybe it was a 32/64 bit issue.  I've also seen some
weird behavior where the odbc gui is just finicky about how you enter the
data for it to save properly.

On Wed, Feb 4, 2015 at 2:23 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  I had to open a TAC case because our wallboard stopped working. The ODBC
 drivers in the 3.70 and 4.1 informix were not working. The TAC solution was
 to somehow find me an ancient 3.0 version of the Informix SDK. It did start
 working, but it just doesn’t make sense and now our wallboard doesn’t work.
 That’s because of code differences in the SDK. We are trying to correct,
 but something seems off…



 Maybe my google wizardry level is down, but is there somewhere that lists
 the acceptable SDK versions for each CCX version? This client is on 10.0
 SU1… I don’t have that running elsewhere so I can’t compare.



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

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



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Re: [cisco-voip] Prep CAD Client Update Ahead of time?

[Justin Steinberg]

it appears that 9.0(2)SU2 ISO is showing CAD 8.x as well.

I am planning an upgrade from 9.0(2)SU1 to SU2 and trying to get the MSI in
advance.  I do have access to a second 9.0(2)SU2 system, does anyone know
if I download the CAD msi file from one cluster and edit the MSI installer
in Orca if I can make the changes ?I see in the PROPERTY section of
Orca the LDAPHOSTONE and LDAPHOSTTWO settings which specify the IP of the
server.

Does anyone know if those are the only two values I need to modify if I
take a MSI from one cluster to prepare for an upgrade to another cluster ?

On Wed, Oct 1, 2014 at 5:46 PM, Ryan LaFountain (rlafount) 
rlafo...@cisco.com wrote:

  Hi Guys,

  I’ve filed CSCur11033 for this in 10.0 and 10.5. It looks like we
 updated the ISO build process in 9.0(2) to resolve CSCuh47596 but we
 haven’t updated it since. Hence the 10.0 and 10.5 Preconfiguration Tools
 installing the 9.0(2) version of CAD.

  It should be available on Bug Search Tool soon. Thanks for bringing this
 to my attention.

 Thank you,

 Ryan LaFountain
 Unified Contact Center
 Cisco Services
 Direct: +1 919 392 9898
 Hours: M - F 9:00am - 5:00pm Eastern Time

  On Oct 1, 2014, at 9:28 AM, Ryan LaFountain (rlafount) 
 rlafo...@cisco.com wrote:

  Hi All,

  I’ll look into this and report back with any new bugs I file or any
 resolution.

  Apologies for this not working correctly.

 Thank you,

 Ryan LaFountain
 Unified Contact Center
 Cisco Services
 Direct: +1 919 392 9898
 Hours: M - F 9:00am - 5:00pm Eastern Time

  On Oct 1, 2014, at 9:11 AM, Clifford McGlamry cmcgla...@forsythe.com
 wrote:

   Yep, that’s the one.  It also affects all the 10.0 and 10.5 builds.
 It’s maddening that they won’t take 5 minutes and make the image
 correctly.  But using the files we grabbed off the deployed lab image and
 the 3 files that are in the root of the installer directory, we were able
 to assemble what SHOULD have been there and get them created.


 How are you these days Chris?  Haven’t talked to you in a while.


 *Cliff McGlamry, CCIE Collaboration #24757*
 *Master Consultant*
 (678) 934-0348   direct
 (404) 969-9806   mobile
 (678) 934-0448   fax
 cmcgla...@forsythe.com



 *Forsythe Technology, Inc.*
 400 Interstate North Parkway

 Suite 860
 Atlanta, GA 30339
 http://www.forsythe.com









 image001.gif




 *From:* Chris Clouse [mailto:chris.clo...@cdw.com chris.clo...@cdw.com]
 *Sent:* Wednesday, October 01, 2014 8:57 AM
 *To:* Clifford McGlamry; Matthew Loraditch; cisco-voip@puck.nether.net
 *Subject:* RE: Prep CAD Client Update Ahead of time?



 I know there are several bugs around this process.  I know this one covers
 9.x train.  CSCuh47596


 *~Chris*


 *From:* Clifford McGlamry [mailto:cmcgla...@forsythe.com
 cmcgla...@forsythe.com]
 *Sent:* Tuesday, September 30, 2014 4:42 PM
 *To:* Chris Clouse; Matthew Loraditch; cisco-voip@puck.nether.net
 *Subject:* RE: Prep CAD Client Update Ahead of time?



 Tried this, and it doesn’t work.  The installation files that are there
 will build a version 8 client on all of the 9.x and 10.x builds.


 If you get root access, you can pull out the files for the agent,
 supervisor and desktop admin (grab the full CAD directory structure), and
 put them on your desktop machine.


 Then get the configuremsi.exe, and the other two files in the directory
 with it and drop them into the CAD directory.  Run the configuremsi.exe
 file and it will create the proper MSI files you will need.


 Not sure why this is so hard for Cisco.  I have gotten the complete run
 around on this issue.


 *Cliff McGlamry, CCIE Collaboration #24757*
 *Master Consultant*
 (678) 934-0348   direct
 (404) 969-9806   mobile
 (678) 934-0448   fax
 cmcgla...@forsythe.com



 *Forsythe Technology, Inc.*
 400 Interstate North Parkway

 Suite 860
 Atlanta, GA 30339
 http://www.forsythe.com









 image001.gif




 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Chris Clouse
 *Sent:* Friday, September 19, 2014 8:52 AM
 *To:* Matthew Loraditch; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Prep CAD Client Update Ahead of time?



 You can utilize the CAD Preconfiguration tool.

 Running the Preconfiguration Tool

 The Preconfiguration tool is located on the CAD 9.0 installation DVD. Run
 this tool if client MSIs are needed before the Unified CCX server is
 installed.

 To run the Preconfiguration tool:

 1.On the CAD 9.0 installation DVD, navigate to the Installer folder.

 2.Copy the Installer folder and all its contents to any location on a
 Windows PC.

 3.In the Installer folder, double-click ConfigureMsi.exe. The
 Preconfiguration tool starts.

 4.When prompted, provide the following information:

 Language of the contact center■

 IP address of the primary Unified CCX server (server 1)■

 IP address of the backup Unified CCX server (server 2) (or 'none' if the
 system is not high availability)■

 ■License type (Standard, 

Re: [cisco-voip] UCCX 8.5.1sr3 to sr4 to Refresh Upgrade 10.5.1

[Justin Steinberg]

Sounds like Jason is doing HA.  Anthony I didn't see any sub listed in your
tasks, so I think that accounts for some portion of the difference.
On Nov 5, 2014 11:12 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com
wrote:

  The funny part is that this is the replicated offline environment.  Once
 we test scripts, databases, etc then production gets migrated J



 *From:* Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com]
 *Sent:* Wednesday, November 5, 2014 10:47 PM
 *To:* Heim, Dennis; Jason Aarons (AM); cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] UCCX 8.5.1sr3 to sr4 to Refresh Upgrade 10.5.1





 Dennis,



 You're on to something here, however, resources such as time and money are
 not always in abundance.  Sometimes, you're asked to do things as quickly
 and as cheaply as possible.  And by sometimes, I mean most of the time.

 On Wed Nov 05 2014 at 6:55:04 AM Heim, Dennis dennis.h...@wwt.com wrote:

  If you have the resources available (time and/or money), my preference
 is to migrate the environment to an offline environment.  If you have the
 compute resources do it there, or if not some type of Lab a Service (LaaS).
 Maybe use Veeam to migrate virtual machines, if you are already virtual
 (make sure to sure mac addresses to manual prior to copying).



 Then you need to determine what you time frame is.. can you lose a few
 days of: cdr’s, historical reports, etc. Migrating VM’s between environment
 would need to be worked out. However, that allows you to identify any
 issues, develop a realistic timeframe for each aspect of the upgrade
 process. Most importantly, it allows a lot of the work to be done not in
 the middle of the night.



 *Dennis Heim | Collaboration Solutions Architect*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Anthony Holloway
 *Sent:* Tuesday, November 04, 2014 11:37 PM
 *To:* Jason Aarons (AM); cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] UCCX 8.5.1sr3 to sr4 to Refresh Upgrade 10.5.1



 Wow.  Are you going MCS to UCS too?  If not, I'm doing that exact upgrade
 very soon and no documentation I have reviewed indicated how long upgrades
 would take, let alone 5 hours.  Right now my plan has a time budget of 3
 hours for the same, which does not include moving ISOs to the datastore (I
 like doing local upgrades) nor upgrading clients.



 Which portion is taking 5 hours?  Could you break it down by task for us?
 E.g., L2 to SU4, Switch Version to SU4, RU COP, RU to 10.5, Switch Version
 to 10.5.



 Did you disable IO Throttling first?



 Based on my research here are the summary steps to this exact upgrade
 (sans client upgrades):



 1. Upload ISOs to DataStore

 2. Disable IO Throttle on Publisher

 3. Mount 8.5(1)SU4 ISO

 4. Upgrade Publisher to 8.5(1)SU4

 5. Switch Version to 8.5(1)SU4

 6. Patch Publisher with RU COP

 7. Shutdown Publisher

 8. Modify VM Settings: RAM from 4GB to 8GB

 9. Power On Publisher

 10. Mount 10.5(1)SU1 ISO

 11. Upgrade Publisher to 10.5(1)SU1

 12. Switch Version to 10.5(1)SU1



 On Tue Nov 04 2014 at 11:04:24 PM Jason Aarons (AM) 
 jason.aar...@dimensiondata.com wrote:

  One UCCX node in ha pair 5 hours doing upgrades...it would be more
 enjoyable watching paint dry...there has got to be a better method in the
 future for L2 upgrades than this..





 Sent from my Verizon Wireless 4G LTE Smartphone

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Re: [cisco-voip] Broadcast Hunt Group query

[Justin Steinberg]

you will probably crash the system if you put 4,000 phones in a broadcast
hunt group :)

You would need some third party application like Singlewire Informacast,
SA-Announce Synapps, etc.  to get this type of functionality

On Thu, Nov 6, 2014 at 7:59 AM, Andy andy.ca...@gmail.com wrote:

 I know that this is going to bring a smile to some people, but I’ll ask
 the question.
 I have a request come in
 “Can we use the phones to mark the start and finish time for marking the
 silence on 11 Nov”

 I thought about using a broadcast hunt group, but there are 4,000 handsets.

 I have read somewhere that you can only use a handfull of devices in a
 broadcast hunt group, so I’m looking for some divine inspiration should
 anyone have any to spare.
 obviously if it can’t be easily achieved then I do have the option to say
 no.

 Andy
 andy.ca...@gmail.com




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Re: [cisco-voip] UCCX 8.5.1sr3 to sr4 to Refresh Upgrade 10.5.1

[Justin Steinberg]

i'll take a stab at time estimates.  that being said, i have heard of
switch-versions taking much longer or sitting at SELINUX loading for longer
than the typical 15-20 minutes.

checking for existing snapshots before doing anything is really important.
  if there are snapshots that someone left on their from the past, it will
most likely completely hose the upgrade times and make the system extremely
slow.

1. Upload ISOs to DataStore
2. Disable IO Throttle on Publisher
3. Mount 8.5(1)SU4 ISO
4. Upgrade Publisher to 8.5(1)SU445 - 60 minutes
5. Switch Version to 8.5(1)SU4 60 minutes (includes switch version process,
reboot, SELINUX loading, services startup)
6. Patch Publisher with RU COP 5 minutes
7. Shutdown Publisher 10 minutes
8. Modify VM Settings: RAM from 4GB to 8GB
9. Power On Publisher 15 minutes
10. Mount 10.5(1)SU1 ISO
11. Upgrade Publisher to 10.5(1)SU1  45-60 minutes
12. Switch Version to 10.5(1)SU1 60 minutes (includes switch version
process, reboot, SELINUX loading, services startup)
13. Rerun the CAD Client Configuration Tool (10 minutes)

On Wed, Nov 5, 2014 at 12:36 AM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Wow.  Are you going MCS to UCS too?  If not, I'm doing that exact upgrade
 very soon and no documentation I have reviewed indicated how long upgrades
 would take, let alone 5 hours.  Right now my plan has a time budget of 3
 hours for the same, which does not include moving ISOs to the datastore (I
 like doing local upgrades) nor upgrading clients.

 Which portion is taking 5 hours?  Could you break it down by task for us?
 E.g., L2 to SU4, Switch Version to SU4, RU COP, RU to 10.5, Switch Version
 to 10.5.

 Did you disable IO Throttling first?

 Based on my research here are the summary steps to this exact upgrade
 (sans client upgrades):

 1. Upload ISOs to DataStore
 2. Disable IO Throttle on Publisher
 3. Mount 8.5(1)SU4 ISO
 4. Upgrade Publisher to 8.5(1)SU4
 5. Switch Version to 8.5(1)SU4
 6. Patch Publisher with RU COP
 7. Shutdown Publisher
 8. Modify VM Settings: RAM from 4GB to 8GB
 9. Power On Publisher
 10. Mount 10.5(1)SU1 ISO
 11. Upgrade Publisher to 10.5(1)SU1
 12. Switch Version to 10.5(1)SU1

 On Tue Nov 04 2014 at 11:04:24 PM Jason Aarons (AM) 
 jason.aar...@dimensiondata.com wrote:

  One UCCX node in ha pair 5 hours doing upgrades...it would be more
 enjoyable watching paint dry...there has got to be a better method in the
 future for L2 upgrades than this..


  Sent from my Verizon Wireless 4G LTE Smartphone
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[cisco-voip] PCD CUCM Migration Upgrades with UCCX

[Justin Steinberg]

Anyone have any thoughts on using PCD to upgrade CUCM in environments with
UCCX ?

Assuming customer is running v8 CUCM  UCCX on MCS hardware and wants to
upgrade to 10.5.

PCD can take CUCM from v8 MCS direct to ESXi and 10.5.  But, UCCX isn't
supported for PCD migrations.

CUCM 10.5 isn't supported with UCCX v8 and likewise, UCCX 10.5 isn't
supported with CUCM v8.

So I was thinking of using the process below:

Maintenance Window 1
1) Virtualize UCCX v8 on ESXi

Maintenance Window 2
1) Install RU upgrade on UCCX to 10.5 on inactive partitions
2) PCD Upgrade/Migration CUCM to 10.5 (same IP/hostnames)
3) Once CUCM is complete, initiate switch-version on UCCX to 10.5.

Does anyone see any issues with temporarily having CUCM at 10.5 before the
switch-version begins on UCCX?Does UCCX check the switch-version to see
if the CUCM is compatible?

I haven't found this documented anywhere yet, so I've stayed away from PCD
for the time being but curious if anyone has found a good way to handle PCD
in environments with UCCX.

Justin
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Re: [cisco-voip] CUCM 10.5 Tomcat Subject Alternate Name (SAN) (Alternatehostname) via CLI

[Justin Steinberg]

I thought you could only add a single SAN via command line.

On Wed, Oct 8, 2014 at 11:20 AM, Jason Aarons (AM) 
jason.aar...@dimensiondata.com wrote:

Trying to add an alternatehostname in CUCM 10.5  I got the below
 error.  Anyone spot my problem?



 set web-security CH Hawaii Department of Boating Honolulu Hawaii US
 ton.state.hi.us HI-IT-UC-CM-P.ton.state.hi.us myphone.ton.state.hi.us

 Expected 4 mandatory and up to 2 non-mandatory parameter(s)
 but 8 parameter(s) were found

 Executed command unsuccessfully
 Error executing command
 admin:



 Names changed to protect the innocent :)





 admin:set web-security ?
 Syntax:
 set web-security orgunit orgname locality state [country]
 [alternatehostname]
 orgunit  mandatory   organizational unit
 orgname  mandatory   organizational name
 locality mandatory   location of organization
 statemandatory   state of organization
 country  optional   country code can not be changed
 alternatehostname  optional   alternate host name

 admin:set web-security

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Re: [cisco-voip] EWS Limits Throttling Policy

[Justin Steinberg]

I've successfully used the 'paged view functionality' on the later versions
of Connections that works around this issue by staying under the EWS
default limits.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/unified_messaging/guide/10xcucumgx/10xcucumg020.html#83993


On Thu, Sep 11, 2014 at 10:52 AM, Daniel Pagan dpa...@fidelus.com wrote:

  Folks:



 I’m hoping someone can share their experience with the Cisco recommended
 method for removing EWS limits on Exchange 2010 SP2 RU4 and higher. In
 earlier releases of Ex2010 the process of setting a throttling policy
 applied only to the UM service account, and any throttling performed would
 be applied to that service account and not to the target mailbox. Please
 correct me if my understanding is incorrect, but with E2010 SP2 RU4 and
 higher, the policy is to be applied to every target mailbox, which seems
 like it would impact all other EWS applications impersonating these target
 mailboxes.

 *Removing EWS Limits from Exchange 2010 SP2 RU4 and Later*

 Microsoft has enabled the client throttling policy feature by default. If
 there is no throttling policy already configured, Microsoft Exchange
 applies a default policy to all users. The default throttling policy is
 tailored for end user's load and not for an enterprise application like,
 Cisco Unity Connection using impersonation. If any Cisco Unity Connection
 users who are configured for unified messaging have mailboxes in Exchange
 2010, configure the Exchange 2010 EWS limits for the unified messaging
 users mailbox by creating and applying a new mailbox policy to the unified
 messaging user mailbox account. If you do not configure EWS limits,
 messages may not be synchronized, and status changes (for example, from
 unread to read), changes to the subject line, and changes to the priority
 may not be replicated. In addition, attempts to access Exchange calendars
 and contacts may fail.



 *The MS KB referring to the throttling policy change*:
 http://support.microsoft.com/kb/2713371



 Perhaps my understanding is wrong, but it seems like a backwards move. Has
 anyone seen any adverse effects of applying the Cisco recommended
 throttling values as the system default? Perhaps any problems where
 applying the throttling policy to the target mailbox impacts other EWS apps
 like BlackBerry Enterprise? Are you applying the throttling policy for
 every single UM enabled mailbox, individually, via management shell?



 Thanks!



 - Dan

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[cisco-voip] Interested UCCX Script Issue

[Justin Steinberg]

I am working on a script where I call the CCX application via HTTP trigger.

The CCX script will use the  'place call' step to call a CUCM hunt group.
 Then I need to determine the line group member (extension would be fine)
of the person who answered the call.  I then need to pass the extension of
the phone that answered back from the script via HTTP to the requesting
application.

Unfortunately, I have not been able to find a CCX editor step that can tell
me the extension of the line group member who answered the call.   All
'get' steps just show the extension of the hunt pilot used to route the
call to the group of phones, I can't actually see the individual person who
answered.

The phones in the group are not agents, so there is no CSQ or 'select
resource' in play here.   I'm just using CCX to leverage the scripting
capabilities to interface with the other application.

Perhaps Java would be of use here, though I am not sure what code would be
used.

Any thoughts or ideas ?

Justin
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Re: [cisco-voip] Possible to force SIP refer transfer from IP phone

[Justin Steinberg]

Couldn't find a way to add the delay on the dialer side, i'm just not
familiar with that system.

I ended up moving the extensions that people dial from Route Patterns
pointing at the dialer to Unity Connection, where I play a short call
handler prompt and then have Unity Connection transfer the call to a new
hidden route pattern.   As long as the IP phone user completes the
supervised transfer during the call handler, the dialer/acd will see the
original PSTN caller.The call volume of manual transfers from IP phones
to the external dialer/acd is low, so this should work just fine.


On Tue, Aug 12, 2014 at 10:39 PM, Brian Meade bmead...@vt.edu wrote:

 Justin,

 Since these phones do not support blind transfer, it won't be possible to
 do something like that.

 Can you add a delay before checking the number so that it checks the
 connected number after the transfer is completed?

 Brian


 On Tue, Aug 12, 2014 at 7:12 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 I have a unique requirement between a new 10.5 CUCM and a third party
 dialer/ACD.

 CUCM is integrated to PSTN via SIP trunk to 2911 PRI gateway.   CUCM is
 also integrated via SIP trunk to the dialer/ACD via SIP trunk directly
 between CUCM and the dialer/ACD.

 CUCM takes inbound call to IP phone. User needs to be able to transfer
 the external caller to the dialer/ACD.  However we need the initial invite
 sent to the dialer to have the Calling Party Number set to the external
 PSTN user, so the dialer can properly do a database lookup of the external
 party and match the caller ID.

 When we do this transfer via IP phone, it always starts off as a warm
 transfer so the dialer/ACD gets the internal IP phone number for the DB
 lookup.

 Is there anyway for a IP phone (or jabber) to utilize SIP refer to
 complete the transfer so the dialer/ACD sees the external party number as
 caller ID on the transferred leg?

 As a secondary option, I have considered possibly delaying the actual
 transfer to the dialer/ACD to give the IP phone user enough time to
 complete the supervised transfer. But not sure if unity connection call
 handler is the best way or if there is another way, perhaps in CUCM
 dialplan.

 Thanks

 Justin

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[cisco-voip] Possible to force SIP refer transfer from IP phone

[Justin Steinberg]

I have a unique requirement between a new 10.5 CUCM and a third party
dialer/ACD.

CUCM is integrated to PSTN via SIP trunk to 2911 PRI gateway.   CUCM is
also integrated via SIP trunk to the dialer/ACD via SIP trunk directly
between CUCM and the dialer/ACD.

CUCM takes inbound call to IP phone. User needs to be able to transfer the
external caller to the dialer/ACD.  However we need the initial invite sent
to the dialer to have the Calling Party Number set to the external PSTN
user, so the dialer can properly do a database lookup of the external party
and match the caller ID.

When we do this transfer via IP phone, it always starts off as a warm
transfer so the dialer/ACD gets the internal IP phone number for the DB
lookup.

Is there anyway for a IP phone (or jabber) to utilize SIP refer to complete
the transfer so the dialer/ACD sees the external party number as caller ID
on the transferred leg?

As a secondary option, I have considered possibly delaying the actual
transfer to the dialer/ACD to give the IP phone user enough time to
complete the supervised transfer. But not sure if unity connection call
handler is the best way or if there is another way, perhaps in CUCM
dialplan.

Thanks

Justin
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Re: [cisco-voip] MRA

[Justin Steinberg]

you manually copy the SANs from Expressway C GUI and just paste them into
the SAN section of Expressway E GUI.  Then generate CSR for E, then setup
TLS link between CE


On Tue, Jul 15, 2014 at 11:49 AM, Heim, Dennis dennis.h...@wwt.com wrote:

 Working through MRA. TLS is required which require a SAN certificate, I
 get that. However, the Expressway-E does not populate the chat nodes
 subject alternative names without the TLS link between CE coming up. What
 am I missing?



 *Dennis Heim | Collaboration Solutions Architect*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]





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Re: [cisco-voip] Disappearing Agents in CAD

[Justin Steinberg]

See if CAD is running in compatibility mode on those Agent's PCs.I had
an issue with an upgrade to CCX 10.0su1 and I believe it was related since
the previous version of CAD was running in compatiblity mode.  Upgrading
CAD kept that setting and the CCX 10.0su1 didn't seem to like it.


On Mon, Jul 14, 2014 at 3:43 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  Has anyone seen something like this, Agents stay online and take calls,
 but appear and disappear in Supervisor?

 CCX 10.0SU1





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





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Re: [cisco-voip] MRA/Expressway

[Justin Steinberg]

It should match the XMPP domains configured on your CUPS / IMP server.

V10 configures these by default based on the domains you are using.

In v9 the XMPP is set by default to the domain name of the server, but can
be changed to another domain.

For .local environments, I have changed the XMPP domain to be the actual
public FQDN and put that on the internal CUPS/ IMP server, and then the
expressway.
On Jul 9, 2014 12:18 AM, Heim, Dennis dennis.h...@wwt.com wrote:

 I am trying to configure MRA for the first time. We have split dns zones,
 so acme.loc (internal DNS) and acme.com (external DNS). We are struggling
 with determining what should be in the subject alternative names (SAN) for
 the Expressway-E certificate?



 Thanks,



 *Dennis Heim | Collaboration Solutions Architect*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]





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Re: [cisco-voip] MRA Expressway Edge 1 or 2 Ports?

[Justin Steinberg]

Single port solution without advanced Networking license will only work if
you assign the expressway E single port a valid public address directly to
expressway E.  If your DMZ uses private addresses you need the advanced
networking license.  You can still use a single nic with advanced
networking.

Expressway MRA has advanced networking as a $0 part for CUWL or enhanced
licensing.  You just need to order that license and install it.
On Jun 26, 2014 2:04 AM, Boon ciscovoipu...@gmail.com wrote:

 We've rolled out MRA on Expressway 8.1.1 but we can only get it to work
 properly with a 2-port Expressway-E setup which requires the Advanced
 networking license.

 Can anyone confirm if they've got it to work properly using the 1-port
 method without the Adv Net license?

 We had one-way audio issues using the single port solution. Also, the
 documentation is not clear on this point.

 TIA



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Re: [cisco-voip] IMP 10.x Source the SIP traffic from a particular node to OCS2007R2

[Justin Steinberg]

The SIP federation service does this, so you could disable that service on
certain nodes.
On Jun 26, 2014 11:32 AM, Jason Aarons (AM) 
jason.aar...@dimensiondata.com wrote:

  I’ve setup a Partitioned Intradomain Federation between IMP 10.5 and OCS
 2007R2.



 We have 3 nodes running IMP.  Can I set a service parameter to source the
 IMP traffic from a particular node?



 Jason Aarons

 Consultant

 Dimension Data

 904-338-3245

 Planned PTO: July 4 – Aug 12th



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Re: [cisco-voip] SCCP Session Border Controllers

[Justin Steinberg]

is anyone actually using CUBE Line Side ?


On Wed, Jun 11, 2014 at 3:30 PM, Brian Meade bmead...@vt.edu wrote:

 I've done a lot of phone proxy work.  It doesn't really work for new
 phones that used signed config files.  It was pretty much abandoned
 completely once Phone VPN was introduced.  If you're using a CUCM 7.x or
 earlier environment, phone proxy is still a good choice. It also works
 pretty well in newer CUCM versions if you're running 7940s/7960s.

 I would suggest just doing Phone VPN if possible.  It's much easier and
 works on pretty much all of the newer phones.




 On Wed, Jun 11, 2014 at 2:54 PM, Euan McGregor e...@mcgregor.co wrote:

 Thanks, I was looking at the ASA someone told me that the phone proxy on
 the ASA just did not work, have you had an experience in the ASA Phone
 Proxy?

 Does the line side support on the CUBE support SCCP I thought it was only
 SIP and the newer 9900 phones.

 Thanks


 On 11 June 2014 18:40, Heim, Dennis dennis.h...@wwt.com wrote:

 CUBE supports line side registrations



 *Dennis Heim | Collaboration Solutions Architect*

 World Wide Technology, Inc. | +1 314-212-1814

 [image: twitter] https://twitter.com/CollabSensei

 [image: chat][image: Phone] +13142121814[image: video]





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Brian Meade
 *Sent:* Wednesday, June 11, 2014 12:19 PM
 *To:* Euan McGregor
 *Cc:* cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] SCCP Session Border Controllers



 ASA can do TLS proxy.



 Depending on phone models/CUCM version, you could use the ASA phone
 proxy feature as well if you're not mixed mode on CUCM.



 On Wed, Jun 11, 2014 at 11:23 AM, Euan McGregor e...@mcgregor.co
 wrote:

 Hello all,



 We are looking to replace our, now Avaya, Sipera Session Border
 Controller box. We are looking for something that will do SCCP and
 essentially proxy the session back to call manager.



 Does anybody know if there any products or solution either paid or open
 source that can do this for us?



 Any help appreciated.



 Thanks,



 Euan


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Re: [cisco-voip] SIP phones CMR

[Justin Steinberg]

The only data I see in the CMR are the IP addresses of the endpoints and
the payloadcapability.All the packets sent/received, jitter, latency,
etc values are always just 'null'.

Sample below.

[image: Inline image 1]


On Mon, Apr 28, 2014 at 2:10 PM, Ryan Ratliff (rratliff) rratl...@cisco.com
 wrote:

 What specific data are you looking for?  The newer phones have dropped MOS
 and Q-Factor subjective ratings in lieu of concealed seconds and severely
 concealed seconds.

 You won't see any raw MOS values reported by newer phones.

 -Ryan

 On Apr 25, 2014, at 1:16 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

 I've noticed on the last few systems I have worked on that the SIP based
 phones are not showing any CMR data in the CUCM CAR system.

 This is happening on atleast 8945s  the new 7841s.   Also I have noticed
 that the Jabber for Windows phones aren't reporting it either.

 This is happening for both on-net IP phone to IP phone calls and also
 off-net SIP calls to CUBEs (then to SIP Provider) and also off-net calls
 via CUCM SIP trunk to Cisco 29xx PRI gateway.

 Is there something I need to turn on for the CMR data to be generated ?
  Or is this just not supported in the SIP phones / CUCM ?

 Justin
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[cisco-voip] SIP phones CMR

[Justin Steinberg]

I've noticed on the last few systems I have worked on that the SIP based
phones are not showing any CMR data in the CUCM CAR system.

This is happening on atleast 8945s  the new 7841s.   Also I have noticed
that the Jabber for Windows phones aren't reporting it either.

This is happening for both on-net IP phone to IP phone calls and also
off-net SIP calls to CUBEs (then to SIP Provider) and also off-net calls
via CUCM SIP trunk to Cisco 29xx PRI gateway.

Is there something I need to turn on for the CMR data to be generated ?  Or
is this just not supported in the SIP phones / CUCM ?

Justin
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Re: [cisco-voip] MTP allocation failure - CDDR cause code 47

[Justin Steinberg]

Any change device mobility could be in play for this user changing the
device pool/region ?


On Fri, Apr 18, 2014 at 11:12 AM, Brian Meade bmead...@vt.edu wrote:

 There's some scenarios where you do not get a media resource list
 exhausted alert.  One scenario is where the CCM service has an incorrect
 count of available resources and you just see the MTP fail to send an
 OpenReceiveChannelAck or send one saying it failed to open a port.  In that
 case, CUCM didn't think the media resources were actually exhausted so no
 alert.

 We'll need to see CCM traces for one of these calls to see what is
 actually happening.  Have you tried resetting the MTP?

 Brian


 On Fri, Apr 18, 2014 at 10:55 AM, Erick Wellnitz 
 ewellnitzv...@gmail.comwrote:

 Ryan,

 When you mention stuck sessions do you mean x number of stuck sessions
 preventing further allocation of resources or this particular device has a
 stuck session?

 If it were the first scenario, wouldn't I get a media list exhausted
 alert from RTMT?




 On Fri, Apr 18, 2014 at 7:36 AM, Ryan Ratliff (rratliff) 
 rratl...@cisco.com wrote:

  Stuck sessions on the MTP perhaps or leaked calls in UCM?

  You're going to have to dig into traces to confirm.

 -Ryan

  On Apr 17, 2014, at 10:52 PM, Erick Wellnitz ewellnitzv...@gmail.com
 wrote:

  Well, identical except for MAC and description.

  It worked for three days now it fails.  The last successful call
 allocated the MTP without issue.

  I personally tested it a dozen or so times before giving it back to
 the user.


 On Thu, Apr 17, 2014 at 9:45 PM, Erick Wellnitz ewellnitzv...@gmail.com
  wrote:

 Super copy identical.


 On Thu, Apr 17, 2014 at 4:33 PM, Ryan Ratliff (rratliff) 
 rratl...@cisco.com wrote:

 47 = codec mismatch

 Is it _really_ identical?

 -Ryan

 On Apr 17, 2014, at 4:02 PM, Erick Wellnitz ewellnitzv...@gmail.com
 wrote:

 I have a VPN phones configured to use a trused relay point in order to
 provide connectivity to other VPN phones.

 One of the phones gets fast busy and traces show MTP allocation
 failure.  This was the only phone trying ot utilize this MTP and is
 identical to other VPN phones that work as expected.

 Any ideas?


 Thanks!


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Re: [cisco-voip] Collaboration Edge for Jabber

[Justin Steinberg]

You need the advanced networking key so you can specify the public IP.

There is no charge for that license when used for collaboration edge (i.e.
Cisco expressway)
On Mar 20, 2014 5:12 AM, Bruno Nonogaki (AM) 
bruno.nonog...@dimensiondata.com wrote:

  If you don't have this key, you have to configure the public address
 directly in the vcse interface. NAT will not work.

 Regards,

 Bruno
 Em 20/03/2014 03:11, Dana Tong dana_t...@bridgepoint.com.au escreveu:

 The SDP in the logs appears to show :



 C = IN IP4 192.168.x.x





 I've emailed my Cisco rep to see if he can get me a demo option key for
 the Dual Network Interface option so I can try the Static NAT config on the
 VCSE.



 *From:* Dana Tong
 *Sent:* Thursday, 20 March 2014 3:21 PM
 *To:* Dana Tong; Jason Aarons (AM); Ki Wi; Cisco VoIP List
 *Subject:* RE: [cisco-voip] Collaboration Edge for Jabber



 Just to clarify, normal inbound/ outbound video calls work.



 The Expressway does not have the Dual Network Interface option key.



 Cheers





 *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *Dana Tong
 *Sent:* Thursday, 20 March 2014 2:14 PM
 *To:* Jason Aarons (AM); Ki Wi; Cisco VoIP List
 *Subject:* Re: [cisco-voip] Collaboration Edge for Jabber



 Hey all,



 Back on this testing for MRA on Collab Edge... I've run up Wireshark on
 the computer that I signed in on.



 When I establish a call to mobile, or the MCU from the Internet I can see
 that my destination address is the PRIVATE IP of the Expressway Edge.

 The Expressway E is configured with a 192.168.x.x IP.



 I thought there was a way you could configure the VCS E with what it's
 Public NAT'd address somewhere? Can anyone recall where I might be able to
 fix this?



 Thanks

 Dana





 *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *Jason Aarons (AM)
 *Sent:* Thursday, 6 February 2014 5:50 AM
 *To:* Ki Wi; Cisco VoIP List
 *Subject:* Re: [cisco-voip] Collaboration Edge for Jabber



 Until VCS 8.1.1 comes out it's a feature preview (Mobile and Remote
 Access) that doesn't have TAC support.  I have several interested customers
 waiting on VCS 8.1.1 and a formal Jabber for Windows that maps to it (yeah
 it's hidden feature in 9.6.0) and for Jabber for Mac 10.0 and Jabber for
 blah to all support TLS based Jabber solution.



 I wouldn't call it Collaboration Edge for Jabber as 'Collaboration Edge'
 is a umbrella architecture meaning different things to different people.



 Let us know how your testing goes.  Without formal documentation/release
 the 'caveats' aren't fully known yet to everyone.  I expect by March/April
 everything we know will have changed.







 *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *Ki Wi
 *Sent:* Wednesday, February 05, 2014 11:27 AM
 *To:* Cisco VoIP List
 *Subject:* [cisco-voip] Collaboration Edge for Jabber





 Anyone tested it with VCS 8.1 Release?

 Does Jabber functions properly when connected to public internet without
 VPN? Any function which is not working at this preview release ?



 itevomcid

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Re: [cisco-voip] Changing CUCM SystemServer from IP to FQDN

[Justin Steinberg]

here is how I understand it.

If DNS has the _cisco-uds._tcp.customer.com SRV record, Jabber starts the
service discovery process using that SRV.

Jabber takes the results from the SRV, then performs a query against the
CUCM defined in the SRV using:

https://homeCluster:8443/cucm-uds/servers

Unfortunately, the results to the above query are based on the CUCM System
 Server.   If that is defined as IP, then Jabber gets IPs.

Then Jabber takes those IPs, and performs the check below.

https://IP:8443/cucm-uds/clusterUser?username=userid

Since Jabber uses IP, it throws a cert validation error because the IP
doesn't match the FQDN (or SAN) of the certs.

There is a defect on CUCM CSCul15900 and a COP to correct that, but it only
applies to the second URL above.   The first URL continues to return IPs
and this causes the issue.

If I remove the _cisco-uds SRV and go without using service discovery, I
don't encounter the issue.  Because Jabber doesn't perform the associated
queries listed above.

I've opened TAC case for this 629194597, and the Presence engineer
indicated changing CUCM SystemServer to FQDN was an option.

There is also a defect on the Jabber client CSCul39696, which seems to
indicate that jabber will allow CUCM SystemServer to just be a hostname
and the tomcat to only have the FQDN.   So it seems like eventually, we
could get by with CUCM SystemServer just set to hostname, but I don't yet
believe this is fixed in an available verson of Jabber.

It seems to be, that if CUCM team opened CSCul15900  to address:

https://IP:8443/cucm-uds/clusterUser?username=userid

They could also fix:

https://homeCluster:8443/cucm-uds/servers

and that would fix the problem.







On Tue, Feb 18, 2014 at 10:27 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.com wrote:

  The doc I have is a draft version so may be subject to change.

  What is directing you to use the FQDN for helping jabber validate certs?
 I don't see anything perusing the server setup guide or install guide for
 9.6.

 Sent from my iPhone

 On Feb 18, 2014, at 8:35 PM, Justin Steinberg jsteinb...@gmail.com
 wrote:

   What document is that?

 The reason for changing the systemserver to fqdn is more because of the
 Jabber for Windows certificate validation process than for CE.

 That being said, if Jabber wants FQDN and Collab Edge doesn't, we will
 have a problem.
 On Feb 18, 2014 2:45 PM, Ryan Ratliff (rratliff) rratl...@cisco.com
 wrote:

 Using FQDN in System-Server on UCM is specifically not supported in the
 Collab Edge deployment guide I'm looking at.

  It's ok to use a hostname there but it must be resolvable by the
 Expressway-C (and the phones of course).

 -Ryan

  On Feb 18, 2014, at 11:36 AM, Erick Wellnitz ewellnitzv...@gmail.com
 wrote:

  We were but it interferes with EMCC.  The phone only does DNS lookups
 for the domain it is assigned so we switched back to IP addreses.


 On Tue, Feb 18, 2014 at 8:39 AM, Justin Steinberg 
 jsteinb...@gmail.comwrote:

  Is anyone using CUCM deployments where they set the SystemServer
 value to the FQDN of the CUCM nodes ?

 I'm in the process of deploying Jabber for Windows 9.6+ with
 Collaboration Edge, and there are requirements around certificate
 validation that seem to require the CUCM SystemServer value set to a FQDN.

 I'm concerned around making this change and curious if others are using
 FQDN.

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[cisco-voip] Changing CUCM SystemServer from IP to FQDN

[Justin Steinberg]

Is anyone using CUCM deployments where they set the SystemServer value to
the FQDN of the CUCM nodes ?

I'm in the process of deploying Jabber for Windows 9.6+ with Collaboration
Edge, and there are requirements around certificate validation that seem to
require the CUCM SystemServer value set to a FQDN.

I'm concerned around making this change and curious if others are using
FQDN.
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Re: [cisco-voip] Changing CUCM SystemServer from IP to FQDN

[Justin Steinberg]

What document is that?

The reason for changing the systemserver to fqdn is more because of the
Jabber for Windows certificate validation process than for CE.

That being said, if Jabber wants FQDN and Collab Edge doesn't, we will have
a problem.
On Feb 18, 2014 2:45 PM, Ryan Ratliff (rratliff) rratl...@cisco.com
wrote:

  Using FQDN in System-Server on UCM is specifically not supported in the
 Collab Edge deployment guide I'm looking at.

  It's ok to use a hostname there but it must be resolvable by the
 Expressway-C (and the phones of course).

 -Ryan

  On Feb 18, 2014, at 11:36 AM, Erick Wellnitz ewellnitzv...@gmail.com
 wrote:

  We were but it interferes with EMCC.  The phone only does DNS lookups
 for the domain it is assigned so we switched back to IP addreses.


 On Tue, Feb 18, 2014 at 8:39 AM, Justin Steinberg jsteinb...@gmail.comwrote:

  Is anyone using CUCM deployments where they set the SystemServer value
 to the FQDN of the CUCM nodes ?

 I'm in the process of deploying Jabber for Windows 9.6+ with
 Collaboration Edge, and there are requirements around certificate
 validation that seem to require the CUCM SystemServer value set to a FQDN.

 I'm concerned around making this change and curious if others are using
 FQDN.

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Re: [cisco-voip] 7945 VPN option

[Justin Steinberg]

Are these new phones on the internal LAN?  They will need to be in order to
download the initial config files before you take them remote.


On Mon, Jan 27, 2014 at 10:27 AM, Brian Meade (brmeade)
brme...@cisco.comwrote:

  James,



 A restart of TFTP service may do the trick since the web page  is showing
 the correct common phone profile assuming you tried to pull the file from
 the publisher.  IF you tried another node, it may be a replication issue.
 Check “utils dbreplication runtimestate” to check overall replication state
 and “utils dbreplication status” to check for any mismatched tables.



 Brian



 *From:* James Dust [mailto:james.d...@charles-stanley.co.uk]
 *Sent:* Monday, January 27, 2014 10:23 AM

 *To:* Brian Meade (brmeade); 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 Thanks Brian,



 It looks as though (from the config page) the phone is not getting the
 Common Phone Profile, although it is definitely configured on the device in
 CUCM.



 Would a reset of the Common Phone Profile be a good place to start? Or is
 there something else I should be looking at?



 Kind regards



 James







 *From:* Brian Meade (brmeade) [mailto:brme...@cisco.combrme...@cisco.com]

 *Sent:* 27 January 2014 15:08
 *To:* James Dust; 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 http://x.x.x.x:6970/SEPMAC.cnf.xml



 Replace x.x.x.x with a node on your cluster running TFTP service.



 *From:* James Dust 
 [mailto:james.d...@charles-stanley.co.ukjames.d...@charles-stanley.co.uk]

 *Sent:* Monday, January 27, 2014 10:01 AM
 *To:* Brian Meade (brmeade); 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 Hi Brian,



 Yes the Common Phone Profile is definitely applied, can I get the config
 file from the web page of the phone and if so where?



 Kind regards



 James Dust
 Technical Infrastructure Engineer
 Charles Stanley  Co Ltd
 Tel: 020 7149 6314
 Mob: 07989 491136
 mailto: james.d...@charles-stanley.co.uk



 *From:* Brian Meade (brmeade) [mailto:brme...@cisco.combrme...@cisco.com]

 *Sent:* 27 January 2014 14:43
 *To:* James Dust; 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 James,



 Do you have the correct Common Phone Profile applied to the phone in CUCM
 with the VPN settings?  Does the phone’s config file have the VPN settings?



 Thanks,

 Brian



 *From:* James Dust 
 [mailto:james.d...@charles-stanley.co.ukjames.d...@charles-stanley.co.uk]

 *Sent:* Saturday, January 25, 2014 8:54 AM
 *To:* Brian Meade (brmeade); 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 Afternoon Brian,



 I have checked this and Auto Network Detect is off.



 Do you have any other suggestions?



 The firmware level is: SCCP45.9-3-1SR1-1S





 *From:* Brian Meade (brmeade) [mailto:brme...@cisco.combrme...@cisco.com]

 *Sent:* 24 January 2014 20:28
 *To:* James Dust; 'cisco-voip@puck.nether.net'
 *Subject:* RE: 7945 VPN option



 If you’re using Auto Network Detect, that’s probably the issue since you
 can’t manually connect with Auto Network Detect turned on the VPN profile.



 Brian



 *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *James Dust
 *Sent:* Friday, January 24, 2014 3:02 PM
 *To:* 'cisco-voip@puck.nether.net'
 *Subject:* [cisco-voip] 7945 VPN option



 Evening everyone,



 We have been deploying Cisco 7945 IP phones to our home users for use with
 our ASA VPN solution. The user plugs their phone into the the home DSL and
 connects using the by enabling the VPN option on the phone.



 I have reviewed a batch of 7945's and configured then in the normal manor,
 however when trying to enable the VPN I get the message 'key not enabled
 here'



 I have looked to the options on the phone on cucm but can't see anything
 onviouslt wrong.



 Has anyone seen this?


 Regards

 James

 Sent from my iPhone


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