Re: [cisco-voip] Jabber/Webex E911Notification

[Florian Kroessbacher]

Hy

we put this on a normal internal Webserver.

If you start Jabber you have to say ok, only after that it's registering.
so we have made a small integration that updates your UC Service Profile
(or if u aren't running a 12.x Version of CUCM) or your Device Config to do
not have this Configured for 3 Months, so u start Jabber and it get's
registered.

You should always test the Popup on Release Changes

Rapidly and Jabber is something .

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<https://twitter.com/flohATinnsbruck>
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Am Di., 11. Jan. 2022 um 16:37 Uhr schrieb Adam Pawlowski :

> Is anyone else using the E911Notification popup for Jabber / Webex (UCM)?
>
>
>
> Where do you host it?
>
> Is there anything you do to try and mitigate an outage with it causing
> Jabber phones not to be available?
>
>
>
> Trying to come up with the best of the worst out of all of the choices I
> have available for where to put it, ideally that doesn’t require editing
> the Jabber config if the location has to change rapidly.
>
>
>
> Regards,
>
>
>
> Adam Pawlowski
>
> SUNYAB NCS
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Re: [cisco-voip] E164 Alt Ext and Cisco Unity

[Florian Kroessbacher]

Hy

what is in the SIP Message towards the Unity. Is there the Fully DN in the 
Message, so u should be able to change the forwarding Information to thst value 
on the SIP Trunk to Unity with a LUA Script
Am 5. Okt. 2021, 22:19 +0200 schrieb Austin Williams 
:
> Hi All,
>
> We are running into an issue and have sought help from Cisco TAC who has not 
> been able to help us on this issue. Currently we are running a Full +E164 
> numbering Plan with both 5 digit Enterprise Alternate numbers and +E.164 
> Alternate numbers. So to give an example of a Test phone I have set up:
>
>
> • Directory number: \+150259663629
> • Enterprise Alt
> • Number Mask: X
> • Alternate Number: 63629
> • Route Partition: -EALT-PT
> • +E.164 Alternate Number
> • Number Mask: XX
> • Alternate Number: 5025966329
> • Route Partition: 0-KHC-EntAltExt
>
>
> When this phone is dialed by either the 5 digit or 10 digit and forwards to 
> voicemail the redirected and last redirected number that Unity Receives is 
> the Dialed number, either the 5 digit or 10 digit, and as we have not 
> configured ALT extensions on the mailboxes it fails to find the correct box. 
> If I set the Alt ext on each Unity mail box it does function and I can do 
> that for the 10 digit, I am going to run into an Issue setting the 5 digit as 
> an Alt extension on the mailboxes as we are going to overlap on multiple 
> sites.
>
> What we are trying to do is find a way to tell the call manager to send the 
> full +1XX as the Last Redirected Number when either the 11 digit 
> full, 10 digit or 5 digit number is dialed. We have tried a Called Number 
> Transformation to no result and have messed with the device pool setting  
> “Redirected transformation CSS” on a test DP to no effect either.
>
> Does anyone have any suggestions
>
> CUCM version: 12.5.1.14900-63
> Unity Version: 12.5.1.14900-45
>
> Our Unity mailboxes are built with +1XX as their D
>
> Thanks,
> Austin John Williams
> Email: austinwi...@gmail.com
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Re: [cisco-voip] Pick up groups with no members SQL quire CUCM 10.5

[Florian Kroessbacher]

Hy

i think u only have to following

select name from pickupgroup where pkid not in (select distinct fkpickupgroup 
from pickupgrouplinemap)
On 18. May 2021, 10:11 +0200, Matthew Collins , wrote:
> Hi All,
>
> Can someone point me I right direction for a SQL query I can run to show me 
> all pickup groups with no directory number associated with them.
>
> I had a search and can find a command that will return all pickup groups with 
> their members.
>
> run sql select n.dnorpattern, n.description, pg.name as PickupName from 
> numplan n join  pickupgrouplinemap pglm on n.pkid = pglm.fknumplan_line join 
> pickupgroup pg on pglm.fkpickupgroup = pg.pkid
>
>
> I could work backwards from that output, but hoping someone has the knowledge 
> to just show pickup groups with no members.
>
> Thanks in advance.
>
>
>
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Re: [cisco-voip] digitized voice quality

[Florian Kroessbacher]

Hy @ all

we have this on normal Jabber and Jabber for VDI on some Devices with BiB 
Recording enabled.

Interessting would be this i think

https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-windows/212176-How-to-collect-an-Audio-Capture-Dump-wit.html

cheers
On 15. Jul 2020, 17:12 +0200, Palmer, Brian via cisco-voip 
, wrote:
> Call recording might be one mechanism to verify what the user is experiencing 
> unless you have experienced the same.  Quality of service is always a thought 
> as well.  Just throwing additional thoughts into what everybody else 
> mentioned.. all of which are great.
>
>
>
>
> From: cisco-voip  On Behalf Of Scott Voll
> Sent: Monday, July 13, 2020 10:49 AM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] digitized voice quality
>
> All--
>
> we are using Jabber 12.8 with CM 12.5 in a voice only mode.  we have users 
> reporting digitized voice call quality.  has anyone else been dealing with 
> this?
>
> we have had multiple users say if they add Video to the call it makes it 
> better.  Anyone else seeing that?  Any ideas why that would be better?
>
> what else can we do to make this work better?  TAC has not been able to help 
> with this after multiple weeks of engagement.
>
> any thoughts?
>
> TIA
>
> Scott
>
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Re: [cisco-voip] 15m10s timer for Jabber

[Florian Kroessbacher]

Hy out there

i think this could be the standard reinvite timer. Is there some Firewall in 
between which blocks out the reinvite
Am 16. Apr. 2019, 05:59 +0200 schrieb Lelio Fulgenzi :
>
> Was having constant disconnects on a off-premise (expressway) Jabber 
> conversation and before I blamed the other person, I thought I’d look into 
> the length of the calls that were cut off.
>
> Turns out each of the three times I was cut off, the calls lasted 15m10s. I 
> went back to some calls two weeks ago and same thing, cut off at 15m10s.
>
> This smells of a timer.
>
> Any ideas?
>
> I’ll have to open a case, after testing on-prem call length.
>
> Lelio
>
>
> -sent from mobile device-
>
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
> <>
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Re: [cisco-voip] better debug ccsip messages invites?

[Florian Kroessbacher]

load it into translatorx

--
Florian Krößbacher
florian.kroessbac...@gmail.com

Am 21. Sep. 2018, 18:03 +0200 schrieb Anthony Holloway 
:
> You're going to want "show log | i INVITE sip:" to get request lines, 
> otherwise you'll also get other Headers which contain the word INVITE.  E.g., 
> the Cseq Header.
>
> But yes, I agree that viewing SIP message traces inside of IOS is painful, 
> and it's always better to pull the information out and view it offline.  Even 
> if it's just Notepad++.
>
> It would be nice if Cisco had some specific SIP show commands, like how we 
> have show call active|history, maybe we could get show sip call 
> active|history [call-id], or something like that.
>
> > On Fri, Sep 21, 2018 at 8:32 AM Jason Aarons (Americas) 
> >  wrote:
> > >
> > >
> > > Please ask the devs to add some better sip debugs on CUBEs.
> > >
> > > How about debug ccsip messages invites or something that display filters 
> > > on type of sip message?
> > >
> > > I am getting better at show log | inc INVITE  
> > >
> > > Surely I can’t be the first to ask for this.
> > >
> > > -jason
> > >
> > >
> > >
> > > This email and all contents are subject to the following disclaimer:
> > > "http://www.dimensiondata.com/emaildisclaimer;
> > > ___
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Re: [cisco-voip] Customizing called party name/number display for route patterns

[Florian Kroessbacher]

Hy out there

what about CURRI if u were on >10
Am 20. Sep. 2018, 23:43 +0200 schrieb Lelio Fulgenzi :
>
> I half want to try the old trick of a CTI route point and use forwarding, but 
> not sure I want to intercept that complexity.
>
> -sent from mobile device-
>
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
> On Sep 20, 2018, at 5:32 PM, Brian Meade  wrote:
>
> > For name modification, CUCM doesn't have much.  Probably would need a LUA 
> > script to update the name in the SIP messaging.
> >
> > > On Wed, Sep 19, 2018 at 9:33 PM Lelio Fulgenzi  wrote:
> > > >
> > > > It’s been a while, and I know I can use route lists and route groups to 
> > > > customize called party number display but I can’t recall what my 
> > > > options are, if any, to customize the called party name display.
> > > >
> > > > For example, if I dial 4 and want only 4 to continue to 
> > > > display, I can do the modifications on the route list/group, ie prefix 
> > > > 999, and build appropriate rules on expressway.
> > > >
> > > > But what if I’d like to display “WebEx Pilot” on the phone when they 
> > > > call a particular route pattern?
> > > >
> > > > I can’t seem to find any option for that.
> > > >
> > > > -sent from mobile device-
> > > >
> > > > Lelio Fulgenzi, B.A. | Senior Analyst
> > > > Computing and Communications Services | University of Guelph
> > > > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | 
> > > > N1G 2W1
> > > > 519-824-4120 Ext. 56354 | le...@uoguelph.ca
> > > >
> > > > www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
> > > >
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Re: [cisco-voip] what are the cool kids using to read CUCM logs nowadays?

[Florian Kroessbacher]

Maybe this is something

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

--
Florian Krößbacher
florian.kroessbac...@gmail.com


Am 17. Mai 2018, 18:12 +0200 schrieb Lelio Fulgenzi :
> Awesome info! Thanks.
>
> My hope is to just get an idea of what’s going on. But I agree, I usually 
> tend to go to the TAC for more detailed help.
>
>
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
> From: Anthony Holloway 
> Sent: Thursday, May 17, 2018 10:56 AM
> To: Ryan Huff 
> Cc: Lelio Fulgenzi ; voyp list, cisco-voip 
> (cisco-voip@puck.nether.net) 
> Subject: Re: [cisco-voip] what are the cool kids using to read CUCM logs 
> nowadays?
>
> I also second TranslatorX as my first stop.  Make sure to have CDR enabled on 
> the cluster, and just dump all of the compressed logs into it, pop open the 
> Call List, find your call, double click it to see some good high level 
> details (disconnect cause, duration, codecs, dtmf, ip addresses, device 
> names) about the call, then generate a filter if you need to see the ladder 
> diagram.  I don't use it for more complex log analysis, for that I sail over 
> to Ryan's island and ask to use his Windows ME box and fire up Notepad++.
>
> The find in files feature is amazing, but you will need to un-compress the 
> logs first (RTMT can do this for you). The highlighting is helpful, and if 
> you know regex, you're golden.  PS Some folks in TAC have syntax highlighting 
> for CM traces in Notepad++, but so far they have not been willing to share it 
> with me.
>
> Which brings me to my final point.  Reading CM traces is not an easy effort, 
> and you should give yourself a time limit, then just open a TAC case (or send 
> it to Daniel Pagan; the human parser).  The Troubleshooting IP Telephony book 
> is pretty good, and there's a few nuggets on the mailing list, but really, if 
> you're not working in TAC, or a TAC like job function, you're swimming up 
> stream.  You might have better luck performing controlled experiments, 
> swapping components in and out to isolate the root cause.
>
> With that said, you'll never get better at log parsing, if you're not log 
> parsing.  So, if you're sick and twisted like Ryan and Daniel able to devote 
> some time to log parsing, go for it.
>
> On Thu, May 17, 2018 at 8:33 AM Ryan Huff  wrote:
> > quote_type
> > For the money (free) and usability; yeah, I’d go with Paul’s TranslatorX. 
> > Especially if your troubleshooting call behaviors because of its ability to 
> > organize by session ID and parse out OPTIONS messages.
> >
> > - Geany text editor does a good job because of its ability to highlight 
> > similar entities (Linux, not sure if there is a Windows port).
> >
> > - Notepad++ if I’m on an island and that’s all I have.
> >
> > Really though, anything that can parse the
> > character will work in a pinch but you’ll have to deal with the default 
> > chronologically logged readout. Whereas something like TranslatorX has more 
> > purposeful filters that allow you to follow the call’s message stack a 
> > little easier.
> >
> > -Ryan-
> >
> > On May 17, 2018, at 09:17, Lelio Fulgenzi  wrote:
> > >
> > > I need to do some log analysis to see why a certain group of fax machines 
> > > are getting reorder tone when trying to dial a number. I’ve got some 
> > > other troubleshooting steps to do first, but I have a feeling I’m going 
> > > to have to take a look.
> > >
> > > Is it translator X still?
> > >
> > > ---
> > > Lelio Fulgenzi, B.A. | Senior Analyst
> > > Computing and Communications Services | University of Guelph
> > > Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | 
> > > N1G 2W1
> > > 519-824-4120 Ext. 56354 | le...@uoguelph.ca
> > >
> > > www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
> > >
> > > 
> > >
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Re: [cisco-voip] Jabber IM service Issue

[Florian Kroessbacher]

Hy out there
Have u configured Revers DNS
Important!
>From version X8.8 onward, you must create forward and reverse DNS entries
for all Expressway-E systems, so that systems making TLS connections to
them can resolve their FQDNs and validate their certificates.

Page 19
https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-9/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-9.pdf

--

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2017-12-21 17:15 GMT+01:00 Ahmed Abd EL-Rahman <ahmed.rah...@bmbgroup.com>:

> Hi Brian,
>
>
>
> Under presence there is only one domain and all my servers are in the same
> domain which is the same internally and externally, we just have some users
> accounts on a different domain that’s why I added that domain just to EXP-E
> and added it’s related SRV records.
>
>
>
> Please have a look on the attached logs taken from both Expressway C and E
> for a client which works fine from internal network then connected
> successfully from outside but the IM service is not working from outside.
>
>
>
> I really do appreciate if you can hint me about possible reasons that make
> IM service not working from outside company network.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Best Regards
>
>
>
> Ahmed Abd EL-Rahman
>
> Senior Network Engineer
>
>
>
> *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
> Meade
> *Sent:* Tuesday, December 19, 2017 1:01 AM
> *To:* Ahmed Abd EL-Rahman <ahmed.rah...@bmbgroup.com>
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Jabber IM service Issue
>
>
>
> In IM under Presence->Domains, do you have multiple domains
> listed?  Do you have something other than default under Presence Advanced
> configuration for setting domain names?  Setting the voice services domain
> is one of the few reasons you need to login internally first.
>
>
>
> If you are using UserID@Default Domain then you shouldn't need to login
> internally first.  You may just need to remove the domain portion from the
> username on the username/password page when logging in the first time.
>
>
>
> On Mon, Dec 18, 2017 at 4:06 PM, Ahmed Abd EL-Rahman <
> ahmed.rah...@bmbgroup.com> wrote:
>
> What’s the flexible Jabber ID? Also I have local users created on CUCM
> with no LDAP integration.
>
>
>
> For the domain portion it’s not removed if the first login is through MRA
> and the login failed but as mentioned if I logged in thought inside network
> before the login through MRA goes smooth and login successfully but with
> one exception that IM service is not available.
>
>
>
>
>
>
>
>
>
>
>
>
>
> Best Regards
>
>
>
> Ahmed Abd EL-Rahman
>
> Senior Network Engineer
>
>
>
> *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
> Meade
> *Sent:* Monday, December 18, 2017 11:50 PM
>
>
> *To:* Ahmed Abd EL-Rahman <ahmed.rah...@bmbgroup.com>
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Jabber IM service Issue
>
>
>
> Are you using flexible Jabber ID?
>
>
>
> One thing to make sure is that first time MRA login gets rid of the domain
> portion in the username on the username/password form after entering
> username@domain in the service discovery window.
>
>
>
> On Mon, Dec 18, 2017 at 3:43 PM, Ahmed Abd EL-Rahman <
> ahmed.rah...@bmbgroup.com> wrote:
>
> Hi Brian,
>
>
>
> Will try what you suggested, but other than this issue I have another
> symptom, the user must use Jabber for first time from inside the network
> then afterwards he can use it from outside but if he tries the first time
> from outside the network it won’t register as all, even the login stage is
> not passed.
>
>
>
> Does this point to anything ?
>
>
>
>
>
>
>
>
>
>
>
>
>
> Best Regards
>
>
>
> Ahmed Abd EL-Rahman
>
> Senior Network Engineer
>
>
>
> *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
> Meade
> *Sent:* Monday, December 18, 2017 11:27 PM
>
>
> *To:* Ahmed Abd EL-Rahman <ahmed.rah...@bmbgroup.com>
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Jabber IM service Issue
>
>
>
> Try removing "inspect sip" from the global policy.  You don't want that in
> there with Expressway.  Not sure if XMPP traffic hits that as well or not.
>
>

Re: [cisco-voip] AXL or SQL to run a CTI Search like RTMT?

[Florian Kroessbacher]

Hy

i think u habe to do a join from
numplan (pattern)
to devicenumplanmap (patter 2 device)
and device
table

Maybe
Datadictionary Helps
https://developer.cisco.com/media/UCM10.5DataDictionary/UCM10.5DataDictionary.htm

--
Florian Krößbacher
florian.kroessbac...@gmail.com


Am 26. Okt. 2017, 16:48 +0200 schrieb Nick Barnett :
> We consistently have issues around here where contact center users receive 
> error 10159 "This is a shared line"  This happens because our contact center 
> uses extension mobility. To remedy the error, you have to launch RTMT, go to 
> Voice/Video, CTI then CTI Search. At that point, you put the users number in 
> the search box and hit Finish. It comes back with a table of devices using 
> that number. From there, you have to go to the application user and 
> disassociate all (usually just 2) of the devices returned from the RTMT 
> search.
>
> Is RTMT the only place that has this info? I just went through the AXL 
> toolkit and didn't see anything that looked promising. Any ideas on if this 
> info can be pulled down from a SQL query?
>
> I'm working on automating this whole process because that's the new thing 
> everyone likes to do :)  I already have the code that will remove the 
> appropriate device from the app user and reassociate it successfully. I just 
> need to know which devices need to be removed and added back, and the only 
> place I can find that info is RTMT.
>
> Thanks,
> Nick
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Re: [cisco-voip] Jabber 11.8 issue

[Florian Kroessbacher]

Maby your IM and Prsence server are out of sync, and the user getd no one time 
password

--
Florian Krößbacher
florian.kroessbac...@gmail.com


Am 19. Okt. 2017, 18:11 +0200 schrieb norm.nichol...@kitchener.ca:
> I have a user getting this error:
>
>
>
> And she is the only Jabber user getting it. I have reset everything 
> associated with the CSF but no joy.
>
>
> Any assistance would be appreciated
>
>
>
>
> Thanks
>
>
>
>
>
> Norm Nicholson
> Telecom Analyst
> City of Kitchener
> (519) 741-2200 x 7000
>
>
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Re: [cisco-voip] e164 dialplan conversion

[Florian Kroessbacher]

Hy out there,

throug updateLine this is working

use old pattern & partition
and set newpattern

we have done this for 12000 Lines

Am 21. Sep. 2017, 15:13 +0200 schrieb Bill Talley :
> I know there have been some conversations around this in the past, but I’m 
> hoping there are new methods for converting from a 4-digit to full e164 
> dialplan.
>
> Is there a way to change dialplan entries on CUCM and UCCX without having to 
> either individually touch each dialplan pattern or without deleting and 
> reimporting/recreating? Has anyone seen or used AXL/SOAP to automate 
> modification of dialplan patterns, or used PCP or some third party product to 
> accomplish this?
>
> Thanks for any feedback.
>
> Bill
>
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
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Re: [cisco-voip] Identifying incoming Cell numbers - CM v8 and above

[Florian Kroessbacher]

Hy out there

maybe do this with RemoteDestinations

in CUCM v10 plus uncan do it with an ECCP (through CURRI webservice, cause in 
this version u can do a modify caller name or modify calling name 
(inbound/outbound).

Am 19. Mai 2017, 07:08 +0200 schrieb Gary_Bates_Command_Solutions 
:
>
> Subject: Identifying incoming Cell numbers - CM v8 and above
> > My client has a need to save the cell numbers in CM , when a caller calls 
> > in from a known cell number , they want to display the persons name .
> >
> >
> > Can cell numbers be mapped to a name in CM ?
> >
> > Thx
> > Gary Bates
> > ACT
> > Sent from my iPhone
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Re: [cisco-voip] Jabber Issues

[Florian Kroessbacher]

Hy

Which directory integration are u using, maybe there is a config issue with the 
new CDI,
i was seeing problems iff u are using edi params in the jabber-config.xml and 
the cdi Params weren't there.
So maybe u hae to configure the CDI params wright.

Am 7. Apr. 2017, 07:56 +0200 schrieb naresh rathore :
>
> hi All
>
>
>
>
>
>
>
>
>
>
> i am currently facing two issues with Jabber.
>
>
>
>
>
>
>
>
>
>
> 1. jabber is used in Phone only mode on this particular computer. CUCM 
> version is 10.5.2. when I use jabber 11.7, i can see my mobile number under 
> my Name but when i install jabber 11.8.3 on my windows 7 machine, instead of 
> number, i see my email under my name on top.
>
>
>
>
>
>
> 2. Also on Jabber 11.7, when i hover over the user, it automatically shows 
> the user profile. but on Jabber 11.8.3 when i hover over the user, nothing 
> happens and i have to do righclick and see the profile.
>
>
>
>
>
>
>
>
>
>
> it works Ok with Jabber 11.7 but doesnt work if i upgrade to Jabber 11.8.3
>
>
>
>
>
>
>
>
>
>
> Regards
>
>
>
>
>
>
> Naray.
>
>
>
>
>
>
>
>
>
>
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Re: [cisco-voip] 8861 display

[Florian Kroessbacher]

Have u only installed the firmware or the device pack also?

I think its the "Actionable Incoming Call Alert"

cheers

--

Florian Krößbacher

florian.kroessbac...@gmail.com (mailto:florian.kroessbac...@gmail.com)

On 25 Aug 2016, 18:03 +0200, Scott Voll , wrote:
> here is an example. I have blacked out numbers and names to protect the 
> innocent.
>
> Thanks
>
> scott
>
>
> On Fri, Aug 19, 2016 at 3:17 PM, Scott Voll  (mailto:svoll.v...@gmail.com)> wrote:
> > Nope I've noticed it on my phone also.. i've just ignored it, but 
> > now I have users asking ;-)
> >
> > Scott
> >
> >
> > On Fri, Aug 19, 2016 at 3:16 PM, Brian Meade  > (mailto:bmead...@vt.edu)> wrote:
> > > Do you have a black bar at the top as well?
> > >
> > > Make sure the faceplate is all the way on. I see people pull these off 
> > > and put them back on with the bottom not all the way pushed in resulting 
> > > in this type of stuff.
> > >
> > > On Fri, Aug 19, 2016 at 6:07 PM, Scott Voll  > > (mailto:svoll.v...@gmail.com)> wrote:
> > > > When someone calls her directly the options to answer or decline are 
> > > > partially showing, like the display has shifted down. Anyway of 
> > > > resetting this or adjusting the display?
> > > >
> > > > CM 10.5.2
> > > > firmware 11.x
> > > >
> > > > TIA
> > > >
> > > > scott
> > > >
> > > >
> > > > ___
> > > > cisco-voip mailing list
> > > > cisco-voip@puck.nether.net (mailto:cisco-voip@puck.nether.net)
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> >
>
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[cisco-voip] ILS Service

[Florian Kroessbacher]

Hy VOIP Guys,

which Ports must be opened between the Clusters.

only
TCP 7501 (Used by Intracluster Lookup Service (ILS) for certificate based
authentication.)
TCP 7502 (Used by ILS for password based authentication.)

or is there a need for

TCP 8006 (ILS SDL Port) as well

--

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florian.kroessbac...@gmail.com

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Re: [cisco-voip] Cisco 7960 SCCP enbloc forwarding

[Florian Kroessbacher]

hy out there


on the 7940/60 it's not possible to do enbloc.





--

Florian Kroessbacher 
florian.kroessbac...@gmail.com

On Fri, Jan 16, 2015 at 2:03 AM, Claiton Campos claitoncam...@gmail.com
wrote:

 Does anyone know if you can enable Cisco 7960 SCCP call forwarding enbloc
 instead of digit by digit in CUCM 10.5?
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Re: [cisco-voip] External Call Control Profile

[Florian Kroessbacher]

hy


i thnxs


--

Florian Kroessbacher 
florian.kroessbac...@gmail.com

On Sat, Jan 10, 2015 at 6:57 PM, Anthony Holloway
avholloway+cisco-v...@gmail.com wrote:

 I haven't read the doc yet, and I will, but this seems correct.
 The whole purpose of CURRI is to enhance the routing decision beyond the
 capabilities of CUCM. Which are basically route or block, or ToD filtering.
 If you want to apply an ECCP to all calls being made by an extension, you
 might be able to place a catch all translation pattern in front of all
 other patterns.
 Example:
 1. DN 1000 dials 2000
 2. XLATE  with ECCP attached
 3. DN 2000
 I'll reply again if I find verbiage to support my thought, but my gut tells
 me I won't find anything. For the same reason you won't find documentation
 stating you can't make a call from a route pattern, but you can from a
 translation pattern. It's sort of implied.
 On Fri, Jan 9, 2015 at 6:11 PM Florian Kroessbacher 
 florian.kroessbac...@gmail.com wrote:
 Hy Guys

 I'm Trying do do something with CURRI.
 So i'm testing now my Application.

 The Setting is made on Directory Number BASE (Callmanger 10.5.1.1-7)
 when will Callmanger go to the External Call Controll Server
 * Only on outbound Calls
 * Only on inbound Calls
 * For all Call's inbound/outbound

 In my Test i only see Calls which are going to the Directory Number doing
 Webrequest to the Routing Server

 is this per Design or do i have to Contact TAC

 --

 Florian Kroessbacher
 florian.kroessbac...@gmail.com

 [image: Twitter] https://twitter.com/flohATinnsbruck [image: Google
 Plus] https://plus.google.com/+FlorianKroessbacher [image: Linkedin]
 https://www.linkedin.com/profile/view?id=115928906
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[cisco-voip] External Call Control Profile

[Florian Kroessbacher]

Hy Guys

I'm Trying do do something with CURRI.
So i'm testing now my Application.

The Setting is made on Directory Number BASE (Callmanger 10.5.1.1-7)
when will Callmanger go to the External Call Controll Server
* Only on outbound Calls
* Only on inbound Calls
* For all Call's inbound/outbound

In my Test i only see Calls which are going to the Directory Number doing
Webrequest to the Routing Server

is this per Design or do i have to Contact TAC

--

Florian Kroessbacher
florian.kroessbac...@gmail.com

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Re: [cisco-voip] Basic Jabber Deploy question

[Florian Kroessbacher]

Hy

Look here
https://developer.cisco.com/media/cisco-user-data-serviccs-usd-dev-guide/index.html?getting_started.html

The Client goes to the UDS Service and get's all he needs to know where to
go

cheers

--

Florian Kroessbacher
florian.kroessbac...@gmail.com

[image: Twitter] https://twitter.com/flohATinnsbruck [image: Google Plus]
https://plus.google.com/+FlorianKroessbacher [image: Linkedin]
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2015-01-03 20:54 GMT+01:00 Ed Leatherman ealeather...@gmail.com:

 Interesting, thanks Josh. So does the client just pick a node running tftp
 service on it's own then based on the information from UDS? The reason I
 ask is I only have 2 nodes running that right now and its sounds like
 either one could potentially be used by a client at any given time.

 On Sat, Jan 3, 2015 at 12:31 PM, Josh Warcop j...@warcop.com wrote:

 Great question. There is a mixture of TFTP/UDS and version 9.x or 10.x
 determines what is used and when.

 If you deploy the _cisco-uds SRV record you don't need to specify TFTP
 because the client discovers that service via UDS from the cluster.


 --
 Date: Sat, 3 Jan 2015 10:38:57 -0500
 Subject: Re: [cisco-voip] Basic Jabber Deploy question
 From: ealeather...@gmail.com
 To: j...@warcop.com
 CC: kev...@advancedtsg.com; cisco-voip@puck.nether.net


 Thanks Josh,

 As a follow up question then, lets say my jabber client happens to
 resolve _cisco-uds to a node that isn't running TFTP service - what tells
 it an IP address/hostname of a TFTP node to collect the phone
 configuration, or is that delivered via the UDS connection?

 Kevin: CUCM v9, not running IMP just using jabber in phone-only mode.
 Literally just using it as softphone/video client on some laptops.

 On Wed, Dec 31, 2014 at 3:57 PM, Josh Warcop j...@warcop.com wrote:

 Seems to be some confusion in this area...

 UDS runs on Tomcat and accessed over HTTPS and with recent versions
 supports SAML SSO. It's throttled so there isn't any difference pointing to
 subscribers or publisher first. The UDS service is a network service on
 each node and cannot be disabled. It has no direct dependencies on Call
 Manager or TFTP running on the same node.



 --
 From: kev...@advancedtsg.com
 To: ealeather...@gmail.com; cisco-voip@puck.nether.net
 Date: Wed, 31 Dec 2014 20:24:43 +
 Subject: Re: [cisco-voip] Basic Jabber Deploy question


  What version of CUCM/IMP?

 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Ed Leatherman
 *Sent:* Wednesday, December 31, 2014 3:06 PM
 *To:* Cisco VOIP
 *Subject:* [cisco-voip] Basic Jabber Deploy question

  Hello,

  I'm working on a project to deploy jabber in a limited roll-out as
 phone-only mode for a specific application. I'm interested in using srv
 records as it looks like it might be easier for end-user/students to sign
 into the client, so my question is related to that. I see that the
 _cisco-uds record is used for the client to find call manager; does this
 srv need to point to the publisher node, TFTP node, or CM node(s)? If not
 the TFTP node, how does the client know where pull it's configuration from
 once it has authenticated?

  I have jabber working OK manually just by configuring the client - I'm
 just looking for ease-of-use here for the students and staff that will be
 using it.

  Thanks for any clarification, most of the documents are centered around
 IMP which isn't what i'm after right now.

  Happy New Year!


  --
  Ed Leatherman

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 --
 Ed Leatherman




 --
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Re: [cisco-voip] 10.5 BAT Error/Question

[Florian Kroessbacher]

Hy

we have some Problems with BAT on 10.5, but the Proble is, that there are
Null Values in the file. In 8.6 we can export, change something or add
something and then reimport that.
In 10.5 this isn't working (cause of Null Values in the Export)

--

Florian Kroessbacher
florian.kroessbac...@gmail.com

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2014-12-26 18:18 GMT+01:00 Ryan Huff ryanh...@outlook.com:

 I think I know what the issue is but can't seem to verify ...

 - 4 node 10.5 Cluster with DNS enabled.
 - Currently, I have a known DNS resolution problem in the cluster, I know
 that and am working on it.

 What currently happens is I'll upload a file in BAT then run a job (like
 import) against the file I uploaded and the Job Scheduler log comes back
 with a 'success' result but N/A items processed. When I open the log I get
 Error; cannot reference /./filename.tar.

 I'm guessing that BAT is using the DNS resolver to reference
 node/path/file and since DNS is bunk right now, it can't resolve the node.
 I can't find in the Googles and docs where it definitively says BAT uses
 DNS when enabled.

 Is that my issue or should I be looking elsewhere?

 Thanks,

 Ryan

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Re: [cisco-voip] Cisco Prime License Manager behind a proxy

[Florian Kroessbacher]

Hy out there,

can u share the BUG/Feature Request ID

cheers Florian

Florian Kroessbacher

gmail: florian.kroessbac...@gmail.com


2014-06-24 17:14 GMT+02:00 Mike King m...@mpking.com:

 I'm happy with my answers at this point.  there is a Bug request logged so
 they know someone is looking for the feature, and there is an offline way
 of loading the licenses (Generating the PAK and downloading and then
 uploading it back)

 I'm not interested in a one off fix of TAC doing something via root access.



 On Tue, Jun 24, 2014 at 9:12 AM, Andrew Grech agrec...@gmail.com wrote:

 If you really want this feature maybe you could ask Cisco to login and
 root and edit your system environmental files, however you will lose the
 settings on a switch version.


 On Tue, Jun 17, 2014 at 1:46 AM, Mike King m...@mpking.com wrote:

  This shouldn't be this hard.

 Can the Cisco Prime License Manager  (Version 10) be placed behind a web
 proxy?

 Is there a set of directions?  I can't seem to find anything obvious.

 Mike

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Re: [cisco-voip] CUCM 10.5 and UDS Enhancements ?

[Florian Kroessbacher]

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/rel_notes/10_5_1/CUCM_BK_CE15D2A0_00_cucm-release-notes-1051.html--
Florian Kroessbacher
gmail: florian.kroessbac...@gmail.com

On Sun, May 18, 2014 at 8:01 PM, Heim, Dennis dennis.h...@wwt.com wrote:

 Has anyone seen release notes for 10.5 yet?
 Dennis Heim | Solution Architect (Collaboration)
 World Wide Technology, Inc. | 314-212-1814
 PS Engineering:  Innovate  Ignite.
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
 Jason Aarons (AM)
 Sent: Sunday, May 18, 2014 9:52 AM
 To: cisco-voip (cisco-voip@puck.nether.net)
 Subject: [cisco-voip] CUCM 10.5 and UDS Enhancements ?
 I understanding CallManager 10.5  got some UDS  (User Directory Services) 
 Enhancements.  However I can't find any documentation as to this.
 Did I miss something in the release notes about UDS performance improvements?___
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Re: [cisco-voip] Instructions on installing/configuring PostgreSQL on Windows for Presence and IM?

[Florian Kroessbacher]

hy

have u checked the local fw on the windows server for inbound connection
from the presence server.

have u done the config in the hba.conf and so on

Am Donnerstag, 20. März 2014 schrieb Jeffrey Girard :

 I have searched the Cisco site and there are tons of documents on how to
 configure the database for *nix installs.  I, unfortunately, am not skilled
 in that arena.



 I have downloaded and installed PostgreSQL on a Win2k8R2 server.



 I tried to interpret the Cisco doc instructions as best I could



 I have created a new database called tcmadb



 I have created a new entity under Login Roles called tcuser and gave that
 user superadmin rights



 I went back to the tcmadb database and made the tcuser the owner of the
 database



 In CUPS, I created the new external database:



   Database name - tcmadb

   Username - tcuser

   Pasword - password that I assigned to tcuser when I created it
 in the database

   Hostname - IP address of my win2K8 server

   Port Number - 5432.



 After several failures, I found a thread that indicated that I should
 create a new user in the Windows domain called tcuser.  I did that, and
 gave that user domain admin privileges.



 I get the failure to connect error under External Database Status



 Green check for reachability

 Failure on connectivity to database “Verify the hostname, username, and
 password are valid”



 Does it have anything to do with the postgres user account?



 Jeff





-- 
--
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Re: [cisco-voip] UCM 9.1 SIP Normalization Rules

[Florian Kroessbacher]

what about to change the destination port in the siptrunk config ??--
Florian Kroessbacher
gmail: florian.kroessbac...@gmail.com

On Fri, Mar 14, 2014 at 2:20 PM, Matt Slaga (AM)
matt.sl...@dimensiondata.com wrote:

 I'm attempting to apply a SIP Normalization rule to make a port adjustment on 
 SIP calls between UCM and Lync 2013.
 In the header, UCM sends the port number as 5060.  Lync has problems 
 digesting this when referring the call to another pool and wants to see the 
 header at 5061.
 The script we are attempting to use is below.  It has been applied to the 
 various SIP trunks between UCM and Lync.
 When reviewing traces, the port is not being adjusted.  Either the LUA script 
 is wrong below, or for some reason it is not being applied.
 Any thoughts?
 M = {}
 function M.outbound_INVITE(msg)
 local method, ruri, ver = msg:getRequestLine()
 local uri = string.gsub(ruri, 5060, 5061)
 msg:setRequestUri(uri)
 end
 return M
 [cid:image001.png@01CF3F65.E9BE66D0]
 [cid:image002.png@01CF3F66.5781C450]
 Matt Slaga
 Dimension Data
 Tel:+1-571-203-4132
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Re: [cisco-voip] Calling Party Name and Calling Calling Party Number in a SIP Trunk to VCS

[Florian Kroessbacher]

hy


what about a LUA script on the Sip Trunk, to put the right Display Name the 
customer wants.

Ther u can cut the left side and cut it at the point level 
--
Florian Kroessbacher
gmail: florian.kroessbac...@gmail.com

On Fri, Feb 14, 2014 at 1:30 AM, Jason Aarons (AM)
jason.aar...@dimensiondata.com wrote:

 Running CallManager 9.1 with VCS 8.1
 Is there a standard on how a SIP Video endpoint should show Calling Party 
 Name and Calling Calling Party Number when using URI dialing in SIP to SIP or 
 SIP to H323 (VCS Internetworking)
 Can I show a Friendly name like Jason Aarons vs 
 jason.aar...@coke.commailto:jason.aar...@coke.com on the screen when call 
 comes into VCS Endpoint from my Jabber?
 I have to admit I'm not sure what it shows but the customer said it wasn't 
 right.  Trying to get a camera picture of the screens.___
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