[cisco-voip] Pick up groups with no members SQL quire CUCM 10.5

[Matthew Collins]

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] CUCM upgrade from 10.5 to 11.5

[Matthew Collins]

Agree with Lelio but sometimes parallel upgrade are just not possible.

I’ve done 3 x 10.5 to 11.6 direct upgrade recently any only had one issue that 
wasn’t directly related to the upgrade. Other than that they have been pretty 
smooth.

In your pre-checks don’t just check DB replication but also CUC and UCCX 
replication on the respective servers (utils uccx/cuc dbreplication status). 
Plus check from the pub and sub as on a CUC upgrade recently it kept failing, 
Turned out CUC replication wasn’t set up correctly. Ran the command from the 
Pub and it stated everything was good, Ran from the Sub said it was failed. 
Ended up re-building the CUC Sub. The issue came about from a host name change 
years back and some of the tables where not updated on the sub.

Also double check the memory requirements. As 11.x requires 2 extra gig most of 
the OVA’s.

Upgrade timings going from 10.5 to 11.5/6

Memory upgrade where completed prior to upgrades.

CUCM Upgrade Pub 90 Mins 7.5k OVA
CUCM Upgrade Subs 60 Mins
IM&P Upgrade Pub 70 Mins 5K OVA
IM&P Upgrade Sub 50 Mins

CUCM Switch Versions Pub 40 Mins
CUCM Switch Versions Subs 30 Mins
IM&P Switch Versions Pub 30 Mins
IM&P Switch Versions Sub 30 Mins

UCCX Upgrade Pub 80 Mins – 300 Agent OVA
UCCX Upgrade Sub 60 Mins
UCCX Switch Versions Pub 45 Mins
UCCX Switch Versions Sub 30 Mins



Regards

Matthew Collins


From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: 17 January 2019 16:37
To: SK ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM upgrade from 10.5 to 11.5


I will say one thing, if you have the opportunity to duplicate the environment 
in an off-line network and perform the upgrades there, then your migration will 
basically be downtime involved with shutting down old servers and turning up 
new servers. This is not trivial, by any stretch of the imagination. However, 
once you put some thought into it, I think you’ll find this method extremely 
valuable.

You work out all the kinks in the offline network and repeat until you are 
satisfied.

You’ll need a couple of things to make this work, namely, a change freeze 
window and the “pre 8.0” enterprise parameter.

We’ve done this twice now and it’s worked like a charm.

I would hate to do upgrades on live systems.

Lelio


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

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

[University of Guelph Cornerstone with Improve Life tagline]

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of SK
Sent: Thursday, January 17, 2019 10:48 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] CUCM upgrade from 10.5 to 11.5

We are planning upgrade for our CUCM - CUC -UCCX platform from 10.5 to 11.5 
soon  . UCCX will be on 11.6.2 mostly .

I will appreciate any pointers on known bugs / challenges .

Thank you .
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[cisco-voip] Touch 10 Screen Damage

[Matthew Collins]

Hi All,

Does anyone know where a client could get a touch 10 screen replaced after one 
got smashed?

TIA

Matt


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[cisco-voip] Unity Upgrade 10.X to 11.5 and enterprise replication

[Matthew Collins]

Hi All,

Just a little heads up with Unity upgrades that I got stung with over the 
weekend.

>From what I have learnt via there is two type of replication with Unity 
>connection Db replication and CUC enterprise replication.

Prior to upgrading I checked utils dbruntimestate and everything was good on 
both servers, checked Show CUC cluster status from the Pub was all good, Pub 
was active and Sub was secondary.

Upgraded both servers no issues. Switch version on the pub failed, Checked the 
switch-version logs and could see that enterprise replication wasn't set up in 
the logs to halted with switch version.

Double checked utils dbruntimestate status and CUC cluster from the pub and all 
looked good still, utils dbruntimestate from the Sub was all ok but when I show 
CUC cluster status from the Sub both the PUB and SUB servers where showing up 
as secondary. Could not repair, Long story short had to rebuild the 10.5 build 
from scratch, Restore then upgrade. Looks like a IP change on the SUB last year 
during a DC move didn't update all the files it should have, When looking in 
the root with TAC there were still references to the old IP's for the Sub.

Main point is to check CUC cluster status from both the PUB and the SUB prior 
to any upgrade.


Regards
Matthew Collins
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[cisco-voip] H323 dialling separate IP's for single Expressway E

[Matthew Collins]

Hi all,

We have an external client that want to dial into our SIP VC estate using H323, 
We have this all set up using the format alias@IP and this is working as 
expected with expressway h323 to sip interworking. The trouble is this specific 
client is unable to dial Alias@IP and can only dial IP's on their own (don't 
get me started on this). We have set up a IVR to prompt the user to enter the 
meeting ID to redirect them to the correct endpoint but this is not ideal in 
this situation as they have a scheduler that auto dials the endpoint and cant 
enter DTMF after the call is established.

All the internal endpoints (MX800) are CUCM registered, I was thinking about 
setting up H323 on the devices but can see that H323 is not supported with CUCM 
provisioned MX800's.

Can anyone think of a work around for this, I'm pretty sure we are not able to 
do this without multiple Expressway E's.

We are running X8.9.1 on the expressways, CUCM 11.5.

Thanks in advance.



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[cisco-voip] Voice and video bandwidth generator

[Matthew Collins]

Hi All,

Can anyone recommend a tool that can generate voice and video traffic between 
two end point on the correct ports so we can test the networks performance to 
confirm QOS is in place prior to any UC build is in completed

Regards
Matthew Collins
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Re: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321

[Matthew Collins]

Unrelated to DSP but check out

http://www.cisco.com/c/en/us/support/docs/field-notices/641/fn64190.html

Router shows up as 98% memory utilisation when running IOS XE 3.X

Regards

Matthew Collins

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: 12 May 2017 21:21
To: NateCCIE 
Cc: Cisco VOIP 
Subject: Re: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321

What I am trying to express, although perhaps worded ambiguously, is that all 
your TDM resources are on the NIM and that I don't believe you could share the 
DSP from the backplane along with the NIM resources, for the purposes of TDM.

I would agree that you are correct, in that any unallocated DSP on a NIM could 
be configured for other dspfarm tasks (Ex. Transcoding/Transrating, Hardware 
MTP .. etc).

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

Sent from my iPhone

On May 12, 2017, at 4:03 PM, NateCCIE 
mailto:natec...@gmail.com>> wrote:
What do mean by shareable to the backplane?  It is my understanding that unused 
DSPs on a NIM can be used for conferencing/transcoding/MTP.

Sent from my iPhone

On May 12, 2017, at 2:00 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I think that is where my lack of specificity comes into play; the NIM 
conversation I thought I was participating in was an extension of a convo this 
AM, regarding a T1 PRI, in which those DSP are reserved for TDM only, and not 
shareable to the backplane or vice a versa.

Thanks,

Ryan

On May 12, 2017, at 3:46 PM, NateCCIE 
mailto:natec...@gmail.com>> wrote:
TDM DSPs on the 4ks have to be on the NIM because there is no shared TDM 
clocking backplane like there is on the ISRs/ISR G2.

Dspfarm stuff can use extra dsps on a NIM and the motherboard DSPs can only be 
used for dspfarm tasks.

Sent from my iPhone

On May 12, 2017, at 1:35 PM, Jose Colon II 
mailto:jcolon...@gmail.com>> wrote:
I was under the same assumption of why there was a dsp slot on the NIM. I know 
I read a Cisco doc somewhere that lead me that direction.

On May 12, 2017 2:28 PM, "Ryan Huff" 
mailto:ryanh...@outlook.com>> wrote:
So this is interesting; I was under the impression the backplane DSP could not 
extend to the NIM (and is the fundamental reason, among others, that the NIM 
has its own DSP)  looks like I have a new lab task :).

On May 12, 2017, at 3:09 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
Rashmi Patel ( rashmika.pa...@zones.com<mailto:rashmika.pa...@zones.com> ) - 
12:31 PM
Q: Does that mean conference resource will be used from NIM DSP not from mother 
board DSP resources
Priority: N/A
Dolan Spitler - 12:57 PM
A: When it comes to IP services (xcoding, conferencing, MTP) The motherboard 
DSP can be pooled with the NIM DSPs to increase the DSPfarm scale

Source: https://communities.cisco.com/docs/DOC-7823 (look for the 4000 event on 
Oct 16th)

On Fri, May 12, 2017 at 1:17 PM Brian Meade 
mailto:bmead...@vt.edu>> wrote:
My understanding is any dspfarm resources such as conferencing/transcoding use 
the motherboard resources while the ones on the NIM are just for the voice 
ports themselves.

On Fri, May 12, 2017 at 11:51 AM, Jose Colon II 
mailto:jcolon...@gmail.com>> wrote:
If you will be using DSP's you will need to decided if you will need them on 
the motherboard or on the T1 NIM. On the motherboard I believe it can only be 
used for conferencing. You will need them on the NIM for transcoding.

On May 12, 2017 6:42 AM, "Ryan Huff" 
mailto:ryanh...@outlook.com>> wrote:
T1 PRI commands are substantially different if that is in play.

Sent from my iPhone

On May 12, 2017, at 7:35 AM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
The vast majority of commands are the same. Netflow stuff is changed completely 
if you use that.  Outside of updating interface names, most of our templates 
just worked.



Matthew G. Loraditch – CCNP-Voice, CCNA-R&S, 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 
norm.nichol...@kitchener.ca<mailto:norm.nichol...@kitchener.ca>
Sent: Friday, May 12, 2017 7:31 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Migrating a Cisco 2901 to a Cisco 4321


We have a base config we use for building our 2901/11’s . Will this work on the 
4321 or do I have to start from scratch.



Thanks




Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000


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c

[cisco-voip] UCCX and CUCN HA decommission

[Matthew Collins]

Hi All,

I have got a requirement (against all advise) to permanently delete a UCCX and 
Unity Connection HA servers, There doesn't seem to be any documentation on 
doing this. They are running CSR 10.0.

For Unity I was planning on

Stop all services on HA Unity Server

Delete Call group HA Unity Server-1

Delete Ports with server HA Unity Server

Delete server HA Unity Server from Cluster Page

Reboot


For UCCX I was planning on

Disable CDS and HDS on Primary server

Stop all services on HA UCCX SERVER

Edit UCM Config to remove all reference to DR CUCM

Delete server HA UCCX SERVER from Cluster Page

Perform Cisco Unified CM Telephony Data Synchronization

Reboot



If anyone can point me to any documentation of have any gotchas that would be 
great.

TIA

Matt
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Re: [cisco-voip] Future of Jabber .......

[Matthew Collins]

Received this in one of the Cisco Customer Connection: Briefing Registration 
invites last week

"November 17, 2016 - Revised Jabber Roadmap Update
Cisco Collab Technology Group listened to and has acted upon your feedback on 
the Cisco Jabber roadmap that was delivered on September 15, 2016! Please join 
us to hear about the revised Jabber roadmap for post R11.8 capabilities that 
will include iOS Voice/Video Push notifications (in addition to IM/P 
notifications in R11.8), Multi-Line call support, Contact Center Feature 
enhancements, and more features and capabilities!"

Regards

Matthew Collins 


-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Andrewartha
Sent: 02 November 2016 06:45
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Future of Jabber ...

On 03/10/16 02:11, Lelio Fulgenzi wrote:
> I've seen only one update in the forums, and while it's a good start, 
> there's still some outstanding issues people have concerns with.

Reading the forum now, it looks like they've listened and are going to add 
support IM/phone push notifications by the time iOS 11 comes out.

--
James Andrewartha
Network & Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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[cisco-voip] Limiting the CSQ visible in CUIC reports.

[Matthew Collins]

Hi All,

I’m looking for a way to limit the CSQ a user can report on using CUIC and UCCX 
version 10.6.1.

I’m able to limit the user to view only a saved custom report that I have 
created, and although the end user has the correct CSQ populated in the report 
when they open it, they can edit it and select all the other CSQ should they 
wish before they hit run.

Any help greatly appreciated.

Matthew Collins



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Re: [cisco-voip] jabber-config.xml

[Matthew Collins]

Thanks Sebastian. Will give that a go.

Regards

Matthew Collins 

-Original Message-
From: Sebastian Hagedorn [mailto:haged...@uni-koeln.de] 
Sent: 08 September 2016 09:51
To: Matthew Collins 
Cc: voip puck 
Subject: Re: [cisco-voip] jabber-config.xml


--On 8. September 2016 um 08:13:39 + Matthew Collins  
wrote:

> Has anyone got this working for Mac users, Works fine for Windows but 
> never got it working for Mac users?
>
> My file is the same but has the addition of the below in for Mac, I 
> know the version numbers are now old but this is when I was testing.
>
>   
> 202336
> 10.6.0
> * Test Text * 
> http://internal.company.com/jabber/updates/CiscoJabberMac-10
> .6.0.202336.zip

For the Mac version you have to include the signature in the file name. 
Check the release notes. For example, the current release needs to look like 
this:

Install_Cisco-Jabber-Mac-11.7.0.241535-65675508-MCwCFAHDXcrORkwjFrrQgJtSk2D+7BDDAhQ5RD_GHUmzMZLEf6uvqyUnLsesiw!!.zip
-- 
.:.Sebastian Hagedorn - Weyertal 121 (Gebäude 133), Zimmer 2.02.:.
 .:.Regionales Rechenzentrum (RRZK).:.
   .:.Universität zu Köln / Cologne University - ✆ +49-221-470-89578.:.
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Re: [cisco-voip] jabber-config.xml

[Matthew Collins]

Has anyone got this working for Mac users, Works fine for Windows but never got 
it working for Mac users?

My file is the same but has the addition of the below in for Mac, I know the 
version numbers are now old but this is when I was testing.

  
202336
10.6.0
* Test Text * 
http://internal.company.com/jabber/updates/CiscoJabberMac-10.6.0.202336.zip
   


Regards

Matthew Collins






From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: 08 September 2016 05:12
To: Michael Voity 
Cc: voip puck 
Subject: Re: [cisco-voip] jabber-config.xml

Here is what the section in the jabber-config.xml looks like, assuming I have 
nothing else in this file (which I do, I just didn't share all of it):



 
  
http://internal.company.com/jabber/updates/jabber-update.xml
 


Here is what the whole jabber-update.xml looks like, which I'm hosting on an 
internal web server:



 
  42920
  11.7.0
  false
  
  
  
 
http://internal.company.com/jabber/updates/latest/CiscoJabberSetup.msi
 


Here is a screenshot of what the update looks like, or at least a preview of it:

[Inline image 1]

I download the updates myself from cisco.com<http://cisco.com>, and I get the 
build number from within the read-me after extracting the files.  I make the 
changes to the jabber-update.xml to reflect the what's new and the link to the 
release notes, and then finally I put the MSI in the latest folder, while 
moving the previous version to a versioned out folder structure for redundancy 
purposes, because I like duplicating all of the files on 
cisco.com<http://cisco.com> on my local server.  Just kidding, I trash the old 
file and only keep the latest.

On Wed, Sep 7, 2016 at 3:34 PM, Michael Voity 
mailto:mvo...@uvm.edu>> wrote:
Good Day,

Would someone in VoIP Puck land mind sharing a basic jabber-config.xml file?   
One that allows for auto jabber client updates.

I want to compare what mine looks like with someone’s that is working.

Thanks,

-Mike
--
Michael T. Voity
Network Engineer
University of Vermont


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Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

[Matthew Collins]

Just to close the loop.

Helios have come back and said that their IP Uni intercoms are not designed to 
call between other IP Uni intercoms. They are designed for use between a Helios 
IP Uni intercom and a normal phone.

They recommend their Verso and Vario intercoms for our clients.

Regards

Matthew Collins


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: 13 June 2016 11:23
To: Norton, Mike ; Cisco VoIP Group 

Subject: Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

Hi All,

Just a quick follow up. I managed to get some support with a little white lie. 
I emailed sales and said we were thinking about purchasing 50 more devices but 
unable to get two test units working. Surprise surprise I had a engineer on the 
phone within an hour.

Helios said two intercoms should work together but no one within the tech team 
had ever deployed or tested it. They are able to reproduce the issue and are 
working on a firmware update to fix.

Regards

Matthew Collins


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: 09 June 2016 09:08
To: Norton, Mike mailto:mikenor...@pwsd76.ab.ca>>; 
Cisco VoIP Group mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

Thanks for your reply's Mike and Neal,

We have hard coded the voice vlan as they were booting in the data vlan. From 
the traces its not looking like a network issues with qos ect.

Yes they register to the a CUCM system as 3rd party sip end points.

With regards to locations they are either side of a pair of 2 way airlock doors 
so there is no chance of getting feedback from each other. I did test two 
devices between different floors to completely rule this out but there was 
still issues.

Mike I think you may be right about the speakerphone logic as calls between a 
intercom and a desk phone are ok. Just annoying Helios point blank refuse any 
support.

Regards

Matthew Collins







From: Norton, Mike [mailto:mikenor...@pwsd76.ab.ca]
Sent: 08 June 2016 17:55
To: Matthew Collins mailto:mcoll...@block.co.uk>>; Cisco 
VoIP Group mailto:cisco-voip@puck.nether.net>>
Subject: RE: Helios IP Uni intercom units Dual Button.

Matthew - "The phones are registered and they can place calls between each 
other."

Are they actually meant to be used that way?

Sounds to me like the speakerphone logic is getting confused and/or sucks. E.g. 
the speakerphone logic on one side makes adjustments, it confuses the 
speakerphone logic on the other side, causing it to also make adjustments, 
which makes the first side readjust, etc. etc. Could be that they are meant 
more for calling a standard handset and not really meant for calling each 
other. Just a guess.

Furthermore, are they within "earshot" of each other? 'Cause that would 
definitely cause feedback and weird speakerphone behaviour.

-mn


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: June-08-16 9:18 AM
To: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Helios IP Uni intercom units Dual Button.

Hi all,

I have been asked to configure some Helios IP Uni intercom units and register 
them to the CUCM, I have been able to compete this, The phones are registered 
and they can place calls between each other but the audio quality is poor, and 
when I say poor I don't mean packet loss but the Audio volume keeps getting 
higher then lower, There is a lot of feedback. I might get 10 seconds of clear 
audio then its all start to go bad again. I have tested lots of the audio 
settings but not been able to get the right combo. I have also upgraded them to 
their latest firmware. I'm unable to log a call with Helios for support as we 
didn't purchase them direct. They are telling my to do to the supplier, But we 
are unsure where they were purchased from.

Any thoughts? What was anyone else's experience, Did they just work out of the 
box without tweaking the audio settings?

Thanks in advance.


Matthew
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Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

[Matthew Collins]

Hi All,

Just a quick follow up. I managed to get some support with a little white lie. 
I emailed sales and said we were thinking about purchasing 50 more devices but 
unable to get two test units working. Surprise surprise I had a engineer on the 
phone within an hour.

Helios said two intercoms should work together but no one within the tech team 
had ever deployed or tested it. They are able to reproduce the issue and are 
working on a firmware update to fix.

Regards

Matthew Collins


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: 09 June 2016 09:08
To: Norton, Mike ; Cisco VoIP Group 

Subject: Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

Thanks for your reply's Mike and Neal,

We have hard coded the voice vlan as they were booting in the data vlan. From 
the traces its not looking like a network issues with qos ect.

Yes they register to the a CUCM system as 3rd party sip end points.

With regards to locations they are either side of a pair of 2 way airlock doors 
so there is no chance of getting feedback from each other. I did test two 
devices between different floors to completely rule this out but there was 
still issues.

Mike I think you may be right about the speakerphone logic as calls between a 
intercom and a desk phone are ok. Just annoying Helios point blank refuse any 
support.

Regards

Matthew Collins







From: Norton, Mike [mailto:mikenor...@pwsd76.ab.ca]
Sent: 08 June 2016 17:55
To: Matthew Collins mailto:mcoll...@block.co.uk>>; Cisco 
VoIP Group mailto:cisco-voip@puck.nether.net>>
Subject: RE: Helios IP Uni intercom units Dual Button.

Matthew - "The phones are registered and they can place calls between each 
other."

Are they actually meant to be used that way?

Sounds to me like the speakerphone logic is getting confused and/or sucks. E.g. 
the speakerphone logic on one side makes adjustments, it confuses the 
speakerphone logic on the other side, causing it to also make adjustments, 
which makes the first side readjust, etc. etc. Could be that they are meant 
more for calling a standard handset and not really meant for calling each 
other. Just a guess.

Furthermore, are they within "earshot" of each other? 'Cause that would 
definitely cause feedback and weird speakerphone behaviour.

-mn


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: June-08-16 9:18 AM
To: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Helios IP Uni intercom units Dual Button.

Hi all,

I have been asked to configure some Helios IP Uni intercom units and register 
them to the CUCM, I have been able to compete this, The phones are registered 
and they can place calls between each other but the audio quality is poor, and 
when I say poor I don't mean packet loss but the Audio volume keeps getting 
higher then lower, There is a lot of feedback. I might get 10 seconds of clear 
audio then its all start to go bad again. I have tested lots of the audio 
settings but not been able to get the right combo. I have also upgraded them to 
their latest firmware. I'm unable to log a call with Helios for support as we 
didn't purchase them direct. They are telling my to do to the supplier, But we 
are unsure where they were purchased from.

Any thoughts? What was anyone else's experience, Did they just work out of the 
box without tweaking the audio settings?

Thanks in advance.


Matthew
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Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

[Matthew Collins]

Thanks for your reply's Mike and Neal,

We have hard coded the voice vlan as they were booting in the data vlan. From 
the traces its not looking like a network issues with qos ect.

Yes they register to the a CUCM system as 3rd party sip end points.

With regards to locations they are either side of a pair of 2 way airlock doors 
so there is no chance of getting feedback from each other. I did test two 
devices between different floors to completely rule this out but there was 
still issues.

Mike I think you may be right about the speakerphone logic as calls between a 
intercom and a desk phone are ok. Just annoying Helios point blank refuse any 
support.

Regards

Matthew Collins







From: Norton, Mike [mailto:mikenor...@pwsd76.ab.ca]
Sent: 08 June 2016 17:55
To: Matthew Collins ; Cisco VoIP Group 

Subject: RE: Helios IP Uni intercom units Dual Button.

Matthew - "The phones are registered and they can place calls between each 
other."

Are they actually meant to be used that way?

Sounds to me like the speakerphone logic is getting confused and/or sucks. E.g. 
the speakerphone logic on one side makes adjustments, it confuses the 
speakerphone logic on the other side, causing it to also make adjustments, 
which makes the first side readjust, etc. etc. Could be that they are meant 
more for calling a standard handset and not really meant for calling each 
other. Just a guess.

Furthermore, are they within "earshot" of each other? 'Cause that would 
definitely cause feedback and weird speakerphone behaviour.

-mn


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: June-08-16 9:18 AM
To: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Helios IP Uni intercom units Dual Button.

Hi all,

I have been asked to configure some Helios IP Uni intercom units and register 
them to the CUCM, I have been able to compete this, The phones are registered 
and they can place calls between each other but the audio quality is poor, and 
when I say poor I don't mean packet loss but the Audio volume keeps getting 
higher then lower, There is a lot of feedback. I might get 10 seconds of clear 
audio then its all start to go bad again. I have tested lots of the audio 
settings but not been able to get the right combo. I have also upgraded them to 
their latest firmware. I'm unable to log a call with Helios for support as we 
didn't purchase them direct. They are telling my to do to the supplier, But we 
are unsure where they were purchased from.

Any thoughts? What was anyone else's experience, Did they just work out of the 
box without tweaking the audio settings?

Thanks in advance.


Matthew
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[cisco-voip] Helios IP Uni intercom units Dual Button.

[Matthew Collins]

Hi all,

I have been asked to configure some Helios IP Uni intercom units and register 
them to the CUCM, I have been able to compete this, The phones are registered 
and they can place calls between each other but the audio quality is poor, and 
when I say poor I don't mean packet loss but the Audio volume keeps getting 
higher then lower, There is a lot of feedback. I might get 10 seconds of clear 
audio then its all start to go bad again. I have tested lots of the audio 
settings but not been able to get the right combo. I have also upgraded them to 
their latest firmware. I'm unable to log a call with Helios for support as we 
didn't purchase them direct. They are telling my to do to the supplier, But we 
are unsure where they were purchased from.

Any thoughts? What was anyone else's experience, Did they just work out of the 
box without tweaking the audio settings?

Thanks in advance.


Matthew
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Re: [cisco-voip] Present Callers queued time to UCCX Finesse agent desktop

[Matthew Collins]

Thanks Ed,

I’ll give that ago tomorrow and update.

Regards

Matthew Collins


From: Ed Leatherman [mailto:ealeather...@gmail.com]
Sent: 31 March 2016 14:31
To: Matthew Collins 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Present Callers queued time to UCCX Finesse agent 
desktop

Seems to me like once it gets to the connected branch, its no longer waiting 
which could account for the "Current" wait duration being -1.

Maybe if you change the "Connect" Setting to NO in the Select Resource step, 
which should change the Connected branch to Selected, then do your calculation, 
then manually connect?

I've never tried this before so its just a guess :)

On Thu, Mar 31, 2016 at 4:13 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi All,

Trying to present the callers queued time to the agent when the agent is 
presented the call but not have much luck.

What I have done so far is to get a reporting statistic current Wait Duration 
from CSQ IPCC Express and store that as a Integer Call Variable

[cid:image001.png@01D18B6C.FD28BF90]

I have played around with where I place the two steps but going around in 
circles. In the above example the agents time is queue is showing  blank and 
when I debug the Time in Queue is set at “-1”

TAI

Matt

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--
Ed Leatherman
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[cisco-voip] Present Callers queued time to UCCX Finesse agent desktop

[Matthew Collins]

Hi All,

Trying to present the callers queued time to the agent when the agent is 
presented the call but not have much luck.

What I have done so far is to get a reporting statistic current Wait Duration 
from CSQ IPCC Express and store that as a Integer Call Variable

[cid:image001.png@01D18B2D.095FFD90]

I have played around with where I place the two steps but going around in 
circles. In the above example the agents time is queue is showing  blank and 
when I debug the Time in Queue is set at "-1"

TAI

Matt
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[cisco-voip] Jabber Unity Connection Scaling on 5K OVA

[Matthew Collins]

Hi All,

Has anyone got another reference to the scaling of Cisco Jabber clients with 
the 5K OVA for Connection,

Both the 10.x 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/supported_platforms/10xcucspl.html#pgfId-850223
and 11.x 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/supported_platforms/11xcucspl.html#pgfId-851019
Supported Platforms List document seem to be wrong as the numbers for Jabber 
endpoints on the 2vCPU (5K OVA) go down with reduced single in box users 
instead of up.

TIA


[cid:image002.jpg@01D17627.DB3E9990]


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[cisco-voip] UPPERCASE lowercase host names

[Matthew Collins]

Hi All,

I'm sure I read somewhere on a Cisco Doc that all UC hostname should be set up 
using lowercase name. But I can't seem to find anything documented with the 
exception of CWMS documents.

Can anyone clarify this and hopefully point me to a relevant document.



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Re: [cisco-voip] Filtering out specific number from Jabber

[Matthew Collins]

Hi Ryan,

Thanks for getting back to me.

It's the visibly that the issue. The users add their personal mobile numbers in 
for emergency contact for HR  purposes. Everything was working fine on EDI but 
now they want to go to collab edge they will start seeing the numbers, Looks 
like we are going to have to use one of the "other" telephone fields to store 
these numbers. We will just have to change all the scripts that populate it and 
update the HR system to look at the other field.

I have also put in a request via the PDI helpdesk, If anything comes of that 
I'll share their findings but their initial response wasn't hopeful.

Does anyone know if you can block specific fields being imported from LDAP to 
stop them hitting the CUCM database in the first place?

Regards

Matthew Collins

D   +44 20 3005 3032
M  +44 7767 211 837


VC  mcoll...@block.co.uk<mailto:mcoll...@block.co.uk>
W   www.block.co.uk<http://www.block.co.uk>



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 11 January 2016 12:49
To: Matthew Collins 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Filtering out specific number from Jabber


Hello Matthew,



I apologize, this got away from me too, for the same reasons.



As I said, this is not an elegant solution and a bit clunky however, what you 
could do is write application dial rules in UCM that match the mobile numbers 
and then add or subtract digits that, in effect, make the call un-routeable (or 
route to a null location).



The most significant drawback is that it isn't masking or hiding the mobile 
number from the Jabber client, which is what you're really after I gather; this 
is just preventing the Jabber client from being able to natively dial it. Apart 
from that, if these numbers aren't in a range and are all over the place, you 
could end up with several application dial rules (and possibly 
conflicts/overlaps).



Depending on the other applications that need these mobile numbers, you might 
be able to adjust the other applications to look at different fields for the 
mobile number and then move the mobile numbers to different fields; thereby and 
ultimately, not making it available to the Jabber client.



Thanks,


= Ryan =





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

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

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

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

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


From: Matthew Collins mailto:mcoll...@block.co.uk>>
Sent: Monday, January 11, 2016 5:16 AM
To: Ryan Huff
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: RE: [cisco-voip] Filtering out specific number from Jabber


Hi Ryan,



Sorry to dig up an old thread, Everything got put on hold over Christmas and 
the new year,  but can I ask what your less elegant solution was?





Regards



Matthew Collins













From: Matthew Collins
Sent: 22 December 2015 14:20
To: 'Ryan Huff' mailto:ryanh...@outlook.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: RE: [cisco-voip] Filtering out specific number from Jabber



Thanks Ryan,



Only using UDS so end users get the same experience on premise or working Via 
MRA. UDS is the only supported directory lookup via MRA Unless something has 
changed that I missed.



So I'm working remotely signing in Via MRA, Running Jabber for windows 11.2.



I added the following





   telephoneNumber

   telephoneNumber





into my local jabber-config-user.xml that I use for testing xml settings that's 
stored in .\AppData\Roaming\Cisco\Unified Communications\Jabber\CSF\Config



This should of changed all mobile numbers to the users work numbers.



I reset Jabber and searched for a user that's not a stored user either in my 
address book or a jabber contact but the mobile numbers are still being pulled 
through.





Regards



Matthew Collins





From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:38
To: Matthew Collins mailto:mcoll...@block.co.uk>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Filtering out specific number from Jabber



When using UDS, Jabber doesn't query LDAP. It uses an API on CUCM and is 
limited to data held in the End Use Table of CUCM's database. You should still 
be able to use the config file to redirect field mapping within the Jabber 
client. I seem to recall making this work before.



Let me know your results, if it still doesn't work and you are still married to 
UDS. I do have another, much less elegant solution -but it works. Or, consider 
EDI.



Thanks,



Ryan



On Dec 22, 2015, at 8:20 AM, Matthew Collins 
mailto:mcoll...@block.c

Re: [cisco-voip] Filtering out specific number from Jabber

[Matthew Collins]

Hi Ryan,

Sorry to dig up an old thread, Everything got put on hold over Christmas and 
the new year,  but can I ask what your less elegant solution was?


Regards

Matthew Collins







From: Matthew Collins
Sent: 22 December 2015 14:20
To: 'Ryan Huff' 
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Filtering out specific number from Jabber

Thanks Ryan,

Only using UDS so end users get the same experience on premise or working Via 
MRA. UDS is the only supported directory lookup via MRA Unless something has 
changed that I missed.

So I'm working remotely signing in Via MRA, Running Jabber for windows 11.2.

I added the following


   telephoneNumber
   telephoneNumber


into my local jabber-config-user.xml that I use for testing xml settings that's 
stored in .\AppData\Roaming\Cisco\Unified Communications\Jabber\CSF\Config

This should of changed all mobile numbers to the users work numbers.

I reset Jabber and searched for a user that's not a stored user either in my 
address book or a jabber contact but the mobile numbers are still being pulled 
through.


Regards

Matthew Collins


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:38
To: Matthew Collins mailto:mcoll...@block.co.uk>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Filtering out specific number from Jabber

When using UDS, Jabber doesn't query LDAP. It uses an API on CUCM and is 
limited to data held in the End Use Table of CUCM's database. You should still 
be able to use the config file to redirect field mapping within the Jabber 
client. I seem to recall making this work before.

Let me know your results, if it still doesn't work and you are still married to 
UDS. I do have another, much less elegant solution -but it works. Or, consider 
EDI.

Thanks,

Ryan

On Dec 22, 2015, at 8:20 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi Ryan,

I was under the impression that you could only manipulate feild mappings when 
using BDI or EDI, I didn't think that you could with UDS.

I'll give it a test.

Regards

Matthew Collins


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:02
To: Matthew Collins mailto:mcoll...@block.co.uk>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Filtering out specific number from Jabber

Hey Matthew,

You cannot remove the presence of the field from the Jabber client (i.e hide), 
but you can control the value of the field via jabber-config.xml. So while 
"home phone" displays in the client, you could map it to another (blank) field.

Check out the Jabber xml generator if you have not already,

https://supportforums.cisco.com/document/106926/jabber-config-file-generator

Thanks,

Ryan



Sent from my iPad
On Dec 22, 2015, at 6:54 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi All,

Is there a way to filter out specify AD fields that the CUCM imports from AD, 
For Example AD contains Home telephone and we want to keep that in there as 
other applications need it but we don't want it to appear in the CUCM end users 
contact from either the phone or Jabber.

I know for Jabber we could use EDS/BDS and map the fields to null but we want 
to stick with UDS for Jabber so the users have the same experience internally 
and via edge.


Thanks in advance Matt
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Re: [cisco-voip] UCS Disk Partition for C series

[Matthew Collins]

Hi Andy,

For the raid configuration you will need to look up your TRC and configure raid 
accordingly.  http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware

Regards

Matthew Collins 


-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Andy
Sent: 08 January 2016 13:41
To: cisco-voip voip list 
Subject: [cisco-voip] UCS Disk Partition for C series

Hi,
I’ve just had some C240 m3 servers delivered for an upgrade, but there isn’t 
any partitioning or software pre installed, which is not necessarily and issue 
as the upgrade software should be available via PUT as there is current support 
contract.
My question is is there a doc for best practice in whats required for disk 
spaces etc.
I’m guessing that the 2 partition, 1 for ESXi and 1 for the VM machines


Andy
andy.ca...@gmail.com



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Re: [cisco-voip] Filtering out specific number from Jabber

[Matthew Collins]

Thanks Ryan,

Only using UDS so end users get the same experience on premise or working Via 
MRA. UDS is the only supported directory lookup via MRA Unless something has 
changed that I missed.

So I'm working remotely signing in Via MRA, Running Jabber for windows 11.2.

I added the following


   telephoneNumber
   telephoneNumber


into my local jabber-config-user.xml that I use for testing xml settings that's 
stored in .\AppData\Roaming\Cisco\Unified Communications\Jabber\CSF\Config

This should of changed all mobile numbers to the users work numbers.

I reset Jabber and searched for a user that's not a stored user either in my 
address book or a jabber contact but the mobile numbers are still being pulled 
through.


Regards

Matthew Collins


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:38
To: Matthew Collins 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Filtering out specific number from Jabber

When using UDS, Jabber doesn't query LDAP. It uses an API on CUCM and is 
limited to data held in the End Use Table of CUCM's database. You should still 
be able to use the config file to redirect field mapping within the Jabber 
client. I seem to recall making this work before.

Let me know your results, if it still doesn't work and you are still married to 
UDS. I do have another, much less elegant solution -but it works. Or, consider 
EDI.

Thanks,

Ryan

On Dec 22, 2015, at 8:20 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi Ryan,

I was under the impression that you could only manipulate feild mappings when 
using BDI or EDI, I didn't think that you could with UDS.

I'll give it a test.

Regards

Matthew Collins


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:02
To: Matthew Collins mailto:mcoll...@block.co.uk>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Filtering out specific number from Jabber

Hey Matthew,

You cannot remove the presence of the field from the Jabber client (i.e hide), 
but you can control the value of the field via jabber-config.xml. So while 
"home phone" displays in the client, you could map it to another (blank) field.

Check out the Jabber xml generator if you have not already,

https://supportforums.cisco.com/document/106926/jabber-config-file-generator

Thanks,

Ryan



Sent from my iPad
On Dec 22, 2015, at 6:54 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi All,

Is there a way to filter out specify AD fields that the CUCM imports from AD, 
For Example AD contains Home telephone and we want to keep that in there as 
other applications need it but we don't want it to appear in the CUCM end users 
contact from either the phone or Jabber.

I know for Jabber we could use EDS/BDS and map the fields to null but we want 
to stick with UDS for Jabber so the users have the same experience internally 
and via edge.


Thanks in advance Matt
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Re: [cisco-voip] Filtering out specific number from Jabber

[Matthew Collins]

Hi Ryan,

I was under the impression that you could only manipulate feild mappings when 
using BDI or EDI, I didn't think that you could with UDS.

I'll give it a test.

Regards

Matthew Collins


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: 22 December 2015 13:02
To: Matthew Collins 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Filtering out specific number from Jabber

Hey Matthew,

You cannot remove the presence of the field from the Jabber client (i.e hide), 
but you can control the value of the field via jabber-config.xml. So while 
"home phone" displays in the client, you could map it to another (blank) field.

Check out the Jabber xml generator if you have not already,

https://supportforums.cisco.com/document/106926/jabber-config-file-generator

Thanks,

Ryan



Sent from my iPad
On Dec 22, 2015, at 6:54 AM, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
Hi All,

Is there a way to filter out specify AD fields that the CUCM imports from AD, 
For Example AD contains Home telephone and we want to keep that in there as 
other applications need it but we don't want it to appear in the CUCM end users 
contact from either the phone or Jabber.

I know for Jabber we could use EDS/BDS and map the fields to null but we want 
to stick with UDS for Jabber so the users have the same experience internally 
and via edge.


Thanks in advance Matt
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[cisco-voip] Filtering out specific number from Jabber

[Matthew Collins]

Hi All,

Is there a way to filter out specify AD fields that the CUCM imports from AD, 
For Example AD contains Home telephone and we want to keep that in there as 
other applications need it but we don’t want it to appear in the CUCM end users 
contact from either the phone or Jabber.

I know for Jabber we could use EDS/BDS and map the fields to null but we want 
to stick with UDS for Jabber so the users have the same experience internally 
and via edge.


Thanks in advance Matt
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Re: [cisco-voip] MRA and Phone server selection

[Matthew Collins]

Hi Ryan,

No Apologies needed always appreciate the help gained on this forum.

The CCMCIP profile looks to be my problem, I didn’t have a CCMCIP profile 
created on the IM&P server. I have now created it and each time the jabber 
client signs in I’m getting phone services connecting.

When I look back through the logs I’m still seeing any combination of the 9 
CUCM servers being referenced as CCMCIPServers, Also even with the phone 
service connected I see different servers each time in the client. Currently I 
have both TFTP servers references as the phone services servers.

Either way it seems to be working now so I thank you for that.

Regards

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

[Matthew Collins]

13:56,424 INFO  [lcontrol/CallControlManagerImpl.cpp(567)] [csf.ecc.api] 
[setExtensionMobilityServers] - 
setExtensionMobilityServers(cucms02.company.co.uk, cucmp01.company.co.uk, 
cucmm01.company.co.uk)
13:13:56,424 INFO  [ontrol/src/callcontrol/EmHelper.cpp(266)] [csf.ecc] 
[setExtensionMobilityServers] - 
setExtensionMobilityServers(cucms02.company.co.uk, cucmp01.company.co.uk, 
cucmm01.company.co.uk)
13:13:56,424 INFO  [services/impl/ConfigServiceImpl.cpp(185)] 
[ConfigService-ConfigServiceImpl] [findConfig] - findConfig key : [TftpServer1] 
is an alias for key : [TFTPPrimaryServer] actual lookup done on this key - 
value : [cucmt01.company.co.uk]
13:13:56,424 INFO  [services/impl/ConfigServiceImpl.cpp(185)] 
[ConfigService-ConfigServiceImpl] [findConfig] - findConfig key : [TftpServer2] 
is an alias for key : [TFTPBackupServer1] actual lookup done on this key - 
value : [cucmt02.company.co.uk]
13:13:56,424 INFO  [services/impl/ConfigServiceImpl.cpp(185)] 
[ConfigService-ConfigServiceImpl] [findConfig] - findConfig key : [TftpServer3] 
is an alias for key : [TFTPBackupServer2] actual lookup done on this key - 
value : [cucmt01.company.co.uk]
13:13:56,425 INFO  [lcontrol/CallControlManagerImpl.cpp(527)] [csf.ecc.api] 
[setTFTPServers] - setTFTPServers(cucmt01.company.co.uk, cucmt02.company.co.uk, 
cucmt01.company.co.uk)
13:13:56,425 INFO  [services/impl/ConfigServiceImpl.cpp(185)] 
[ConfigService-ConfigServiceImpl] [findConfig] - findConfig key : [CtiServer1] 
is an alias for key : [CTIPrimaryServer] actual lookup done on this key - value 
: [cucms01.company.co.uk]
13:13:56,425 INFO  [services/impl/ConfigServiceImpl.cpp(185)] 
[ConfigService-ConfigServiceImpl] [findConfig] - findConfig key : [CtiServer2] 
is an alias for key : [CTIBackupServer] actual lookup done on this key - value 
: [cucms02.company.co.uk]
13:13:56,425 INFO  [lcontrol/CallControlManagerImpl.cpp(585)] [csf.ecc.api] 
[setCTIServers] - setCTIServers(cucms01.company.co.uk, cucms02.company.co.uk)
13:13:56,425 INFO  [control/CallControlManagerImpl.cpp(4379)] [csf.ecc.api] 
[setPreferredRegistrationMode] - setPreferredRegistrationMode (Default) to 
(Default)

Regards

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

[Matthew Collins]

Hi Ryan,

Just what the Expressway picked up during auto discovery.

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

Regards

Matthew





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

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

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

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

Within this cluster we have

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

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

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

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


Thanks in advance.

Matthew
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[cisco-voip] MRA and Phone server selection

[Matthew Collins]

Hi All,

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

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

Within this cluster we have

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

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

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

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


Thanks in advance.

Matthew
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Re: [cisco-voip] delete license file in CCx 10.5

[Matthew Collins]

Hi Scott,

I’ve had this before, I was unable to delete from the GUI, Tac got me to delete 
the licence via CLI from both UCCX servers then reboot each server.


Regards

Matthew


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
LaFountain (rlafount)
Sent: 07 October 2015 16:38
To: Scott Voll ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] delete license file in CCx 10.5

Hi Scott,

If it is a permanent license you can do it via the CLI:

Utils uccx list license
Utils uccx delete license 

Only temporary licenses can be deleted from the UI.

Thank you,

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

From: cisco-voip on behalf of Scott Voll
Date: Wednesday, October 7, 2015 at 11:30 AM
To: "cisco-voip@puck.nether.net"
Subject: [cisco-voip] delete license file in CCx 10.5

Why can't I delete a file in CCx 10.5.

Licensing gave me a bad one and now my call center is down.

Scott

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Re: [cisco-voip] Prompt change on unsuccessful transfer from Call handler

[Matthew Collins]

Thanks for the feedback Daniel.

Don’t think it viable to keep changing the files back via Tac after each 
upgrade or patch.

Regards

Matthew Collins


From: Daniel Pagan [mailto:dpa...@fidelus.com]
Sent: 05 August 2015 15:43
To: Matthew Collins ; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Prompt change on unsuccessful transfer from Call 
handler

Most if not all of the system default prompts are stored in the local 
filesystem and can’t be accessed without root permission. These prompts (two 
different files play to form that message) are two of those /PHGreet/ .wav 
files and cannot be changed. Unfortunately I’m unware of any method for 
changing this specific behavior aside from replacing the files by hand with a 
file using the same name via root - a request I worked on for a customer once…. 
Once ☺ Of course the files are overwritten after a patch or upgrade and I doubt 
TAC would assist with this again (but I can be wrong…)

Daniel

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: Wednesday, August 05, 2015 8:40 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Prompt change on unsuccessful transfer from Call handler

Hi all,

CUC 10.5.1.1-7

I have a system call handler and have enabled the during greeting option of 
Allow Transfers to Numbers Not Associated with Users or Call Handlers

[cid:image001.png@01D0D027.06C2DC40]

The problem we are experiencing is when the caller enters a extension number 
during the prompt that is engaged/busy and they don’t have a voicemail box the 
caller is played the prompt “You cannot be transferred to this number, check 
the number and try again”. So the end users keep trying the same number. I 
can’t seem to find out where that specific prompt is recorded to amend to a 
custom recording stating the number could be busy.

Thanks in advance



Regards

Matthew
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[cisco-voip] Prompt change on unsuccessful transfer from Call handler

[Matthew Collins]

Hi all,

CUC 10.5.1.1-7

I have a system call handler and have enabled the during greeting option of 
Allow Transfers to Numbers Not Associated with Users or Call Handlers

[cid:image001.png@01D0CF83.FA0CC270]

The problem we are experiencing is when the caller enters a extension number 
during the prompt that is engaged/busy and they don’t have a voicemail box the 
caller is played the prompt “You cannot be transferred to this number, check 
the number and try again”. So the end users keep trying the same number. I 
can’t seem to find out where that specific prompt is recorded to amend to a 
custom recording stating the number could be busy.

Thanks in advance



Regards

Matthew
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Re: [cisco-voip] LDAP Authentication when CUCM publisher is down.

[Matthew Collins]

Thanks for all your responses.

In our scenario when we lost one of a DC (planned maintenance) that hosted the 
CUCM, IM&P, and UCCX Pubs UCCX fineness agents were unable to log in. Had to 
resort to a couple of local back up agent accounts.

LDAP authentication wasn't working for any service (Jabber, CUCM Web Gui, UCCX 
Finesse Web Gui). I wasn't around to do any testing and as it was only a 4 
hours maintenance period it wasn't logged with us and no logs where collected.



Regards

Matthew Collins


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: 06 July 2015 15:24
To: Lelio Fulgenzi
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] LDAP Authentication when CUCM publisher is down.

I'll be interested to hear your results if you try!

I'm not sure that an ACL would do the trick though, probably would just show up 
in the traces as a time out. You'd probably have to stop the tomcat service on 
the pub (something to tell the cluster not to try and use the PUB as a bind 
source), which is pretty destructive on the pub in a working production 
environment (disclaimer: I do not advocate you do that).



Subject: Re: [cisco-voip] LDAP Authentication when CUCM publisher is down.
From: le...@uoguelph.ca<mailto:le...@uoguelph.ca>
Date: Mon, 6 Jul 2015 10:12:53 -0400
To: ryanh...@outlook.com<mailto:ryanh...@outlook.com>
CC: dpa...@fidelus.com<mailto:dpa...@fidelus.com>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
The worst case scenario, which we ran into, was a scenario where the pub is up 
and accepting auth requests but not able to process them. In our case the 
cluster was up for almost 300 days, and there were memory error alerts popping 
up. It would be nice for the system to understand this issue and go to the next 
node to try the auth process.

Interesting note about LDAPS. We are using that. Not sure if that poses 
additional issues.

Wish there was an easy way to test this out in production. Perhaps a quick ACL 
to block phone agent and desktop agent access to the pub and see what happens. 
And then another test where the ACL blocks access to the LDAP server 
temporarily.

Sent from my iPhone

On Jul 6, 2015, at 10:04 AM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Hi Dan!

Thanks for the clarification/correction  I just happen to have a few 3-node 
cluster hanging around and I just tried this 5 times in a mix of 9.1.1, 10.0 
and 10.5 and here is what I found:

3 times LDAP auth was a seamless failover to the sub
2 times LDAP auth did not work on the sub until I bounced the tomcat service on 
the sub, then it worked fine.

I'm wondering if that, on the times it doesn't work in a failover (because I 
have experienced it a few times) a simple service bounce is all that is needed?

I suppose another cause of LDAP auth failover NOT working (but not always 
intuitive) would be cluster over wan (nodes in the cluster are not all on the 
same segment) and the sub node that LDAP auth is trying to bind from can't talk 
to the AD server.

From: dpa...@fidelus.com<mailto:dpa...@fidelus.com>
To: le...@uoguelph.ca<mailto:le...@uoguelph.ca>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Date: Mon, 6 Jul 2015 13:45:08 +
Subject: Re: [cisco-voip] LDAP Authentication when CUCM publisher is down.
LDAP authentication is used by Tomcat and isn't just restricted to the 
Publisher server - Subscriber nodes handle this as well. DirSync is specific to 
synchronization of LDAP attributes and only runs on the Pub, so synchronization 
would definitely be affected if the Publisher is offline. I suggest to check 
out the Tomcat Security logs off CUCM for more info on user authentication 
against LDAP and your source of failure.

So to answer your question, LDAP authentication should still work when the 
Publisher is offline.

For the UCCX agent concern, authentication of agents occur over AXL to CUCM, so 
if the AXL server is the Publisher, and that's offline or experiencing issue w/ 
Tomcat during an authentication attempt by the UCCX agent, then I would imagine 
seeing a failure. AXL and Tomcat Security logs off the UCM side should shed 
some light on that problem

As for SSO, I checked w/ my teammate and, in his experience, SSO can be handled 
by Subscriber nodes assuming the metadata was imported to those servers - 
authentication occurs against the IdP and not CUCM so this seems logical to me 
as well.

Hope this helps.

- Dan


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Monday, July 06, 2015 9:16 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] LDAP Authentication when CUCM publisher is down.


This has been our experience as well. Glad you started this t

[cisco-voip] LDAP Authentication when CUCM publisher is down.

[Matthew Collins]

Hi All,

CUCM 10.5

Just trying to get some conformation, When LDAP Synchronization and 
authentication is enabled this is performed by the DirSync process that only 
runs on the CUCM Publisher. So If we lose the CUCM Publisher for whatever 
reason it would seem that the Authentication also fails due to the single point 
of failure of DirSync. Should LDAP authentication still work if the CUCM 
Publisher is still down.

So for LDAP users this would stop them signing in to Jabber clients and UCCX 
agents who are ldap’ed synced logging into the finesse webpages. Does anyone 
know is SSO is resilient on the CUCM publisher or would SSO still work in a 
Publisher outage.

Regards

Matthew Collins

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[cisco-voip] Blackberry MVS on CUCM 10.5

[Matthew Collins]

Hi All,

I know Blackberry MVS isn't supported on CUCM 9 and above but has anyone got it 
working on 10.5? A client is upgrading from CUCM 8.6 to 10.5.2 and would like 
to retain the feature as currently there isn't a jabber app for blackberry.

>From a Call manger perspective the MVS client is still available so thinking 
>is should work just unsupported? Think it would just be a case of changing the 
>CUCM IP address on the Bes server.

Regards

Matthew
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Re: [cisco-voip] UCCX Finesse TeamPerformance Gadget height?

[Matthew Collins]

Thanks Kevin for the info. I do wonder who the get to test new products before 
the release them, this seems such a simple request.


Matthew

On 8 May 2015, at 16:47, Kevin Przybylowski 
mailto:kev...@advancedtsg.com>> wrote:

Take a look at the post below in the link, doesn’t look like it’s possible to 
resize that particular box yet – I tested in my lab and the gadget height 
parameter was ignored for me as well:

I was actually working with TAC on a performance issue with some finesse 
clients and they pointed me to this link:
https://communities.cisco.com/message/171139



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: Friday, May 08, 2015 8:35 AM
To: Cisco VoIP Group 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>)
Subject: [cisco-voip] UCCX Finesse TeamPerformance Gadget height?

Hi All,

Anyone know of a way to change the height on the default supervisor Team 
Performance gadget within the UCCX finesse desktop layout. (UCCX v10.6(1))



manageTeam
finesse.container.tabs.supervisor.manageTeamLabel




/desktop/gadgets/TeamPerformance.jsp




 

Currently the gadget can only show about 6 agents without having to scroll down 
within the gadget when the rest of the screen is empty. In most of the other 
gadgets height seems to be a parameter and I have successfully changed these to 
show all the agents without have to scroll down within the gadget.

I have tried to add ?gadgetHeight=450 after the TeamPerformance.jsp but this 
doesn’t seem to have any effect. Any help very much appreciated.


Regards

Matthew Collins
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[cisco-voip] UCCX Finesse TeamPerformance Gadget height?

[Matthew Collins]

Hi All,

Anyone know of a way to change the height on the default supervisor Team 
Performance gadget within the UCCX finesse desktop layout. (UCCX v10.6(1))



manageTeam
finesse.container.tabs.supervisor.manageTeamLabel




/desktop/gadgets/TeamPerformance.jsp




 

Currently the gadget can only show about 6 agents without having to scroll down 
within the gadget when the rest of the screen is empty. In most of the other 
gadgets height seems to be a parameter and I have successfully changed these to 
show all the agents without have to scroll down within the gadget.

I have tried to add ?gadgetHeight=450 after the TeamPerformance.jsp but this 
doesn’t seem to have any effect. Any help very much appreciated.


Regards

Matthew Collins
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Re: [cisco-voip] Media Resource Allocation Bug

[Matthew Collins]

Just hit a bug in UCCX finesse CSCur05888 where the end user stays logged in if 
they close their web browser without signing out first. If user id is all lower 
case works as expected and UCCX signs them out after a time out. If user ID 
contains 1 upper case letter they stay logged in forever.


Regards

Matthew Collins


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Walenta, Philip
Sent: 18 April 2015 02:47
To: Anthony Holloway
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Media Resource Allocation Bug

Now that's almost funny (the bug that is).  I never understand how stuff like 
this slips through QA.

Sent from my iPhone

On Apr 17, 2015, at 8:44 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
I posted that one to the list about one year ago to the day.

https://tools.cisco.com/bugsearch/bug/CSCul53246
On Fri, Apr 17, 2015 at 6:53 PM Heim, Dennis 
mailto:dennis.h...@wwt.com>> wrote:
Wasn’t there a bug back in 8.x/9.x where capitalization impacted how media 
resources were allocated? Does anyone recall the bug id on that?

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
<https://twitter.com/CollabSensei>
[Phone][video]
"Innovation happens on project squared" -- 
http://www.projectsquared.com<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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Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5

[Matthew Collins]

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] 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]<mailto:[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" 
mailto: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

[http://www.cisco.com/c/dam/en/us/td/i/templates/note.gif]
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 cumulative update that incorporates all of the fixes and changes 
from Cisco Unified Communications Manager 10.5(2) along with additional changes 
that are specific to this SU.

Note

You can only install this SU on Cisco Unified Communications Manager Release 
6.1(4x), 6.1(5x), 7.1(3x), 7.1(5x), 8.0(x), 8.5(1x), 8.6(x), 9.0(x), 9.1(x), 
10.0(1), 10.5(1), or 10.5(2) This SU will not install over any 10.5(2)ES’s. 
Upgrades from 6x, 7x, 8.x, and 9.x need to be requested via PUT 
(www.cisco.com/upgrade<http://www.cisco.com/upgrade>) to obtain the necessary 
license. This SU should not be installed on a Business Edition 3000 server.

I would normally always go with what is says in the compatibility matrix, 
however my colleague has just done an upgrade albeit in a lab from 8.5.1 direct 
to 10.5.2 SU1.

Anyone know if he just got lucky or if that is now supported?

Regards

Nick




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Re: [cisco-voip] Extension Mobility and RCC

[Matthew Collins]

Hi Reto,

I came across this feature a few years ago back in CUCM V6 and MOC. This was a 
working feature of lync (Or MOC back then). As the Lycn can’t tell what device 
to control as there are two devices with the end users extension (fixed phone 
and EM profile). To get around this lync places a call to its self. The first 
phone to answer is then selected as the phone Lycn will use until the user 
signs out again.

Matthew



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Reto 
Gassmann
Sent: 28 January 2015 09:57
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Extension Mobility and RCC

Hello Group

we run a UCM / IM&P 9.1 and have RCC set up with Lync.
For a project office we introduced Extension Mobility. Some users now have 
their number shared on a fixed IP Phone in their office and use EM in the 
project office.
When the user is logged on with EM in the project office and they log on to 
Lync the IP Phone in the office dials its own number and plays the busy tone 
for 30 seconds.

Has anybody seen this behaviour or knows why the office phone dials itself?

Any help is appreciated.
Regards Reto
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Re: [cisco-voip] Device pack installation methodology question

[Matthew Collins]

Hi Ryan,

I have just installed the device pack on the publisher first. Then changed the 
device defaults back to their originals, Then reboot the publisher, TFTP and 
subs in turn. No need to mess around with the CM groups.

One tip I got from the forum a couple of years ago when installing device pack 
was to open the device defaults page in IE, Then install the device pack using 
another browser Firefox or chrome, once the device pack is installed go back to 
the IE page without refreshing and select save, This will then revert all the 
firmware version back to the pre device pack state, Saves having to manually 
change each one back.


Regards

Matthew


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: 03 January 2015 20:12
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Device pack installation methodology question

I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support 
for some 88xx phones but I do not want to update the loads for anything else.

The approach I am going to use is:

Drop the publisher out of the CM Group, forcing all phones to the subscriber. 
Install the device pack on the publisher and reboot the publisher. Once the 
publisher is backup, set all the device defaults back to what I want them to be 
then add the publisher back to the CM Group. Then drop the subscriber from the 
CM Group forcing all the phones on the publisher and start the install process 
over for the subscriber. Once everything is back up add the subscriber back to 
the CM Group.

Does that sound reasonable or is there an easier way?

Thanks,

Ryan
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[cisco-voip] UCCX 10.5 Finesse web agent fail over

[Matthew Collins]

Hi All,

Has anyone dealt with UCCX 10.5 Finesse agent fail over. If I stop the service 
via the UCCX Admin GUI the web session reconnects to its backup server, But if 
we lose complete connectivity to the active server the Finesse session isn't 
failed over automatically.

Apart from giving agents both server URL's in the event of a failover is there 
any other method. I have searched the manuals but can't find anything other


Thanks in advance

Matthew
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Re: [cisco-voip] Jabber persistent chat history

[Matthew Collins]

That’s not very scalable though as a room would need to be set up prior and 
then for every separate user you wanted this for or have I got that wrong.

I know there was a feature on Lync called IM forking that would provide that 
exact feature. I did look into this a while ago and Cisco do provide IM Forking 
but only until the user reply’s.

IM Forking
When a user sends an IM to a contact who is signed in to multiple IM clients. 
IM and Presence Service delivers the IM to each client. This functionality is 
called IM forking. IM and Presence Service continues to fork IMs to each 
client, until the contact replies. Once the contact replies, IM and Presence 
Service only delivers IMs to the client on which the contact replied.
Matt

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Josh 
Warcop
Sent: 19 November 2014 01:07
To: Pavan K; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber persistent chat history

That is exactly what persistent chat provides.

Sent from my Windows Phone

From: Pavan K
Sent: ‎11/‎18/‎2014 6:49 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber persistent chat history

Is there a way to do persistent chat across jabber clients

So let's say I am in a jabber conversation on my laptop, can I resume the 
conversation on my jabber mobile and have my conversation history from my 
laptop follow me to my mobile.

Basically like a persistent group chat without a chat room.

Pavan
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Re: [cisco-voip] vSwitch in Active/Standby

[Matthew Collins]

Hi Jason,

In active standby mode both ports should be up/up on the switch.

Regards

Matthew

ǀ BLOCK ǀ
Cisco UK&I Solution Innovation Partner of the Year FY15

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: 19 September 2014 19:26
To: cisco-voip (cisco-voip@puck.nether.net)
Subject: [cisco-voip] vSwitch in Active/Standby

I can’t recall and don’t have access to this right now to ESXi 5 to check.

If I configure in ESXi a vSwitch2 as Active/Standby, then the second NIC in 
Standby on the 3750 Switchport would show down/down right?  Or will it show 
up/up and ESXi will “activate” it as needed.

Discussing if the Standby 3750 Switchport has to go Up and run thru Spanning 
Tree (if portfast is not configured) during Active/Standby failover.


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Re: [cisco-voip] UCCX Wallboard Query

[Matthew Collins]

Hi Neal, Brian,

If I change the command to Const SQL_Command = "select * from rtcsqssummary” it 
looks like it is only pulling the Data from the first CSQ.



Regards

Matthew

ǀ BLOCK ǀ
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From: Haas, Neal [mailto:nh...@co.fresno.ca.us]
Sent: 20 August 2014 15:12
To: Matthew Collins; cisco-voip@puck.nether.net
Subject: RE: UCCX Wallboard Query

Does Const SQL_Command = "select * from rtcsqssummary” Work?


Neal Haas

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]<mailto:[mailto:cisco-voip-boun...@puck.nether.net]>
 On Behalf Of Matthew Collins
Sent: Wednesday, August 20, 2014 6:57 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] UCCX Wallboard Query

Hi All,

I’ve got a wallboard running but its only reporting on a specific queue per 
website, I can change the CSQ its reports on by changing name of the CSQ in the 
below line of code.

Const SQL_Command = "select * from rtcsqssummary where csqname like 
'CSQ_IT_Support’"

I’m trying to change the line to report on all CSQ collectively but everything 
I try breaks it.


Thanks in advance
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[cisco-voip] UCCX Wallboard Query

[Matthew Collins]

Hi All,

I’ve got a wallboard running but its only reporting on a specific queue per 
website, I can change the CSQ its reports on by changing name of the CSQ in the 
below line of code.

Const SQL_Command = "select * from rtcsqssummary where csqname like 
'CSQ_IT_Support’"

I’m trying to change the line to report on all CSQ collectively but everything 
I try breaks it.

Thanks in advance
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Re: [cisco-voip] Cisco MWI on IPC Unigy dealer boards

[Matthew Collins]

Hi All,

Just to let you know that this is now working using CUC >SIP> CUCM >SIP> >IPC 
Unigy, The Unigy system was rebooted and everything stated working as expected.

Regards

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: 19 July 2014 12:59
To: Ryan Ratliff (rratliff)
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Cisco MWI on IPC Unigy dealer boards

HI Ryan,

Thanks for you suggestion, I have now changed the integration between the CUCM 
and CUC to SIP but still the MWI is not working. You mention the CUCM should be 
able to direct the unsolicited Notify message to directly to Unigy.

Is there anything specific config I would need to configure to enable this. I 
cant seem to find any documentation (except for CUC express) or other posts in 
the forums. I have checked the service parameters and enterprise parameter of 
both the CUCM and CUC server but again cant seem to find anything.


From: Ryan Ratliff (rratliff) mailto:rratl...@cisco.com>>
To: Matthew Collins 
mailto:matthewjcoll...@btinternet.com>>
Cc: "cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Sent: Thursday, 17 July 2014, 14:35
Subject: Re: [cisco-voip] Cisco MWI on IPC Unigy dealer boards

You may need qsig tunneling over the sip trunk to get MWI working with the sccp 
unity integration.  I bet you will have better luck doing sip to unity so UCM 
can direct the unsolicited notify out to unigy directly.

Sent from my iPhone

On Jul 17, 2014, at 6:00 AM, "Matthew Collins" 
mailto:matthewjcoll...@btopenworld.com>> wrote:
Hi All,

Hope one of you will be able to help or point me in the right direction.

CUCM and CUC are both 9.1.1,

CUCM to CUC is SCCP

CUCM to IPC Unigy is SIP

Dial plan on CUCM is E164

Dial Plan on Unigy is 4 digits

CUC account has the Unigy 4 digit extension set as a alternate MWI

I have created a specific 4 digit route pattern pointing to the IPC unigy sip 
trunk

If I dial the MWI on and off numbers from the dealer board I can see them hit 
the CUCM but get NU played back from the dealer board

Sip trunk from IPC Unigy has the CUC partition in its CSS

The Unity ports CSS have access to the 4 digit route pattern pointing to the 
IPC unigy.

Thanks

Matthew
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[cisco-voip] Unity Caller Input audit

[Matthew Collins]

Hi All,

I'm trying to audit all of the caller input settings on Cisco Unity Connection 
8.6.2.2-76 local end user mail boxes. These details are not collected on 
the bulk export feature. Any thoughts?

Regards

Matthew
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Re: [cisco-voip] Cisco MWI on IPC Unigy dealer boards

[Matthew Collins]

HI Ryan,
 
Thanks for you suggestion, I have now changed the integration between the CUCM 
and CUC to SIP but still the MWI is not working. You mention the CUCM should be 
able to direct the unsolicited Notify message to directly to Unigy. 
 
Is there anything specific config I would need to configure to enable this. I 
cant seem to find any documentation (except for CUC express) or other posts in 
the forums. I have checked the service parameters and enterprise parameter of 
both the CUCM and CUC server but again cant seem to find anything.
 
  


 From: Ryan Ratliff (rratliff) 
To: Matthew Collins  
Cc: "cisco-voip@puck.nether.net"  
Sent: Thursday, 17 July 2014, 14:35
Subject: Re: [cisco-voip] Cisco MWI on IPC Unigy dealer boards
  


You may need qsig tunneling over the sip trunk to get MWI working with the sccp 
unity integration.  I bet you will have better luck doing sip to unity so UCM 
can direct the unsolicited notify out to unigy directly. 

Sent from my iPhone 

On Jul 17, 2014, at 6:00 AM, "Matthew Collins" 
 wrote:

 
Hi All, 
>
>Hope one of you will be able to help or point me in the right direction. 
>
>CUCM and CUC are both 9.1.1, 
>
>CUCM to CUC is SCCP 
>
>CUCM to IPC Unigy is SIP 
>  
>Dial plan on CUCM is E164 
>
>Dial Plan on Unigy is 4 digits 
>
>CUC account has the Unigy 4 digit extension set as a alternate MWI 
>
>I have created a specific 4 digit route pattern pointing to the IPC unigy sip 
>trunk 
>
>If I dial the MWI on and off numbers from the dealer board I can see them hit 
>the CUCM but get NU played back from the dealer board 
>
>Sip trunk from IPC Unigy has the CUC partition in its CSS 
>
>The Unity ports CSS have access to the 4 digit route pattern pointing to the 
>IPC unigy. 
>
>Thanks 
>
>Matthew   
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[cisco-voip] Cisco MWI on IPC Unigy dealer boards

[Matthew Collins]

Hi All,

Hope one of you will be able to help or point me in the right direction.

CUCM and CUC are both 9.1.1,

CUCM to CUC is SCCP

CUCM to IPC Unigy is SIP
 
Dial plan on CUCM is E164

Dial Plan on Unigy is 4 digits

CUC account has the Unigy 4 digit extension set as a alternate MWI

I have created a specific 4 digit route pattern pointing to the IPC unigy sip 
trunk

If I dial the MWI on and off numbers from the dealer board I can see them hit 
the CUCM but get NU played back from the dealer board

Sip trunk from IPC Unigy has the CUC partition in its CSS

The Unity ports CSS have access to the 4 digit route pattern pointing to the 
IPC unigy.

Thanks

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[cisco-voip] UCCX If conditional statement

[Matthew Collins]

Hi All,

I'm working on a UCCX scrip at the moment and have come across an If 
conditional statement and I just can't get my head around it. Think it's just a 
time check and could replace with a time of day check but would like to 
understand it before I change anything.

Anyone out there able to help me get my head around it?

((D[now].dow != 1 && D[now].dow != 7) && (T[now].getHours() > 7 && 
T[now].getHours() < 18)) || (D[now].dow == 7 && (T[now].getHours() > 8 && 
T[now].getHours() < 12))


Regards
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[cisco-voip] Reading Cups Instant messages after being stored for compliance.

[Matthew Collins]

Hi All,

CUPS or IM&P setup with external database

How do people search through the conversations stored in the external database, 
Is there a handy front end that I'm missing or is it done just buy using SQL 
quires

Failing can anyone recommend one of the 3rd party software prodivers

Thanks in advance

Matt
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Re: [cisco-voip] NTP Issue

[Matthew Collins]

Hi Costas,

I take it you are using 89XX phones?

CSCun11142 is a bug, affecting 89XX phones daylight settings.

Usually daylight saving is 4th Sunday of march, But this year the last Sunday 
is the 5th Sunday.

Regards

Matthew


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
costas georgiou
Sent: 24 March 2014 11:25
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] NTP Issue

Hi All,

I have a CUCM running 8.6.2, I have been informed this morning that some IP 
phones are 1 hr out of sync. This is happening to a few remote sites.  The NTP 
server is seperate and is looked after by the customer.

I have check the CUCM server and get the follwing results which look ok:

admin:utils ntp status
ntpd (pid 26786) is running...

 remote   refid  st t when poll reach   delay   offset  jitter
==
+10.160.x.x.10.107.195.1 4 u  572 1024  377   16.7430.105   0.015
*165.72.x.x.GPS.1 u  659 1024  377   35.434   -0.071   7.146


synchronised to NTP server (165.72.x.x) at stratum 2
   time correct to within 32 ms
   polling server every 1024 s

Current time in UTC is : Mon Mar 24 09:36:02 UTC 2014
Current time in Europe/London is : Mon Mar 24 09:36:02 GMT 2014
Admin:

The clock on the routers sjow the same time.  Any ideas what could be the 
problem?

Thanks in advance for your help.

Regards

Costas



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Re: [cisco-voip] WebEx Meetings Server 2.0 OVA

[Matthew Collins]

Hi all,

Just a pointer after deploying a fresh install. Even though its not documented 
you have to make sure ICMP in enabled between the reverse Proxy and the Admin 
server otherwise the final system check fails.

Regards

Matthew

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of John 
Parduhn
Sent: 20 March 2014 00:05
To: Tim Smith
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] WebEx Meetings Server 2.0 OVA


Sweet!  Thanks, Tim.

John
On Mar 19, 2014 3:53 PM, "Tim Smith" 
mailto:tim.sm...@enject.com.au>> wrote:
Thanks John,

It only looks like one bug, which wouldn't affect us.

CSCum96822

3

Grow fails to update expanded system to same version as source system


Cheers,

Tim

From: John Parduhn [mailto:pard...@gmail.com]
Sent: Tuesday, 18 March 2014 10:40 AM
To: Tim Smith
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] WebEx Meetings Server 2.0 OVA

The upgrade took about six hours start to finish.  Thankfully I had an early 
start to my maintenance window!  It easily could have been exhausting if I had 
to start at 11pm.  I had HA set up, six servers total.  If you know or know 
someone who understands ESX command line, I suggest you involve them.  I was 
really happy to see the new server spin up as smoothly as it did.

Any idea what MR1 is supposed to address?  We found a defect in Productivity 
Tools that started in 1.5 that carried over.  If you change your password due 
to domain security policies, the client sends too many authentication requests 
resulting in account lockouts.

On Monday, March 17, 2014, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:
Hi John,

Thanks for the update.
I put this one on hold after the issues with the media, and the bug with the 
auto upgrade.
There is also an MR1 due shortly, that I probably want to get put on at the 
same time.

How many machines, how long did the bulk of the process take you?

Cheers,

Tim.

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of John Parduhn
Sent: Tuesday, 18 March 2014 1:31 AM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] WebEx Meetings Server 2.0 OVA



Following up with this thread, our upgrade from 1.5 to 2.0 went well.  We went 
with the manual deployment, but copied the disk4 VMDK files from the old to 
upgraded systems using the CLI in ESXi instead of through vCenter.  Overall, 
the documentation is accurate.  We had an issue with the license file 
migration, but a call into TAC remedied it.  Good luck to anyone else out there 
performing the update in the near future.



John



On Mon, Mar 3, 2014 at 7:31 AM, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:

Think I can see where things have gone wrong with OVA vs USB img.

I ordered both e-delivery and physical media just to be safe, and to give the 
guys on site a physical copy of media for DR purposes.



E-delivery order gave me the USB IMG file (it should have been the OVA)

Physical order came via E-delivery as well - it gives me opportunity to 
download USB IMG file.



Need to see if they have also shipped the physical USB key for me!



Either way as previously noted, it looks like these first versions of OVA have 
problems with Automatic installs. Makes me want to wait for fixed version.



Cheers,



Tim





From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Tim 
Smith
Sent: Friday, 28 February 2014 11:56 PM
To: Eric Pedersen

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] WebEx Meetings Server 2.0 OVA


--
John Parduhn
Huron Consulting Group
550 W. Van Buren | Chicago, Illinois 60607
312-880-2648 single-number reach
jpard...@huronconsultinggroup.com
www.huronconsultinggroup.com

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Re: [cisco-voip] Disabling placed call history on 7841's

[Matthew Collins]

Ok,

Thanks for the pointer Steven, I have now kind of got this working on the Idle 
URL, but after the phone request the idle URL the phone is stuck in the 
following state until I go off hook. The Cancel softkey button doesn’t seem to 
do anything

[cid:image002.png@01CF4375.4DDDCD50]

It has requested the Init:CallHistory as the call history is cleared.



Regards

Matthew


From: Stephen Welsh [mailto:stephen.we...@unifiedfx.com]
Sent: 19 March 2014 10:40
To: Matthew Collins
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Disabling placed call history on 7841's

Hi Matt,

Just tested on our 7841 (sip78xx.10-1-1-9) by sending Init:CallHistory using 
PhoneView and it worked fine.

Error 4 is usually an authentication Error, if you are pushing (HTTP POST) the 
CiscoIPPhoneExecute object to the phone I suspect their is something wrong with 
the Authentication URL or an ITL issue?

The CiscoIPPhoneError section here details the error codes:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/all_models/xsi/9_1_1/CUIP_BK_P82B3B16_00_phones-services-application-development-notes/CUIP_BK_P82B3B16_00_phones-services-application-development-notes_chapter_011.html#CUIP_RF_C22B2EF6_00

Assuming you will be hosting that XML file on a web server and pointing the 
Idle URL to it I would expect it to work fine as authentication does not apply 
to that scenario.

Kind Regards.

Stephen Welsh

[cid:image003.png@01CF4375.4DDDCD50]


On 19 Mar 2014, at 10:20, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:


Hi All,

I have a requirement to disable the call history on a selected amount of 7841 
handsets.

To start with the call history has now been moved under the services button and 
not the directory’s button.

I know I can disable the logged calls under the line settings and this stops 
all received and missed calls appearing in the call history but the placed call 
still appear..

I know I can put a dummy URL in the service URL on the specific phone to stop 
user getting to the call history on the phone but the user can still access the 
call history using the up button. This will also disable all other services.

I did have a XML file that worked on the 79XX series phone that cleared down 
the call history, I used to out the URL in the idle timeout and the phone would 
clear down the call history whenever the idle timers expired





But this doesn’t seem to work on the newer sip models. As I get a “xml error 4 
parse error” returned. When I browse to the webpage I get the following 
returned so think the XML is right (for a 79XX phone)


<http://10.1.13.12/clear_phone.xml>




Regards

Matthew Collins
Collaboration Principal Consultant

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Cisco Global EMEAR Partner of the Year 2013
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[cisco-voip] Disabling placed call history on 7841's

[Matthew Collins]

Hi All,

I have a requirement to disable the call history on a selected amount of 7841 
handsets.

To start with the call history has now been moved under the services button and 
not the directory’s button.

I know I can disable the logged calls under the line settings and this stops 
all received and missed calls appearing in the call history but the placed call 
still appear..

I know I can put a dummy URL in the service URL on the specific phone to stop 
user getting to the call history on the phone but the user can still access the 
call history using the up button. This will also disable all other services.

I did have a XML file that worked on the 79XX series phone that cleared down 
the call history, I used to out the URL in the idle timeout and the phone would 
clear down the call history whenever the idle timers expired





But this doesn’t seem to work on the newer sip models. As I get a “xml error 4 
parse error” returned. When I browse to the webpage I get the following 
returned so think the XML is right (for a 79XX phone)


<http://10.1.13.12/clear_phone.xml>




Regards

Matthew Collins
Collaboration Principal Consultant

ǀ BLOCK ǀ
Cisco Global EMEAR Partner of the Year 2013
Visit our dedicated Healthcare site at 
www.block.co.uk/health<http://www.block.co.uk/health>

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Re: [cisco-voip] CUCM 10 Phones

[Matthew Collins]

Hi Leslie,

Below is a screen shot from the insert 79XX phone page from 10.0.(1)

[cid:image001.jpg@01CF3918.E5FFD220]

Regards

Matthew

ǀ BLOCK ǀ
Cisco Global EMEAR Partner of the Year 2013

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Buchanan
Sent: 06 March 2014 04:45
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] CUCM 10 Phones

The SRND specifies the 7900 series phones are still considered collaboration 
endpoints but doesn't indicate which models are allowed.

On Thu, Mar 6, 2014 at 6:28 AM, Leslie Meade 
mailto:leslie.me...@lvs1.com>> wrote:
I know that the 1st gen phones are end of life, but they should still connect 
to a callmanager 10 environment ?
I have a client that wants to upgrade the system, but at this time not willing 
to upgrade the handsets.

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Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number

[Matthew Collins]

Hi Abbas,

Bit long winded but yes it can be done, Create another loopback address on the 
gateway, Set up a h323 router in CUCM using the new loopback address and send 
that specific dial in number down to the H323 router. The TLC scrit can then 
pick up the call and route accordingly.  You will also need to create a dial 
peer to send the public call back to the CUCM to then be routed out MGCP.

Regards

Matthew

ǀ BLOCK ǀ
Cisco Global EMEAR Partner of the Year 2013

From: abbas Wali [mailto:abba...@gmail.com]
Sent: 11 February 2014 11:01
To: Matthew Collins
Cc: Buchanan, James; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a 
new number

TCL scripts on MGCP gateway ??

On 11 February 2014 10:40, Matthew Collins 
mailto:mcoll...@block.co.uk>> wrote:
This can also be set up as part of Single number reach using Mobile Voice 
Access, You need to install a TCL script on the router. Users dial in, Prompted 
for extension number and pin then get secondary dial tone to dial out.

Regards

Matthew

-Original Message-
From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of Buchanan, James
Sent: 11 February 2014 10:21
To: abbas Wali; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a 
new number

Not really. Some have done this using call handlers with pre-programmed speed 
dials. Some have also tried changing the restriction tables in Unity to allow 
for this, but that has a tone of security issues. UCCX would be your best place 
to do this, but you would want to make sure you secured it with a PIN.

James Buchanan | Sr. Network Engineer
Presidio | www.presidio.com<http://www.presidio.com>
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | 
jbucha...@presidio.com<mailto:jbucha...@presidio.com>



PRESIDIO
Practical thinking for a connected world.


Follow Us: www.twitter.com/presidio<http://www.twitter.com/presidio>





From: cisco-voip 
[cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>] 
on behalf of abbas Wali [abba...@gmail.com<mailto:abba...@gmail.com>]
Sent: Tuesday, February 11, 2014 4:11 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] dial into ddi and get a secondary dial tone to call a 
new number

hi all,

is there a feature in CUCM 8.5 where you call into a tollfree number into cucm 
cluster and then from there you call out to different destinations i.e. getting 
a secondary dialtone after the first call connects

Thanks


--

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Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number

[Matthew Collins]

This can also be set up as part of Single number reach using Mobile Voice 
Access, You need to install a TCL script on the router. Users dial in, Prompted 
for extension number and pin then get secondary dial tone to dial out.  

Regards

Matthew

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Buchanan, James
Sent: 11 February 2014 10:21
To: abbas Wali; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a 
new number

Not really. Some have done this using call handlers with pre-programmed speed 
dials. Some have also tried changing the restriction tables in Unity to allow 
for this, but that has a tone of security issues. UCCX would be your best place 
to do this, but you would want to make sure you secured it with a PIN.

James Buchanan | Sr. Network Engineer
Presidio | www.presidio.com
12 Cadillac Drive Suite 130, Brentwood, TN 37027
D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbucha...@presidio.com



PRESIDIO
Practical thinking for a connected world.


Follow Us: www.twitter.com/presidio





From: cisco-voip [cisco-voip-boun...@puck.nether.net] on behalf of abbas Wali 
[abba...@gmail.com]
Sent: Tuesday, February 11, 2014 4:11 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] dial into ddi and get a secondary dial tone to call a 
new number

hi all,

is there a feature in CUCM 8.5 where you call into a tollfree number into cucm 
cluster and then from there you call out to different destinations i.e. getting 
a secondary dialtone after the first call connects

Thanks


--

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.

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