Re: [cisco-voip] Lights for Calls in Queue CCX

[Haas, Neal via cisco-voip]

Has anyone developed a Network attached electric zap…….

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ben 
Amick
Sent: Friday, February 2, 2018 7:39 AM
To: Matthew Loraditch ; Brian Meade 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Lights for Calls in Queue CCX

Just give everyone a noise cancelling headset and make it an audible siren 
instead of a flasher ☺

Ben Amick
Unified Communications Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: Friday, February 2, 2018 10:38 AM
To: Brian Meade >
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Lights for Calls in Queue CCX

Yeah that would do it. I found an example on the cisco forums of a scripting 
setup that should work.

Looking forward to the quote reaction. You need 20 of these (they want one on 
each desk!!) and a drop for each plus a switch because of the number of new 
drops. Because ringing, finesse, and jabber flashing aren’t enough.



Matthew Loraditch

Sr. Network Engineer


p: 443.541.1518



w: 
www.heliontechnologies.com

 |

e: mloradi...@heliontechnologies.com


[cid:image001.png@01D39BFC.616CF1F0]


[Facebook]


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 From: bmead...@gmail.com 
[mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: Friday, February 2, 2018 10:19 AM
To: Matthew Loraditch 
>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Lights for Calls in Queue CCX

Like a SIP Strobe?
http://www.cyberdata.net/voip/011376/

Just use a Place Call step in the script when a call hits the queue.

Might require some external scripting if you need it to keep flashing until the 
queue is empty.

On Fri, Feb 2, 2018 at 9:08 AM, Matthew Loraditch 

Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

[Haas, Neal]

We are creating lighter backgrounds to resolve the not active line color. Also 
I just noticed no Text on the top of the screen!

Thank You,

Neal Haas
NSE, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890

From: Bill Talley [mailto:btal...@gmail.com]
Sent: Thursday, December 14, 2017 7:51 AM
To: JASON BURWELL <jason.burw...@foundersfcu.com>
Cc: Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

Thanks for your feedback Jason.  I'll try a more solid background and will 
report back to the group with results.

On Thu, Dec 14, 2017 at 9:47 AM, JASON BURWELL 
<jason.burw...@foundersfcu.com<mailto:jason.burw...@foundersfcu.com>> wrote:
Yes, dark wallpapers bring out plain white text for us. It’s been like this 
since upgrading off of I think FW11.5x or whatever version created an entire 
white block for each label which pretty much killed the wallpapers. We are 
using wallpaper that is mostly solid but a watermark of the company logo does 
appear under the text and it looks fine. None of our wallpapers use gradients 
though. Maybe that’s the issue, it having a problem interpreting and deciding 
what color text to use? This is on 8851 and 8865.

Jason

From: Bill Talley [mailto:btal...@gmail.com<mailto:btal...@gmail.com>]
Sent: Thursday, December 14, 2017 10:22 AM

To: JASON BURWELL 
<jason.burw...@foundersfcu.com<mailto:jason.burw...@foundersfcu.com>>
Cc: Haas, Neal <nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>>; NateCCIE 
<natec...@gmail.com<mailto:natec...@gmail.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

So your dark backgrounds on 12.0.1SR1 have white text?   Are they solid 
backgrounds or gradients?  We had white text over the same background on 
firmware 11.7.1.17 but after upgrading to 12.0.1SR1 the text changed to gray.   
Downgrading to 11.7.1.17 reverted text to white.  Using solely 8845s and CCM 
11.5.1.SU3 on our end.
Sent from a mobile device with very tiny touchscreen input keys. Please excude 
my typtos.

On Dec 14, 2017, at 9:06 AM, JASON BURWELL 
<jason.burw...@foundersfcu.com<mailto:jason.burw...@foundersfcu.com>> wrote:
Not sure, I saw your picture and I have not had that issue anywhere on the 12x 
train. We have light and dark wallpapers. The Text for Speed Dials/Features 
changes automatically between Black and White color text depending on the 
wallpaper chosen so the text always clear and visible. I do recall an issue 
with text being seen back in 11.5 or 11.7 somewhere. I’ve been through a lot of 
ES releases prior to 12 so hard to remember.


Jason

From: Bill Talley [mailto:btal...@gmail.com]
Sent: Thursday, December 14, 2017 9:07 AM
To: JASON BURWELL 
<jason.burw...@foundersfcu.com<mailto:jason.burw...@foundersfcu.com>>
Cc: Haas, Neal <nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>>; NateCCIE 
<natec...@gmail.com<mailto:natec...@gmail.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

Issue with display text still exists on SR1.  That’s what we were using.
Sent from a mobile device with very tiny touchscreen input keys. Please excude 
my typtos.

On Dec 14, 2017, at 7:24 AM, JASON BURWELL 
<jason.burw...@foundersfcu.com<mailto:jason.burw...@foundersfcu.com>> wrote:
I recommend loading the latest available firmware for the 8800 phones. Right 
now 12-0-1SR1-1 is available. In some cases I have had to request the latest ES 
form TAC. The 11.7x firmware has been some of the buggiest phone firmware I 
have seen for Cisco phones in the last 15 years. 12x seems to be much better 
than 11.7 but I am still getting issues with phones freezing up occasionally 
and other weird problems. Waiting to see if 12-0-1SR1-1 finally completely 
resolves those problems.

Jason

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Haas, 
Neal
Sent: Thursday, December 14, 2017 2:01 AM
To: Bill Talley <btal...@gmail.com<mailto:btal...@gmail.com>>; NateCCIE 
<natec...@gmail.com<mailto:natec...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color


Thats exactly what I am seeing. When you change the background to a lighter 
coloring, you can read everything. I will need to put my glasses on to see if 
it is blurry..



Fine quality control again by Cisco. I have already started the complaint train 
with TAC, tomorrow I will start with the account rep.


Neal

From: Bill Talley <btal...@gmail.com<mailto:btal...

Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

[Haas, Neal]

I want to do that but we are hitting the answer/hold bug.


I think my solution is to create a background image that doesn't look as bad as 
the default cisco one.


any other issues that you have seen with 12.0(1)?


Neal Haas
ITSD Analyst - Communications
559-600-5890

From: Bill Talley <btal...@gmail.com>
Sent: Wednesday, December 13, 2017 7:12:32 PM
To: Haas, Neal
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

We have the same issue and have been unable to find a fix short of reverting to 
11.7 firmware.

Sent from a mobile device with very tiny touchscreen input keys. Please excude 
my typtos.

> On Dec 13, 2017, at 8:50 PM, Haas, Neal <nh...@co.fresno.ca.us> wrote:
>
> I wanted to push the new 12.0(1) firmware to our 8851/8861 phones. I did a 
> test and you cannot even see the line text on dark backgrounds, which is 90% 
> of the images . Anyone know how to fix line text color? The only image that 
> really works is the one that looks like someone socked the phone.
>
> Neal
>
> 
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Re: [cisco-voip] Voicemail history

[Haas, Neal]

1. What does your company want for a retention period for deleted VM. We 
started at 2 weeks, now I think we are 6 weeks.

2. for regulatory or legal requirements (state or local laws) what are those, 
we post on an MOU we only keep them for a specific period of time. If we have a 
request for anything outside of the MOU, we say we dont keep backups for that 
long.

3. do you use the integration to outlook? VM files will be kept under the 
outlook retention time also..

4. we have 6,000 VM boxes, our database size is 20 gig total, we set the VM box 
size to be small.


Neal Haas


From: cisco-voip  on behalf of Parker 
Pearson - Donoma 
Sent: Wednesday, September 6, 2017 8:13 PM
To: Ben Amick; Cisco VOIP
Subject: Re: [cisco-voip] Voicemail history

It really depends on whether you have regulatory or legal requirements to 
retain your voicemails.  Voicemails are an electronic record that can be 
requested for audits or legal discovery.

I understand the desire to groom out the messages in 21-30 days to keep the 
database humming along - but it has to be balanced with the possible need for 
retention & compliance.  A conversation across functions is probably a really 
good idea to find out what your stakeholders need too.

My understanding is that Unity Connection has a limit for VM storage - I think 
it's 2 TB. Depending on the size of your organization and how many voicemails 
your users are averaging in their mailboxes - that limit can get tight.   An 
easy option to satisfy both IT management & retention compliance is a solution 
like Message Vault.  Retains records as needed, enables a very short retention 
length that keeps your UCN database tidy.

Hope that helps.


[Donoma Software Web Page]

Parker Pearson
Vice President, Marketing & Business Development

Donoma Software
1750 Kraft Dr. Suite 1200 Blacksburg, VA 24060
540.443.3577
par...@donomasoftware.com
www.donomasoftware.com
[Facebook] [Twitter] 
  [LinkedIn] 



From: cisco-voip  on behalf of Ben Amick 

Date: Wednesday, September 6, 2017 at 10:18 PM
To: Cisco VOIP 
Subject: [cisco-voip] Voicemail history

So I was just wanting to reach out and see what everybody thinks is a 
reasonable unity message aging scheme? What point does it get to a point where 
you start having storage issues?
Ben Amick
Telecom Support Analyst

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Re: [cisco-voip] call queue light (BLF) for UCCX calls in queue

[Haas, Neal]

That would be cool never thought of that.

We usually do one of two things. 1. Send email if no one is logged into the 
queue, 2. Phone log in, this will show you the actual number of calls in the 
queue.

Thank You,

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Conyers
Sent: Wednesday, August 30, 2017 12:07 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] call queue light (BLF) for UCCX calls in queue

Does anyone have any experience with scripting a "calls in queue" light 
(possibly a BLF indicator) for Contact Center Express? Our old legacy system 
can easily light a lamp on a phone when a call is in queue and also have the 
light flutter when multiple calls are in queue. I have yet to be able to script 
this in CCX script editor. We have used Informacast paging to send a paging 
message to certain phones but it is a third system to integrate into a call 
center. Ideally I would like to only integrate CUCM and CCX. I assume an 
"outdialContact" would work on CCX but was wondering if there was a way to do 
it using "Get Reporting Statistic" parameter that might be easier.
I came across this post on Cisco support, but it doesn't fully explain it, so I 
am stuck on exactly how to do this. Below is a sample of my script that uses 
Informacast. Is there a better way to accomplish this without using the 
Informacast system, and also a way to make the lamp flutter like it used to in 
the old system?
[cid:image001.jpg@01D32189.44B3DA30]
Thoughts?
Thanks,
James

James Conyers
Telecommunications Engineer II
303-871-7992

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Re: [cisco-voip] UCCX 11.6 pre-release communication and webinar

[Haas, Neal]

We try to block Java as a whole on our network, to many security holes. When 
will Cisco Drop Java and move to HTML 5?

Thank You,

Neal
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Charles Goldsmith
Sent: Thursday, August 17, 2017 6:52 AM
To: Abhiram Kramadhati (akramadh) 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX 11.6 pre-release communication and webinar

Abhiram, I understand why it was moved, but having a plugin is not a good 
resolution to the issue, and IMHO, not a good use of development time.  I 
understand it was probably easier than converting it into something else, like 
HTML 5, but that's the right direction.

Ben stated it much better than I can, why a plugin is not a good direction.

Thanks!!


On Thu, Aug 17, 2017 at 8:24 AM, Abhiram Kramadhati (akramadh) 
> wrote:
You are right! The newer updates to the browser blocking Java applets was the 
reason this step was taken:
https://support.mozilla.org/t5/Problems-with-add-ons-plugins-or/Why-do-Java-Silverlight-Adobe-Acrobat-and-other-plugins-no/ta-p/31069

So, the only change done is to make the RTR applet downloadable on a separate 
browser instance – and that way the applet does not try to load in the browser 
and then get blocked. It was an applet before too, but used to load in the 
browser as a plugin. Now, we download it and run it as a Java app on the PC.

Regards,
Abhiram Kramadhati
Technical Solutions Manager, CCBU
CCIE Collaboration # 40065


From: Charles Goldsmith >
Date: Thursday, 17 August 2017 at 10:31 PM
To: "Abhiram Kramadhati (akramadh)" 
>
Cc: "cisco-voip@puck.nether.net" 
>
Subject: Re: [cisco-voip] UCCX 11.6 pre-release communication and webinar

What's the reasoning for moving the Real Time Reporting Tool to a java applet?  
Isn't that a step back in the industry when we see so many things moving to 
HTML 5?  No one likes java, except maybe a few programmers...

On Wed, Aug 16, 2017 at 12:19 AM, Abhiram Kramadhati (akramadh) 
> wrote:
Important pre-release communication: 
https://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express/211582-Tech-Note-on-UCCX-11-6-Pre-Release-Commu.html

Webinar for APJC: UCCX 11.6 introduction and upgrade best practices

Registration link: 
http://cisco.cvent.com/events/webinar-uccx-11-6-preview-and-looking-ahead/event-summary-30265d15f92346b5a314b8b22be1e262.aspx

22nd August. 3:00PM - 4:30PM, Sydney time
Agenda:

  *   Introduction to UCCX 11.6
  *   11.6 demo
  *   Upgrade best practices and planning for 11.6 migration

Presented by Product Management and the Customer Success team.

Regards,
Abhiram Kramadhati
Technical Solutions Manager, CCBU
CCIE Collaboration # 40065


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Re: [cisco-voip] Configure 3845 router for 911 calls.

[Haas, Neal]

Everything is done in the CUCM.

Pots are configured with MGCP.

Create New Partition. (site Name)
Assign 911 Route for site A to Site Partition 1
Assign 911 Route for Site B to Site Partition 1
Assign all other Routes to both Partitions.

Only 911 calls are treated different per site - if you do not take phones 
between sites, you won't have a problem. Two POTS lines per site is correct.

I don't think I missed anything, it's not hard to accomplish the separate 911 
calls between PSAPS.

Neal

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Thursday, July 27, 2017 2:40 PM
To: Norton, Mike ;
Subject: Re: [cisco-voip] Configure 3845 router for 911 calls.


Let me rephrase the questions with more understanding update after I talk to 
Mike and few other people.



We have 2 locations. one is main building were everything reside and the second 
 building is remote building shared with someone else.

About 50 people reside at  the secondary remote location. This people use main 
locations CUCM over the WAN for calling. Remote location is about 25 minutes 
from main building.



So if these people at remote location call 911, emergency responder will see 
main building location not remote location where the call is residing.

Therefore we are tasked to put together plan to give users option to be able to 
call 911 and show their true locations.



We are forced to find spare 3845 router to configure for this users for 911 
call. below is what we have.

ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules for POT Lines.

Amphenol Cable to connect to FOX and 110 block.

CUCM version: 11.0



The reason we ordered 2 POT lines instead of circuit is because they are cheap. 
Circuit cost about $500 a month while POT line cost about $40 a line. Since 
this router is only used for 911 call we don't want to spend that much money.



The question, has any one configured router for 911 and have basic config that 
I can use for my config. Since I haven't done voice gateway configuration 
before, I just need help with this.



Any help is greatly appreciated.



Thanks

Hamu


From: Norton, Mike >
Sent: Monday, July 24, 2017 3:58 PM
To: Hamu Ebiso
Subject: RE: Configure 3845 router for 911 call.


For the IP phones to be usable during WAN outage, you need something they can 
register to. E.g. SRST or CME on the router, or else a CUCM server on-site.



The config you posted shows that this router already has T1 lines. You can send 
911 calls to T1 lines, I don't understand why you want to add analog POTS lines 
if you already have T1s.



If you want more suggestions then perhaps take this conversation back on to the 
mailing list, others besides me might be able to give you ideas.



-mn



From: Hamu Ebiso [mailto:hebiso2...@hotmail.com]
Sent: July 24, 2017 2:45 PM
To: Norton, Mike >
Subject: Re: Configure 3845 router for 911 call.



Thank you Mike for following up!!



I Just clarified with person who gave me that project to put it together. That 
person was not really familiar with voice staff just mentioned analog phones. 
There is no analog phones here.  These users have 7942 IP phones and the idea 
was to user POT lines to call out to 911 if WAN connect down at the site. The 
idea was, since these users are in some else's building, we need to find out 
how to make them call out to 911 if WAN connection is down. Since I am not 100% 
sure of how we can achieve this. I am ok with your suggestion how to achieve 
this.



ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules

Amphenol Cable.



thanks

Hamu





From: Norton, Mike >
Sent: Monday, July 24, 2017 3:26 PM
To: Hamu Ebiso
Subject: RE: Configure 3845 router for 911 call.



Hi Hamu, I don't fully understand your call flow. POTS lines would normally 
connect to an FXO card. In that case, the easiest way to handle incoming calls 
is to do "connection plar " on the voice-port, where  is the extension 
number where you want to send the calls. But since you mention analog phones 
then it is not clear to me exactly what you are trying to do.



Maybe take the conversation back to the mail list and give more explanation of 
the call flow, might be able to get more help that way.



-mn



From: Hamu Ebiso [mailto:hebiso2...@hotmail.com]
Sent: July 21, 2017 5:49 PM
To: Norton, Mike >
Subject: Re: Configure 3845 router for 911 call.





Thank you very much Mike. I really appreciate your help.

As I mentioned it before, These employees are residing in someone else's 
building and using CUCM across WAN. The reason behind trying to use POT Lines 
is, If WAN is done these 

[cisco-voip] Call Limiting

[Haas, Neal]

I am working on a project to track calls by ID. I thought law firms might have 
this, but can't find any that work with CUCM.


1.   Unique ID

2.   allow only Unique ID to call specific numbers

3.   Limit call to 5 minutes.

4.   Record all calls

5.   Generate reporting by Unique ID

6.   Block calls to specific numbers.

Thank you for any suggestions.

Neal Haas

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Re: [cisco-voip] CDR on cucm analysis reports

[Haas, Neal]

I don’t think you can do more than 60 days on CUCM, it just does not have the 
storage to do that. We use Variphy Insight. They will give you a demo for a 
month or two. Vary easy to setup.

Thank You,

Neal Haas
NSE, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Gary 
Parker
Sent: Monday, May 22, 2017 1:06 AM
To: Brian Meade 
Cc: cisco-voip@puck.nether.net; Jonatan Quezada 
Subject: Re: [cisco-voip] CDR on cucm analysis reports


> On 21 May 2017, at 22:34, Brian Meade  wrote:
> 
> CAR DB only holds up to 30 days.  You need a billing server to offload CDR to 
> if you need to keep info longer than that.
> 
> On Fri, May 19, 2017 at 4:29 PM, Jonatan Quezada 
>  wrote:
> does anyone have any insight on where to adjust how far back to report on. I 
> get a limit when I try a report older than a month. We should be able to go 
> back for a year? right. if this is a setting for how long to archive call 
> details, where do i change that?

If you don’t already have billing server, can’t get the budget for one or want 
something a bit different to the regular packages, I can highly recommend 
Damien Hauser’s CUCM/ELK integration tools. You it’s free (all based on open 
source tools) and gives you a full elasticsearch database of CDR/CMR database 
with a Kibana frontend and logstash ingest. There’s even a load of 
pre-configured reports and visualisations for you to modify for your own needs.

GitHub repo here:
https://github.com/damhau/cucm-cdr

Some more detailed installation instructions here:
https://damienetwork.wordpress.com/2015/10/09/elk-setup-for-cucm-cdr/

You *will* need a working knowledge of linux to get this working but the 
results are worth it, imho. We get a lot more useful technical information out 
of this than we do our Tiger call logger/billing platform.


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

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Re: [cisco-voip] 4g Router for site remote access?

[Haas, Neal]

Yes we use them, Verizon is the best connection in our area. Cannot do VoIP 
calls over them. We have 1 out of 4 that needs to be reset every night. Could 
be the tower we are connecting to.

I actually used another type cradle point. You can do two SIM cards. Worked 
good. They use cradle point in Red Boxes.

Thank You,

Neal Haas
NSE, Communications

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
'Jonatan Quezada'
Sent: Wednesday, May 3, 2017 3:27 PM
To: cisco-voip@puck.nether.net; Adrian Arevalo-Orozco 
; Fernando Fernandez Lopez 
; Rodolfo Ramirez 
Subject: [cisco-voip] 4g Router for site remote access?

Is there anyone actually using these?

http://www.cisco.com/c/en/us/products/collateral/routers/819-integrated-services-router-isr/datasheet_c78-732558.html

I am looking at trying 2 of these at our farthest outreaches for IT remote 
access and possibly, depending on coverage some SIP SRST testing in the event 
of ppp failure to the pub and sub nodes?

if someone has these deployed for them in some case, how is the performance, 
caveats, gotchas? please advise.



--
Johnny Q
Voice Technology Analyst - TelNet
Chemeketa Community College
johnn...@chemeketa.edu
Building 22 Room 131
Work 5033995294
Mobile 9712182110
wMobile 5039323385
SIP 9717185162
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Re: [cisco-voip] Question regarding Rightfax Configuration and "Transmission Error"

[Haas, Neal]

Do you use SIP or PRI, we use PRI 99.9% of the time, ATT does not allow T.38 
over SIP.

I will likely get a SIP trunk from Comcast or Time Warner in the future, they 
both do T.38 over SIP. Both say they can do Faxes without issue.

Thank You,

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
Meade
Sent: Thursday, April 6, 2017 7:09 AM
To: Frank Arrasmith 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Question regarding Rightfax Configuration and 
"Transmission Error"

I've had the most luck turning ECM off and running RightFax at only 9600.  
Faxes are slower but usually higher quality and less failures.

On Wed, Apr 5, 2017 at 8:06 PM, Frank Arrasmith 
> wrote:
Calling all Rightfax gurus,
   I have the following question regarding the Rightfax configuration and 
transmission errors.

Background:
My enterprise has CUCM 10.5 with a pretty dialed in Fax/T38 setup over SIP and 
MGCP gateways.  For the most part, faxing is pretty solid inbound and outbound 
to and from PSTN GW's, CUBEs, ,analog VG's, and regular fax machines(we have a 
lot).  We have a SIP trunk connected to Rightfax 10.5 server, which is managed 
by another group where we have limited access/experience with Rightfax 
configuration settings.  It took us awhile to get the Rightfax servers with the 
correct t38 setup because they had only run traditional CAS T1 prior to us, so 
it was new to everyone, but it is up and stable except for the following issue..
Symptom :
The problem we see with Rightfax is with Transmission errors.  It starts with 
our internal customers reporting that they do not receive a fax even when the 
sender receives confirmation.  Upon further review we see that the suspect call 
is listed as a "Transmission Error" in Rightfax, so the fax never gets 
delivered to the customers account/mailbox even though the call completes.
Analysis:
Since we are running T38, we can packet capture at the server, and  we see 
normal fax protocol exchange, except for the suspect calls where we see "RTN" 
Messages.  My understanding of the T.30 protocol is that when a RTN message is 
delivered to the sender, that is an indication for the sender to slow down and 
resend the last page. We actually see in the messages where the RTN message 
gets sent, and the sender complies with the notice, and sends again at 12000, 
then call completes as normal with an EOP message and a DCN message.  In these 
cases,the Rightfax team actually looks at the fax image and may see a bit of 
blurriness, but perfectly legible text ,even tho its still marked as 
transmission failure. We have asked them if there is a setting that can be 
tuned where the RTN does not cause the service to mark the transmission as 
failure. To this they reply, "It was working fine before when we were on CAS, 
so it must be on your end."

I understand where there are cases where the RTN message is sent because the 
call quality is actually terrible, but in those cases, there is usually several 
RTN messages and the sender will drop down to 4800 or below , and then usually 
give up and the call will fail.  This type of failure is rare (unless we have a 
major outage) and in this case the sending fax sees that it failed and will 
proceed with its normal retries.
Question:
Is this RTN to transmission failure hard coded, or is this a configurable 
setting?  If this were a regular fax machine, i think this would be a non issue 
as the receiver would see the crappy page as well as the good copy of that was 
sent again in their bundle of received pages.
Any insight is greatly appreciated and for anyone just getting into FAX over IP 
with Cisco, I highly recommend the following book and Cisco Live presentations 
from these guys from Cisco TAC.

Fax, Modem, and Text for IP Telephony 
[Book]
from Textbooks.com
by David Hanes, Gonzalo Salgueiro





Thanks,
 Frank


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Re: [cisco-voip] ATT BVOIP outage

[Haas, Neal]

We have noticed that if you dial the full 11 digits, the call has a better 
success rate of going through.

Thank You,

Neal Haas
NSE, Communications

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Haas, 
Neal
Sent: Monday, April 3, 2017 11:03 AM
To: 'Michele Harttree (Americas)' <michele.hartt...@dimensiondata.com>; 
Mergenthal, Chase <chase.mergent...@bestbuy.com>; Balk, David <db...@nm.org>; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATT BVOIP outage

Yep, seeing outages in Fresno

Thank You,

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Michele Harttree (Americas)
Sent: Monday, April 3, 2017 10:35 AM
To: Mergenthal, Chase 
<chase.mergent...@bestbuy.com<mailto:chase.mergent...@bestbuy.com>>; Balk, 
David <db...@nm.org<mailto:db...@nm.org>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] ATT BVOIP outage

Yes, I am seeing this issue with one of my customers.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Mergenthal, Chase
Sent: Monday, April 03, 2017 1:31 PM
To: Balk, David; cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] ATT BVOIP outage


Not yet…

--
Chase Mergenthal | SNR: +1 612 292 1616 | M: +1 952 463 5665
7601 Penn Avenue South Minneapolis, MN 55423

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Balk, 
David
Sent: Monday, April 03, 2017 11:25 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] ATT BVOIP outage

Anyone getting lit up by this.  My ATT reps are saying a nationwide outage.

David Balk
IS Team Lead - Voice
NMH Telecom
Northwestern Memorial Health Care

541 N. Fairbanks Court, Suite 2401
Chicago, Illinois 60611
312.926.2642 office

db...@nmh.org<mailto:db...@nmh.org>
northwesternmedicine.org<http://www.northwesternmedicine.org/>

[NM-Logo-stacked-2090.png]


This message and any included attachments are intended only for the addressee. 
The information contained in this message is confidential and may constitute 
proprietary or non-public information under international, federal, or state 
laws. Unauthorized forwarding, printing, copying, distribution, or use of such 
information is strictly prohibited and may be unlawful. If you are not the 
addressee, please promptly delete this message and notify the sender of the 
delivery error by e-mail.


itevomcid
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Re: [cisco-voip] ATT BVOIP outage

[Haas, Neal]

Yep, seeing outages in Fresno

Thank You,

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Michele Harttree (Americas)
Sent: Monday, April 3, 2017 10:35 AM
To: Mergenthal, Chase ; Balk, David 
; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATT BVOIP outage

Yes, I am seeing this issue with one of my customers.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Mergenthal, Chase
Sent: Monday, April 03, 2017 1:31 PM
To: Balk, David; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATT BVOIP outage


Not yet…

--
Chase Mergenthal | SNR: +1 612 292 1616 | M: +1 952 463 5665
7601 Penn Avenue South Minneapolis, MN 55423

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Balk, 
David
Sent: Monday, April 03, 2017 11:25 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] ATT BVOIP outage

Anyone getting lit up by this.  My ATT reps are saying a nationwide outage.

David Balk
IS Team Lead - Voice
NMH Telecom
Northwestern Memorial Health Care

541 N. Fairbanks Court, Suite 2401
Chicago, Illinois 60611
312.926.2642 office

db...@nmh.org
northwesternmedicine.org

[NM-Logo-stacked-2090.png]


This message and any included attachments are intended only for the addressee. 
The information contained in this message is confidential and may constitute 
proprietary or non-public information under international, federal, or state 
laws. Unauthorized forwarding, printing, copying, distribution, or use of such 
information is strictly prohibited and may be unlawful. If you are not the 
addressee, please promptly delete this message and notify the sender of the 
delivery error by e-mail.


itevomcid
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Re: [cisco-voip] Anti-Tromboning Tie Lines to PBX

[Haas, Neal]

Back in the Day we had 11 Nortel PBX’s. I think that we had 11 PRI’s registered 
as MGCP. I don’t think we ever got the PRI’s to drop calls when a phone was 
transferred back to CUCM. Our next step was to look at SIP, but it just was not 
worth the cost to put a SIP trunk on the Nortel PBX to solve our issues (was 
like $30,000 or $40,000 to do so)

Best thing we do was purchase more CISCO phones :>

Thank You,

Neal Haas
NSE, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Tuesday, March 7, 2017 2:54 PM
To: Cisco VoIP Group 
Subject: [cisco-voip] Anti-Tromboning Tie Lines to PBX

I'm not sure how often this gets configured out in the wild, but I do know that 
there are few different flavors of this feature, depending on your scenario.  
E.g., CUBE Media 
Anti-Tromboning

The one I'm interested in would involve a tie line to a legacy PBX, and I'm not 
married to the IP technology: SIP, H323, MGCP, whichever will get the job done.

CUCM ---IP---> VGW ---T1/PRI---> PBX

The call scenario I'd like to see if we can avoid is when a call is established 
from CUCM to the other PBX, and then a phone on the far side transfers the call 
to an IP Phone on the CUCM side, I'd like the PRI usage to drop to 0 trunks in 
use, instead of nailing up 2 trunks for the duration of that call.

I'm looking for your experience and feedback in configuring anti-tromboning in 
this scenario.  One of the hardest parts in researching this, is knowing what 
terms to search for.  I think I've narrowed down the terminology to QSIG Path 
Replacement, as described in the CUCM 
SRND.

Though, without real-world working experience, it's hard to know if I'm right 
or not.

So, I know there's some good legacy telco knowledge out on this list.  What do 
you know?  Thanks!
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Re: [cisco-voip] Nuance/UCCX 10.6

[Haas, Neal]

Looking for a basic example of this process:


1.   UCCX Record Message Step.

2.   Deliver message to Agent.


Thank You,

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Tanner Ezell
Sent: Tuesday, October 11, 2016 9:48 AM
To: Aaron Banks 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Nuance/UCCX 10.6

I'm not sure why Nuance would enter the equation at all. Take the message via 
the UCCX record message step, deliver that to the agent as the message. No need 
to involve Unity or Nuance.

On Mon, Oct 10, 2016 at 8:51 PM, Aaron Banks 
> wrote:


Has anyone ever used Nuance to deliver speech to text messages to an agent?  
The case is, a caller leaves a voicemail message, and then that message is 
delivered to an agent.  I have never heard of this before and not even sure how 
it would work given that the voicemail message would reside in unity.

Thanks

Aaron

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Re: [cisco-voip] 8851 and Transfers

[Haas, Neal]

No side Car, and this is a shared line across multiple phones.

Thank You,

Neal Haas

From: Carlo Calabrese [mailto:carlo_calabrese2...@yahoo.com]
Sent: Wednesday, October 5, 2016 2:17 PM
To: Haas, Neal <nh...@co.fresno.ca.us>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 8851 and Transfers

Does this have a Side Car? we had a problem with this awhile back, not sure if 
same model but we had to roll back.

On Tuesday, October 4, 2016 7:16 PM, "Haas, Neal" 
<nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>> wrote:

We have 8851 phones, just upgraded to the new firmware, now we have issues.


When a phone call is in the process of being transferred and a new call comes 
in the phone will not let you finish the transfer. How can we stop this issue?

Anyone have a solution - besides rolling back the firmware?




Neal Haas
?
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[cisco-voip] 8851 and Transfers

[Haas, Neal]

We have 8851 phones, just upgraded to the new firmware, now we have issues.


When a phone call is in the process of being transferred and a new call comes 
in the phone will not let you finish the transfer. How can we stop this issue?

Anyone have a solution - besides rolling back the firmware?




Neal Haas
?
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Re: [cisco-voip] Trouble Getting 8831 to register

[Haas, Neal]

You might see this: 
https://supportforums.cisco.com/discussion/11905636/cisco-cp-8831-phone-not-registering


Thank you,
Neal Haas

Neal Haas
IT Analyst
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890
FAX (559) 455-4747

County of Fresno
Information Technology Services Division
2048 N Fine Ave
Fresno, CA 93727

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Thomas Theisen
Sent: Monday, June 27, 2016 7:31 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Trouble Getting 8831 to register


I have been working with TAC trying to get our first 8831 phone to register 
with our 10.0.1.11900-2 CUCM but we have had no success.  The strange thing is 
that I found that the registration is "Rejected" before I even patch the phone 
into the network.  TAC updated the load information for the 8831 to 
sip8831.9-3-3-TO-10-3-1 so I think I have the most up-to-date load for the 
phone.  I am using a switch port that had a 7965 attached to it so I know the 
pathway to CUCM is clear.  This is the first sip phone we have tried to 
introduce to the system but we have sip trunks so I know the correct ports are 
enabled for sip.

--
Thomas Theisen
Network Tech IV
University of Rhode Island
401-874-7870
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Re: [cisco-voip] Helios IP Uni intercom units Dual Button.

[Haas, Neal]

Are they on the correct VLAN? You might need to force this on the switch port.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: Wednesday, June 8, 2016 8:18 AM
To: Cisco VoIP Group 
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] 8851 Rollout

[Haas, Neal]

We are going from 7900 series phones to 8851 phones.

Is there a software that will copy SCCP phones to SIP phones?

Neal Haas

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Re: [cisco-voip] cucm9 insert Local enduser via bat AD sync enable

[Haas, Neal]

How about an Application user account or an End User account?

Thank you,
Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of abbas 
Wali
Sent: Thursday, February 18, 2016 10:38 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] cucm9 insert Local enduser via bat AD sync enable

hi all,

need to add around 100 local users in CUCM 9 which is sync with AD

​cant add them to AD as these are external accounts.

any idea?​

--
​thanks ​

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Re: [cisco-voip] UCCX 9 prompts gibberish

[Haas, Neal]

this has worked 100% of the time for the past 6 years, For thousands of 
recordings.

http://snafder.blogspot.com/2011/01/saving-wav-files-in-ccitt-u-law-format.html

no issues. If it doesn’t work then you missed a step.

Neal

From: abbas Wali [mailto:abba...@gmail.com]
Sent: Tuesday, February 16, 2016 7:17 AM
To: James Buchanan <james.buchan...@gmail.com>
Cc: Haas, Neal <nh...@co.fresno.ca.us>; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX 9 prompts gibberish

​yes seen them but again they save it in g711's. also in the new audacity there 
is no​ CCIT U-Law !!

On 16 February 2016 at 15:12, James Buchanan 
<james.buchan...@gmail.com<mailto:james.buchan...@gmail.com>> wrote:
Even to g729?

On 16 Feb 2016, at 10:11 AM, Haas, Neal 
<nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>> wrote:
I use audacity all of the time to convert to a “Cisco” format.. here are two 
links for you
http://www.netcraftsmen.com/uc-toolkit-using-audacity-to-create-and-edit-cisco-uccx-prompts/

http://xyfon.com/tech-tips/saving-wav-files-for-cisco-unified-call-centre-express-prompts-uccx-using-audacity/


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of abbas 
Wali
Sent: Tuesday, February 16, 2016 7:09 AM
To: James Buchanan <james.buchan...@gmail.com<mailto:james.buchan...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] UCCX 9 prompts gibberish

indeed that is the case.
thanks alot.

any free tool to record g729. Have tried Audacity but cant bring it as low as 
8kbps.

On 16 February 2016 at 14:29, James Buchanan 
<james.buchan...@gmail.com<mailto:james.buchan...@gmail.com>> wrote:
Hello,
In the System settings, are you set to use G729 for your prompts or G711? UCCX 
will not play one or the other. If you record G711 and upload to a system set 
to play G729, that'll be the result.
Thanks,
James

On Tue, Feb 16, 2016 at 9:26 AM, abbas wali 
<abba...@gmail.com<mailto:abba...@gmail.com>> wrote:
Hi guys,

Just need a quick help here.

Every prompt I record ( via UnityC or Audacity etc ) upload and can only hear 
gibberish.

But when I load an already saved file in G729 – it plays okay.

I have checked an my regions for  phone dpool and application trigger are in 
the same region set to g711.

The only other thing is that I am calling from a softphone vpn’ed. ( but that 
shouldn’t make any difference )

Please help.
Thanks in advance.

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--
Abbas Wali



--
Abbas Wali
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Re: [cisco-voip] UCCX 9 prompts gibberish

[Haas, Neal]

I use audacity all of the time to convert to a “Cisco” format.. here are two 
links for you
http://www.netcraftsmen.com/uc-toolkit-using-audacity-to-create-and-edit-cisco-uccx-prompts/

http://xyfon.com/tech-tips/saving-wav-files-for-cisco-unified-call-centre-express-prompts-uccx-using-audacity/


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of abbas 
Wali
Sent: Tuesday, February 16, 2016 7:09 AM
To: James Buchanan 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX 9 prompts gibberish

indeed that is the case.
thanks alot.

any free tool to record g729. Have tried Audacity but cant bring it as low as 
8kbps.

On 16 February 2016 at 14:29, James Buchanan 
> wrote:
Hello,
In the System settings, are you set to use G729 for your prompts or G711? UCCX 
will not play one or the other. If you record G711 and upload to a system set 
to play G729, that'll be the result.
Thanks,
James

On Tue, Feb 16, 2016 at 9:26 AM, abbas wali 
> wrote:
Hi guys,

Just need a quick help here.

Every prompt I record ( via UnityC or Audacity etc ) upload and can only hear 
gibberish.

But when I load an already saved file in G729 – it plays okay.

I have checked an my regions for  phone dpool and application trigger are in 
the same region set to g711.

The only other thing is that I am calling from a softphone vpn’ed. ( but that 
shouldn’t make any difference )

Please help.
Thanks in advance.

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--
Abbas Wali
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Re: [cisco-voip] Installation new phone firmware cop file installation

[Haas, Neal]

Just on the TFTP server.

Thank you,
Neal Haas

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Reto 
Gassmann
Sent: Friday, January 29, 2016 6:55 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Installation new phone firmware cop file installation

Hello Group

I have a UCM cluster 10.5 with 8 servers. TFTP service runs on the publisher.
I now have a new 8831 to install and have to Upgrade the Firmware 
(cmterm-8831-sip.10.3.1SR2EU.k3.cop.sgn) in order to get the 8831 running.

Is it ok to install the cop file on the TFTP Server and restart the TFTP after 
installtion or so I have to install the cop file on every server in the cluster?

Tanks a lot and have a great weekend
Regards Reto
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Re: [cisco-voip] BLF Pickup button

[Haas, Neal]

Take a look at this DOC. Search for “Configure BLF Call Pickup”

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_1_1/ccmfeat/CUCM_BK_C3E0EFA0_00_cucm-features-services-guide-91/CUCM_BK_C3E0EFA0_00_cucm-features-services-guide-91_chapter_0111.html


Thank you,
Neal Haas


From: Michael T. Voity [mailto:mvo...@uvm.edu]
Sent: Monday, November 16, 2015 7:40 AM
To: Haas, Neal; 'voip puck'
Subject: Re: [cisco-voip] BLF Pickup button

Neal,

Thanks for the response,That thought did cross my mind,  the users didn't 
want it that feature.

-Mike



Michael T. Voity

Network Engineer

University of Vermont




On 11/16/2015 10:19, Haas, Neal wrote:
Why don’t you configure it as a Line if you want to pick it up?

Thank you,
Neal Haas


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

Hello,

My system is CUCM 10.5.2.12900-14

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

In the BLF config I clicked "Call Pickup"

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

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

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

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

Ideas?

Thanks,

-Mike



--

Michael T. Voity

Network Engineer

University of Vermont

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Re: [cisco-voip] CUCM with Lync RCC

[Haas, Neal]

Translated “To work properly though, Microsoft wants to be the ingress for all 
your calls” MS wants you to ditch Cisco and go with their Phone system :>

Thank you,
Neal Haas


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Matt 
Slaga (AM)
Sent: Tuesday, September 29, 2015 8:11 AM
To: Mike King; george.hend...@l-3com.com
Cc: Cisco VoIPoE List
Subject: Re: [cisco-voip] CUCM with Lync RCC

Skype for Business Call via Work may be a solution for you.  It is Microsoft’s 
replacement for the antiquated RCC solution.  To work properly though, 
Microsoft wants to be the ingress for all your calls.

https://technet.microsoft.com/en-us/library/dn933908.aspx



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Mike 
King
Sent: Tuesday, September 29, 2015 10:44 AM
To: george.hend...@l-3com.com
Cc: Cisco VoIPoE List 
>
Subject: Re: [cisco-voip] CUCM with Lync RCC


George,

Microsoft has recently rebranded Lync to Skype for Business:
https://support.office.com/en-us/article/Lync-is-now-Skype-for-Business-%E2%80%94-see-what-s-new-aba02d7e-c801-4a82-bccd-e7207240f612

They have the following help guide:
https://support.office.com/en-us/article/Make-a-Skype-for-Business-call-but-use-your-PBX-desk-phone-for-audio-6a316c11-a05e-460c-b969-32ff0ad848e6


Unfortunately, they seem to be hiding the implementation details, this looks 
more user facing.

On Tue, Sep 29, 2015 at 9:24 AM, 
> wrote:
We are in the process of upgrading our Clients from Office Communicator to 
Lync.  One of the things we’ve used with MOC is CUCILync, which works great and 
we can make calls on our desk phone in MOC.  However, with Lync, this doesn’t 
seem to work the same.  I’ve installed a compatible version of CUCILYNC that 
works with Lync, but don’t see any way to make desk phone calls in Lync.  There 
is an option in Lync when I right-click a contact for “Place a Call” and seems 
to be an Icon for CUCILYNC, but it does nothing when I click on it.

Also, if we wanted to just remove CUCILYNC, to have call/phone control in Lync, 
would this require we stand up a Cisco Presence server?

Thanks,

Bill Hendrix


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Re: [cisco-voip] Changing the enable password on VG224/2901/2911/881's

[Haas, Neal]

TAC can reset them for you, or just boot a Linux distro and reset them yourself 
(2nd one is not supported :)

Thank you,
Neal Haas


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
norm.nichol...@kitchener.ca
Sent: Saturday, August 22, 2015 8:13 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Changing the enable password on VG224/2901/2911/881's


Is there a simple process to change these passwords without reloading O/S ?




Thanks




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


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[cisco-voip] New 8851 Phones

[Haas, Neal]

So we got some 8851 phones into test. Only thing we don't like is on a 
multi-line setup the last line you talked on is selected as primary. How do you 
force the primary line to always be the first line?

You can do this on the 7965 phones..

Thank you,
Neal Haas



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Re: [cisco-voip] freeware billing server

[Haas, Neal]

Are you talking about CDR? FYI Cisco CDR is in SQL statements, you could just 
import into MS SQL and run a query to get what you need.

You could also look at this: 
https://uccisco.wordpress.com/2014/05/16/cisco-cdr-call-detail-record-reporting/

Or just get http://www.variphy.com


Thank you,
Neal Haas

Neal Haas
IT Analyst, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890
FAX (559) 455-4747

County of Fresno
Information Technology Services Division
1020 S Tenth St
Fresno, CA 93702

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Abebe 
Amare
Sent: Tuesday, June 23, 2015 2:26 AM
To: cisco voip
Subject: [cisco-voip] freeware billing server

Hi,

I am looking for a freeware billing server for CUCM. Do you guys know any that 
can do the job?

Thanks in advance

Abebe
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Re: [cisco-voip] What is better than MediaSense for recording calls?

[Haas, Neal]

I have used:

Calabrio – for are call center agents can we say $$$, and the it doesn’t 
have the in-depth management that I would like. WFM is nice!
VPI – We use this for EMS and 911 Calls. If you screw up people could die. 
Price is good and easy to use. Great product.
We looked at media since and everyone says it is way too hard. So we don’t use 
it.

Thank you,
Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Chris 
Ward (chrward)
Sent: Monday, June 22, 2015 6:41 AM
To: Tim Smith; Derek Andrew; Cisco VoIP Group
Subject: Re: [cisco-voip] What is better than MediaSense for recording calls?

Just logged into Search and Play and see your issue with the call duration, let 
me ask about that. I misunderstood at first.

+Chris
TME - MediaSense and Unity Connection

From: Chris Ward (chrward)
Sent: Monday, June 22, 2015 9:35 AM
To: 'Tim Smith'; Derek Andrew; Cisco VoIP Group
Subject: RE: [cisco-voip] What is better than MediaSense for recording calls?

Hi Tim,

Yes, still separate recordings, call stitching, where we build you a single 
file from multiple call legs will probably come in our next major release.

Let me look into the time thing, I think we should just assume 0 as the min if 
nothing is entered. I will see if we can get a bug or enhancement filed on that.

Also, since it seems like you have some experience with MediaSense, if you have 
feedback or product direction questions, I can put you in touch with one of our 
PMs. We are always looking to help validate the roadmap and make sure we 
haven’t missed any smaller pieces.

+Chris
TME - MediaSense and Unity Connection

From: Tim Smith [mailto:tim.sm...@enject.com.au]
Sent: Monday, June 22, 2015 1:12 AM
To: Chris Ward (chrward); Derek Andrew; Cisco VoIP Group
Subject: RE: [cisco-voip] What is better than MediaSense for recording calls?

Hi Chris,

Seems better, but all separate recordings still.

Is there some logic behind that?

Also the search is pretty strange, you can’t just say longer than 30 seconds 
for example, you have to specify the upper bound as well?


Cheers,

Tim

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Tim 
Smith
Sent: Saturday, 20 June 2015 12:59 AM
To: Chris Ward (chrward); Derek Andrew; Cisco VoIP Group
Subject: Re: [cisco-voip] What is better than MediaSense for recording calls?

Hi Chris,

Awesome, TAC told me about that in ver 11, but said it wasn’t available til mid 
August ☺
I’ll download it and check it out.



Cheers,

Tim.

From: Chris Ward (chrward) [mailto:chrw...@cisco.com]
Sent: Friday, 19 June 2015 10:40 PM
To: Tim Smith; Derek Andrew; Cisco VoIP Group
Subject: RE: [cisco-voip] What is better than MediaSense for recording calls?

Hi Tim,

We added BiB call association in 10.5 and completed our call association story 
(CUBE and GW support) in the 11.0 release. Maybe an upgrade would help you out. 
11.0 is available on cisco.com right now.

+Chris
TME - MediaSense and Unity Connection

From: Tim Smith [mailto:tim.sm...@enject.com.au]
Sent: Friday, June 19, 2015 4:22 AM
To: Chris Ward (chrward); Derek Andrew; Cisco VoIP Group
Subject: RE: [cisco-voip] What is better than MediaSense for recording calls?

For recording them – yes ☺

Finding them, managing them etc.. ?
I am struggling with the way it splits calls and you have to manually correlate 
them with CCID at the moment. (CUBE not BIB)

Has anyone used any of the third party UI’s?
I was checking out the Aurus5 one.

Donoma and Mida look good, but overkill for this particular application.

Cheers,

Tim.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Chris 
Ward (chrward)
Sent: Wednesday, 17 June 2015 3:10 AM
To: Derek Andrew; Cisco VoIP Group
Subject: Re: [cisco-voip] What is better than MediaSense for recording calls?

Also… nothing is better than MediaSense for recording calls! ☺

+Chris
TME - MediaSense and Unity Connection

From: Chris Ward (chrward)
Sent: Tuesday, June 16, 2015 1:09 PM
To: 'Derek Andrew'; Cisco VoIP Group
Subject: RE: [cisco-voip] What is better than MediaSense for recording calls?

Derek,

A product like Calabrio Search and Play can plug in on top of MediaSense to 
provide this functionality. There are others as well, including NICE, or Mida 
solutions. This means you wouldn’t have to reconfigure your recording 
architecture or purchase a new recording product. You would really just be 
purchasing a new GUI.

Just so you know, MediaSense will be adding the groups/roles/permissions 
feature in our 11.5 release which should be out at the end of 2015 or beginning 
of 2016. So depending on your urgency, you could just wait.

Also, in 11.0, we eliminated the server SKU so that you wouldn’t actually have 
to purchase anymore server licenses for more instances/clusters, you just need 
to provide the virtualization resources to run it on.

+Chris
TME - MediaSense and Unity Connection

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On 

[cisco-voip] Leap Second - 6/30

[Haas, Neal]

Could the Leap Second on June 30th cause any issues with the Call 
Managers/Telephony? Anyone worried?

http://www.usatoday.com/story/news/nation-now/2015/06/09/leap-second-atomic-clock-earth-rotation/27786459/


Thank you,
Neal Haas



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Re: [cisco-voip] CUCM 10.5 and OpenText 10.6

[Haas, Neal]

We had the same thing with our Accuroute Fax server. We called TAC and it was 
the time out on the SIP trunk needed to be increased. As soon as we called in 
the TAC person had us fixed up in minutes.

I don’t know the exact thing that had to be changed; the guy that did the work 
is out this week.

If you have 79XX phones, update the COP file for your phones, transfers while 
on a call with a head set are broken. Calls will just drop with a transfer……


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Countryman, Edward
Sent: Thursday, April 30, 2015 3:06 PM
To: Cisco VOIP
Subject: [cisco-voip] CUCM 10.5 and OpenText 10.6

We have had right fax 10.5 up and running with a SIP trunk from our CUCM 8.6.2 
environment for a while now.

This past weekend we updated to CUCM 10.5.2 and are no longer able to fax out 
from RF.

From the trace, the SIP call appears to fail with the following error after the 
INVITE

488 Not Acceptable Here

Do would anything have changed in 10.5 to possibly be causing this?   Weird 
thing is the inbound faxes are working fine.
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Re: [cisco-voip] Upgrading CUCM from 8.5 to 10.5

[Haas, Neal]

10.5 is VM only, 8.6.1 usually was MCS hardware.

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


Neal Haas

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


Supported vs the system will upgrade are two different things.
On 27/03/2015 1:51 AM, Nick 
csv...@googlemail.commailto: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.pdfhttp://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/upgradehttp://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] Permission Information for helpdesk user to be able to add End Users in CCMAdmin

[Haas, Neal]

Sorry, “Seems a helpdesk admin role would have been included in the box!” this 
is Cisco we are talking about……..


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Tuesday, February 24, 2015 1:56 PM
To: Kevin Przybylowski; cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Permission Information for helpdesk user to be able 
to add End Users in CCMAdmin

So I HAVE to create a custom role.

Somehow I thought I was missing a built-in role.  Seems a helpdesk admin role 
would have been included in the box!

From: Kevin Przybylowski [mailto:kev...@advancedtsg.com]
Sent: Tuesday, February 24, 2015 4:46 PM
To: Jason Aarons (AM); cisco-voip 
(cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: RE: Permission Information for helpdesk user to be able to add End 
Users in CCMAdmin


Check out this doc for a custom role:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmsys/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100_chapter_0100.html#CUCM_TK_C0C2BD4E_00

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Tuesday, February 24, 2015 4:31 PM
To: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: [cisco-voip] Permission Information for helpdesk user to be able to 
add End Users in CCMAdmin

I have a helpdesk user that I want to be able to add end users (non ldap).I 
gave him the role of Standard CCM User Management but he still can’t create end 
users under User Management  End Users.  He is able to view existing users  
but not modify.

CUCM 10.5.1



[cid:image001.jpg@01D0503A.3CACD7B0]


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Re: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4 before answer

[Haas, Neal]

Check CPU usage on the VM, we just encountered a bug were CPU is at 100%, had 
to go to failover and everything is working fine now. This really hit OWA, but 
response time for everything was really slow.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Monday, February 02, 2015 9:53 AM
To: cisco-voip (cisco-voip@puck.nether.net)
Subject: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4 
before answer

Customer states that it's taking too long for Unity Connection to answer when 
he presses the Messages button.  It's SCCP integrated.  The Hunt Pilot looks 
good, no duplicate or unassigned DNs, no overlapping number range.  Been in 
production and was working fine previously.

I did notice the checkbox in the Hunt List For Voice Mail Usage is unchecked. 
  After hours we can check that.


CUCMC 10.5.1.1-7

CXN 10.5.1.11900-13

Jason Aarons
Consultant
Dimension Data
904-338-3245 mobile


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Re: [cisco-voip] UCCX 9x Desktop Administrator Work Flow Configuration Reason Codes gone?

[Haas, Neal]

You will also noticed that you can only select 1 reason code now, older version 
you could select multiple.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Monday, January 19, 2015 11:03 AM
To: Anthony Holloway
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] UCCX 9x Desktop Administrator  Work Flow 
Configuration  Reason Codes gone?

Yes I forgot it was under the Desktop Suites.  I was thinking that was just 
CAD/Supervisor ☺

From: avhollo...@gmail.commailto:avhollo...@gmail.com 
[mailto:avhollo...@gmail.com] On Behalf Of Anthony Holloway
Sent: Monday, January 19, 2015 12:12 PM
To: Jason Aarons (AM)
Cc: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] UCCX 9x Desktop Administrator  Work Flow 
Configuration  Reason Codes gone?


For reference, here is the latest 10.6 version and it shows that CDA is still 
there.

[Inline image 1]

On Mon, Jan 19, 2015 at 9:53 AM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:
In UCCX 9x the 32-bit Desktop Administrator app is gone from Plug Ins.

This means Reason Codes should now be configured from the top right drop down 
for Desktop Administrator ?

Figure 5 page 28 does not match the Web GUI.  It shows the old 32-bit 
application.
http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_9_0/user/guide/cda90ccxug.pdf


What am I mis-understanding?  Did a upgrade from 7.0.1SR5 x to 9.0.2 and the 
Agents noticed all the Reason Codes are gone…..

[cid:image002.jpg@01D034BC.026A6ED0]

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Re: [cisco-voip] unity connection full mailboxes - notifying callers

[Haas, Neal]

We ended up increasing the quota size, with 6000 users we only use 20% of 
storage.

Also making the aging policy faster. No one retrieves deleted voicemail.

We did SMTP, but once we did single inbox we turned SMTP off. 80% of our users 
use Single Inbox and love it! 

Neal Haas


-Original Message-
From: Erick Bergquist [mailto:erick...@gmail.com] 
Sent: Friday, January 16, 2015 8:39 AM
To: Chris Ward (chrward)
Cc: Haas, Neal; Justin Steinberg; Cisco VOIP
Subject: Re: [cisco-voip] unity connection full mailboxes - notifying callers

Thanks for the additional info.

Guess there are couple options then on 9.1 -- increase quota, set outside 
caller messages to be impacted by quota, set message aging policy to delete 
messages in inbox or saved messages after x days.  I guess we could also 
disable single inbox and do SMTP notifdication onlly w/accept and relay to smtp 
address.

On Fri, Jan 16, 2015 at 9:28 AM, Chris Ward (chrward) chrw...@cisco.com wrote:
 Neal,

 Welcome to the future! I think this was a feature we added in 10.0 as Single 
 Inbox drove people away from dialing into their voicemail box and therefore 
 no one knew when their account was full and so they would just see the 
 consequences of a full inbox with no warnings or alerts. The SMTP alerts were 
 added to remedy this.

 +Chris
 TME - MediaSense and Unity Connection

 -Original Message-
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf 
 Of Haas, Neal
 Sent: Friday, January 16, 2015 10:07 AM
 To: 'Erick Bergquist'; Justin Steinberg
 Cc: Cisco VOIP
 Subject: Re: [cisco-voip] unity connection full mailboxes - notifying 
 callers

 We upgraded to 10.5.1.1-7 on unity conn, and all of a sudden end 
 users were complaining that they were getting emails saying that there 
 email box was full.  (but I check all of the time I don’t know why I 
 would get this email..last checked 60 days ago)

 So for the time being I would say that notification is working.

 We didn’t change ANY settings, just a regular upgrade.

 Neal Haas

 -Original Message-
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf 
 Of Erick Bergquist
 Sent: Thursday, January 15, 2015 11:31 PM
 To: Justin Steinberg
 Cc: Cisco VOIP
 Subject: Re: [cisco-voip] unity connection full mailboxes - notifying 
 callers

 Has there been any update or better way to handle this in 9.1 since the last 
 discussion in this thread?

 Running into this issue on CUC 9.1.1 with a client. Users never check 
 messages via TUI and aren't getting notified of their mailbox being over 
 quota due to external messages still being allowed to be left.
 They noticed a problem when they deleted messages in outlook and they weren't 
 being deleted but showing as saved messages or outlook deleting the message 
 then it showing back up in their inbox because unity wasn't putting it in 
 deleted items folder.

 I can see issue with SMTP notification like others have mentioned, as that is 
 done via another method and if they aren't using a SMTP SmartHost at all it 
 requires another messaging interface to be configured and setup. Some places 
 are getting strict about basic SMTP relays as well. How does that work with 
 Office365 for instance?

 Erick




 On Wed, Dec 4, 2013 at 6:08 PM, Justin Steinberg jsteinb...@gmail.com wrote:
 The 10.0 feature will help with single inbox users, but I really wish 
 there was a better way.

 Something like provide the option to modify the behavior of the 
 message aging rules with a new option that ages the message out of 
 unity connection, breaks the ews link to exchange, and leaves the WAV 
 file in the users mailbox as a normal email.  I also support some law 
 firms and they have no quota in exchange and are frustrated with what 
 they perceive as an artificial limit imposed by unity.

 This is especially bad for migrations from unity to connection.

 While we are on the topic, on my wish list is more collaboration 
 between the unity team and jabber group. It would be nice if the 
 jabber voicemail tab could show the quota status. Forward voicemails 
 in jabber. Dispatch messages in jabber.

 On Dec 3, 2013 2:34 PM, Chris Ward (chrward) chrw...@cisco.com wrote:

 Well, the SMTP notification can be used outside of Single Inbox too 
 to help alert users.



 “ISM affects what subscribers hear when they call other subscribers 
 from their primary or alternate extensions and are forwarded to the 
 greetings of the subscribers they call. If they then leave a 
 message, ISM affects what the called subscriber hears and can do 
 when listening to the message. When ISM is enabled, Cisco Unity 
 recognizes that the calling extension is associated with a 
 subscriber and accordingly plays the internal greeting of the called 
 subscriber.
 Additionally, when the called subscriber later listens to the 
 message, Cisco Unity plays the recorded voice name of the subscriber 
 who left the message and allows

Re: [cisco-voip] licensing turnaround time 8.6 to 10.5

[Haas, Neal]

It just took us 39 days to get our licensing for 9.1.1 to 10.5


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Bill 
Talley
Sent: Thursday, December 11, 2014 3:50 PM
To: Josh Warcop
Cc: cisco-voip
Subject: Re: [cisco-voip] licensing turn around time 8.6 to 10.5

I can send you a list of several SRs offline.  We provide Cisco SOs, UCSS 
contracts, Cisco.comhttp://Cisco.com usernames, LCU files and PLM license 
requests.  The standard response WAS 5-7 business days, but within the past two 
weeks the standard response has increased to 10-12 business days while awaiting 
approval from a product manager.

Sent from an Apple iOS device with very tiny touchscreen input keys.  Please 
excude my typtos.

On Dec 11, 2014, at 5:45 PM, Josh Warcop 
j...@warcop.commailto:j...@warcop.com wrote:
I'm not sure what licensing team you're working with but there is a specific 
migration team for Collab products. We do several upgrades monthly for clients 
and it has never been over 24 hours. Understanding everything that needs to be 
done prior to engaging TAC is usually the fault of the engineers or client.

SWSS Contract in place? If not still current  ESW/UCSS? Have you verified with 
UCSS support?  Has LCT been submitted properly with the online form? Has PUT 
been used properly to acquire any licensing?

This can all be done in advance and within a day. If you can't get the stuff 
via the automated methods and open a case presenting everything properly is 
necessary to get past level 1 GLO and to migration support.

Like I mentioned earlier we are never two business days beyond getting ELM 
licensed after an upgrade.

Thanks!

Sent from my Windows Phone

From: Bill Talleymailto:btal...@gmail.com
Sent: ‎12/‎11/‎2014 6:36 PM
To: Erick Wellnitzmailto:ewellnitzv...@gmail.com
Cc: cisco-voipmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] licensing turn around time 8.6 to 10.5
Hahahahahahahahahahaha...

Sorry I shouldn't laugh, but Cisco's license upgrade process for UC 10 is 
laughable at best.  Licenses have to be approved by a product manager and its 
now increased to a 10-12 business day process.   Luckily theres a lengthy grace 
period before you're locked out of provisioning the servers.

Sent from an Apple iOS device with very tiny touchscreen input keys.  Please 
excude my typtos.

 On Dec 11, 2014, at 5:19 PM, Erick Wellnitz 
 ewellnitzv...@gmail.commailto:ewellnitzv...@gmail.com wrote:

 All,

 Quick question about licensing for CUCM/UCONN

 Going from 8.6 to10.5 I should be able to run the license count tool and 
 submit to licensing along with a contract number, correct?  Is there usually 
 a long turn around time or is it still a day or less?

 Thanks!
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Re: [cisco-voip] CUCM 8.6 promotion of new pub

[Haas, Neal]

All subs will need to be rebooted, hence the re-registration of all phones.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of TG
Sent: Saturday, November 01, 2014 7:23 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CUCM 8.6 promotion of new pub

Hi all,
In order to promote a new publisher to existing phones, cert consolidation is 
required on existing cluster to authenticate the new publisher and be able to 
register successfully. Since this consolidation step (even without resetting 
the service) generates a reset on all phones of existing cluster, is there a 
way to control this behavior in order to prohibit 20 or 30,000 phones from 
resetting all at the same time?


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Re: [cisco-voip] Phone question

[Haas, Neal]

James

We use these, they work well (we use valcom at our Coroners Facility - they 
push that button 20 - 40 times a day without issue), and there are others like 
(www.cyberdata.net/products/voip/digitalanalog/intercom_endpoints.html) I Have 
used cyberdata for another application I reset that end point minimum of 2 x's 
a month. I don't know the usage on it.


Neal Haas


-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Andrewartha
Sent: Wednesday, October 22, 2014 7:32 PM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Phone question

On 22/10/14 22:56, Jose Colon II wrote:
 We do this with lobby phones... We make a lobby partition. Then 
 create a translation pattern that is blank and apply the partition. 
 Then in the partition add the DN you want it to dail.
 
 What happens is that when that phone is pulled off hook. It auto dials 
 the number we set in the translation pattern.
 
 Not sure if that helps you but we use it for phones in areas where you 
 need to call someone to bizz in someone at the door.

Speaking of door entries, what do people use for intercoms? I've found 
http://www.valcom.com/products/ip_solutions/ip_intercomdoorphones.htm
which seems alright, but I was wondering if there's any better options.

Thanks,

--
James Andrewartha
Network  Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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[cisco-voip] Phone question

[Haas, Neal]

Is there a way to have a phone when picked up automatically conferenced into an 
ongoing call?

We have a walk up counter they call a lot into a translator service, we want to 
make it easy for the client to get onto the call. Simply just pick up the 
handset.

Is this possible?


Neal Haas

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[cisco-voip] Phone Configuration

[Haas, Neal]

We use several Translations services, is there a way to configure a phone to 
join into a phone call without pushing anything?
Basically, our Employees call translation service and gives the account 
information, then gestures to the client to pick up the phone, at this point 
the client phone just joins into the conference without the customer pushing 
any buttons.

Is this possible?


Neal Haas
IT Analyst, Communications

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Re: [cisco-voip] UCCX Reporting

[Haas, Neal]

Do you know if you use Historical reporting or CUIC?


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
Meade
Sent: Thursday, September 18, 2014 11:33 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] UCCX Reporting

Hey all,

I know nothing about UCCX reporting.  Does anyone know if there's a canned 
report I can pull to see how many calls had hold times greater than x minutes?

If not, do you know where in the DB this info is stored so I can write a query?

Thanks,
Brian
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[cisco-voip] Upgrade Question

[Haas, Neal]

So just checking with people, what is an estimated cost to have an outside 
company virtualize your environment? (Just to show up and do the work, company 
already has the UC Chassis)

$5,000 or more? Any companies that have paid someone to do this type of work 
want to share?

My boss asked me how much money we are saving doing this work ourselves.


Neal Haas
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Re: [cisco-voip] Upgrade Question

[Haas, Neal]

We are CUCM 9.X, 4 MCS's 5000 phones. Straight MCS to VM no upgrade.

thanks


Neal Haas

From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com]
Sent: Tuesday, September 09, 2014 2:04 PM
To: Haas, Neal; cisco-voip@puck.nether.net
Subject: RE: Upgrade Question

I guess it depends on the size of the environment and does it involve an 
upgrade at the same time.. I can tell you it takes me about 24 total hours of 
work usually 3 business days to do a start to finish Physical to Virtual 
migration of something physical to a BE6K environment. This is with CUCM and 
UCXN only and always involves an upgrade to 9x/10x. If you are already on 8 or 
9 and don't want to upgrade and just DRS restore it'd be less time.

These are small 100-300 users installs.

HTH


Matthew G. Loraditch - CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Haas, 
Neal
Sent: Tuesday, September 09, 2014 4:45 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Upgrade Question

So just checking with people, what is an estimated cost to have an outside 
company virtualize your environment? (Just to show up and do the work, company 
already has the UC Chassis)

$5,000 or more? Any companies that have paid someone to do this type of work 
want to share?

My boss asked me how much money we are saving doing this work ourselves.


Neal Haas
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Re: [cisco-voip] C240M3 16 drives for CUCM/UCCE

[Haas, Neal]

Raid 10 all of the way! We have 4 C240’s. (Setup 2 Raid 5’s – then create them 
into a Raid 10. There is no button that just says create a Raid 10 from the 
bare drives.)


Neal Haas
IT Analyst, Communications
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Mike 
Hyde
Sent: Thursday, September 04, 2014 9:22 PM
To: Jason Aarons (AM)
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] C240M3 16 drives for CUCM/UCCE

We used SD flash disks for esxi (mirrored) and 2 raid 5 sets for the vms.

Sent from my iPhone

On Sep 4, 2014, at 10:31 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:
We have had customers have more than a single drive failure in a Raid5 set with 
all your vms running.  It was pretty.

So perhaps 2 drives raid 1 for ESXi and ISOs, 7 drives raid 5 for vms, 7 drives 
raid 5 for vms

But yeah the existing layout seems odd….but I do what it says…but I’m not sure 
where the contact center bu gets these setups from.

Jason Aarons
Consultant
Dimension Data
+1-904-338-3245

From: Charles Goldsmith [mailto:wo...@justfamily.org]
Sent: Thursday, September 04, 2014 8:16 PM
To: Jason Aarons (AM)
Cc: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] C240M3 16 drives for CUCM/UCCE


Personally, I wouldn't leave a system in that format, it's a waste of space.  
Cisco's docs seem to change because at one point, the docwiki specified that it 
could be built in either a raid 5 or raid 10 (for the app storage), but that 
has been updated.

I'd personally setup a mirror with the first two for ESXI and then a raid 10 
with the remainder drives.  You get more IOPS out of it and should have plenty 
of space.

On Thu, Sep 4, 2014 at 4:47 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:
C240M3  TRC setup
8 drives raid 5 group 1
8 drives raid 5 group 2

Seems like an odd setup to have the hypervisor running on a Raid 5 array.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/virtual/CUCM_BK_CF3D71B4_00_cucm_virtual_servers/CUCM_BK_CF3D71B4_00_cucm_virtual_servers_chapter_010.html

Do you install Virtual Machines to both Raid5 arrays, or just ESXi on one and 
the VMs on the other?

Customer asked me why Cisco would have ESXi  installed on a 1.93TB raid 5 array 
vs having two drives in Raid 1 mirror.  I told them I would ask around.


Jason Aarons
Consultant
Dimension Data
+1-904-338-3245tel:%2B1-904-338-3245


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itevomcid
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Re: [cisco-voip] CUCM 7.1.5-12009-1 direct upgrade to 10.5?

[Haas, Neal]

It's been awhile, but we had an ES 7.1.5. This ES was for a Raid driver 
issues with the MCS. We upgraded to 8.X (don't remember the exact ver.) but the 
UPGRADE reverted back to the Raid driver issue. We ended up rebuilding with a 
fresh install to each of the subs - then did the pub.

Check with TAC I believe they will tell you with an ES upgrade you should DR.


Neal Haas
IT Analyst, Communications

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Thursday, September 04, 2014 5:15 PM
To: cisco-voip (cisco-voip@puck.nether.net)
Subject: [cisco-voip] CUCM 7.1.5-12009-1 direct upgrade to 10.5?

So it appears this customer has an Engineering Special  for Callmanager 
7.1.5-12009-1 that isn't listed on the Compatibility matrix (ESs never are).

Can they direct upgrade to 10.5.1 or do I need to upgrade them to 7.1(5b) first 
then 10.5.1?

Jason Aarons
Consultant
Dimension Data
+1-904-338-3245

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Re: [cisco-voip] UCCX Wallboard Query

[Haas, Neal]

Does Const SQL_Command = select * from rtcsqssummary” Work?


Neal Haas

From: cisco-voip [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
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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Re: [cisco-voip] Emergency Communication

[Haas, Neal]

I agree with Anthony, Singlewire is the way to go, we have it here and it works 
great. Look at their 911 options.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Anthony Holloway
Sent: Wednesday, July 16, 2014 10:35 AM
To: Lisa Notarianni
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Emergency Communication

Hi Lisa,

Your email reads like you are asking about options to alert your Public Safety 
Answering Point (PSAP), or 911.

If this is the case, then the very first thing you should do it consult with 
them, to see what your options are.  I would guess that intentionally 
establishing a one way audio path to them would be shot down immediately.  
Could it be done?  Yes, I think a clever use of Trusted Relay Points and layer 
3 access lists to block the return RTP streams for these specific flows would 
do it.  The one button feature would be a speed dial.

If you meant on-site security personnel and not your PSAP, then I guess the sky 
is the limit.

CUC is the only product that I can think of that has SMS abilities, but then 
again, it doesn't send the text itself, it relies on an SMS gateway or SMTP 
relay service.  Actually, now that I say SMTP relay, UCCX can do that as well.  
The UCCX option would be easy to handle a one button (either speed dial or 
phone service) to kick off the work flow.

If you wanted to go third party, then take a look at SingleWire Informacast, 
which has a whole host of alerting options, it will make your head spin.

http://www.singlewire.com/informacast.html

The software and basic license is included with purchase of CUCM 10.  And you 
can upgrade to the advanced license with a trial period.  I saw a demo of their 
products once and it was very impressive to me, though I've not actually 
deployed their alerting system yet.  So, I cannot speak to it in any detail.

I hope that was helpful.

On Wed, Jul 16, 2014 at 11:07 AM, Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu wrote:
We have Cisco CUCM 8.6 soon to be upgraded to 10.x.  A request has been asked 
of us to research the possibility of communicating in an emergency situation to 
our Police Department in 2 ways:


1.   From our Cisco phones – possibly creating a 1 button push with one way 
communication so the Police Department can hear what is going on

2.   Texting University Police

Does anyone communicate in either of these ways for security/safety purposes?

Thank you in advance.


[LNsignatureFile]


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Re: [cisco-voip] Disappearing Agents in CAD

[Haas, Neal]

I had this issue about a year ago with a 9.x version. I uninstalled and 
re-installed supervisor desktop. Haven't had an issue since then.


Neal Haas
IT Analyst, Communications
Please report Troubles to the Help Desk. 559-600-5900
Telephone (559) 600-5890
FAX (559) 455-4747

County of Fresno
Information Technology Services Division
1020 S Tenth St
Fresno, CA 93702

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Stoian Adrian
Sent: Tuesday, July 15, 2014 1:58 AM
To: Matthew Loraditch; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Disappearing Agents in CAD

Hello,

As far as I remember this is a firewall issue.
Try disabling the firewall from the agent/supervisor PC and see if it works.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: Monday, July 14, 2014 10:43 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Disappearing Agents in CAD

Has anyone seen something like this, Agents stay online and take calls, but 
appear and disappear in Supervisor?
CCX 10.0SU1


Matthew G. Loraditch - CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518
fax.  410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
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Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830


Acest mesaj si orice fisiere sau documente atasate contin informatii 
confidentiale, clasificate conform regulilor interne ROMTELECOM. Mesajul este 
destinat doar persoanei sau entitatii adresate si altora autorizati sa-l 
primeasca. Daca dvs. nu sunteti in aceasta situatie, prin aceasta va informam 
ca orice dezvaluire, copiere, distribuire sau orice alta actiune bazata pe 
continutul acestor informatii este strict interzisa si pot fi aplicate 
sanctiuni, potrivit legii. Daca ati primit acest mesaj din greseala, va rugam 
sa ne informati imediat si sa stergeti mesajul din sistemul dvs. De asemenea, 
va rugam sa tineti cont ca transmisia nu poate fi garantata ca fiind sigura sau 
fara erori. Va multumim!
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Re: [cisco-voip] 2911 with PVDM-3 and 3x9971s and no Video Conference Homogeneous

[Haas, Neal]

Do you have the “Video Capabilities” Enabled and “RTCP for Video” Enabled on 
the Devices phone Configuration?


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Thursday, June 12, 2014 7:07 AM
To: Brian Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] 2911 with PVDM-3 and 3x9971s and no Video Conference 
Homogeneous

Yes a single Device Pool/same region.  Point to point I get my audio and h264 
video (can see it via by going to phone ip address in web browser and looking 
at stream1/stream2)

Once I hit conference it’s an audio only call.  I was expecting some sort of 
3-way video.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmevconf.html#59665

Perhaps some debug sccp all and callmanager logs.

Anyone have this working ?  I didn’t spec out a full MCU as I understood the 
PVDM3-256 could do some sort of 3-way basic video conference.


From: bmead...@gmail.commailto:bmead...@gmail.com [mailto:bmead...@gmail.com] 
On Behalf Of Brian Meade
Sent: Thursday, June 12, 2014 9:58 AM
To: Jason Aarons (AM)
Cc: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] 2911 with PVDM-3 and 3x9971s and no Video Conference 
Homogeneous


Is the conference bridge in the same region on the phones?  Can the phones 
successfully due H.264 directly between 2 endpoints?

On Wed, Jun 11, 2014 at 10:48 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.commailto:jason.aar...@dimensiondata.com wrote:
I have 3 9971s  same Device Pool, etc in Callmanager 9.1.2.  Running latest 
9971 firmware 9-4-1-9.  Running 2911 with IOS 15.1.4M2.  Video Conference is in 
its own MRG in the MRGL first in CCM.

The 2911 Homogeneous Video Confer Bridge is in use via the PVDM3-256 but not 
doing h264, why?  It’s just an audio call?


2911-ATL#show sccp | begin V_Con
V_Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.64.12, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 16, Reported Max OOS Streams: 0
Layout: default 1x1
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization 
Period: 30
Supported Codec: h264: 2.0 (vga), Frame Rate: 30fps, Bit Rate: 1000-1000 Kbps
   Max MBPS: 72 (x500 MB/s) Max FS: 5 (x256 MBs)
TLS : ENABLED

2911-ATL#show sccp conn
sess_idconn_id  stype mode codec   sport rport ripaddr conn_id_tx

33554467   33554551 vcf   sendrecv g711u   32392 25830 10.102.64.46
33554467   33554550 vcf   sendrecv g711u   22792 21108 10.102.64.48
33554467   33554549 vcf   sendrecv g711u   24078 27538 10.102.64.47

Total number of active session(s) 1, and connection(s) 3

2911-ATL#show call active video compact
2911-ATL#show call active voice compact
callID  A/O FAX Tsec Codec   typePeer Address   IP 
Rip:udp
Total call-legs: 3
   226 ORG T693   g711ulawVOIPP
10.102.64.47:27538http://10.102.64.47:27538
   228 ORG T693   g711ulawVOIPP
10.102.64.48:21108http://10.102.64.48:21108
   229 ORG T693   g711ulawVOIPP
10.102.64.46:25830http://10.102.64.46:25830


Show run | begin dspfarm profile 2 conference video homogeneous
dspfarm profile 2 conference video homogeneous
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec ilbc
codec h264 vga frame-rate 30 bitrate 1mbps
maximum sessions 4
associate application SCCP


2911-ATL#show ver | inc image
System image file is flash0:c2900-universalk9-mz.SPA.151-4.M2.bin

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itevomcid
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Re: [cisco-voip] Cisco Call Recording Solution

[Haas, Neal]

Voice Print, we use them for 911 recording and they have very good prices, very 
good support.
www.vpi-corp.com


Neal Haas

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ken 
Rhodes
Sent: Wednesday, May 28, 2014 10:02 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco Call Recording Solution

Does anyone have a good suggestion about a CUCM call recording solution? We 
currently have Zoom, however the sales and customer service has been lacking so 
we are looking at other providers.

Thanks

Sent from my iPhone
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Re: [cisco-voip] CP-7937 EOL

[Haas, Neal]

It may look like a calculator, but it's the best TI-81 that I have had in a 
long time.


1.  You can connect 2 of them together for better large room coverage.

2.  You can do wireless MIC's, for additional $$

3.  I think the sound is better (and looks better).

4.  You can move the calculator around instead of the whole phone.

5.  And the best of all, no more starship jokes. :


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Nicholas Samios
Sent: Tuesday, April 29, 2014 9:14 PM
To: cisco-v...@puck-nether.net (cisco-voip@puck.nether.net)
Subject: [cisco-voip] CP-7937 EOL

Anyone know the backstory to EOS/EOL - part of the 7900 series retirement or 
just moving away from the rebadged polycom star phones?

Really liked these phones for mid/large sized rooms - the replacement 8831 
model seems to be a few steps backwards (in design anyway) - keypad looks like 
a calculator.

http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7940g/end_of_life_notice_c51-729487.pdf

[cid:image001.jpg@01CF6442.E3872DE0]
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Re: [cisco-voip] 8 channel GSM gateways?

[Haas, Neal]

We have done something a little different, Cisco Router with Verizon Data card, 
 put cisco phones on the edge. Not your case, But could you get a SIP provider, 
would that not be cheaper solution than Cell Phones


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Erick 
Bergquist
Sent: Thursday, April 24, 2014 10:23 AM
To: Lelio Fulgenzi
Cc: Cisco VOIP
Subject: Re: [cisco-voip] 8 channel GSM gateways?

I'd be interested if you find one and go with one. I have dealt with a client 
in same country who does this with FXO ports and a mobile phone device like 
you, but haven't heard of any issues so haven't really looked into alternatives.

On Thu, Apr 24, 2014 at 11:07 AM, Lelio Fulgenzi 
le...@uoguelph.camailto:le...@uoguelph.ca wrote:
this sounds interesting.

are your mobile devices isolated to particular telephone numbers (or groups) 
that allow you to make the route patterns manageable? in North America (if you 
don't already know) with number portability, there is no such grouping.

that being said, my curiosity found this:


http://www.voip-info.org/wiki/view/VoIP+Gateways#CellularMobilePhoneGateways

the dinstar ones look neat.

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

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


From: Robert Kulagowski rkula...@gmail.commailto:rkula...@gmail.com
To: Cisco VOIP cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Sent: Thursday, April 24, 2014 11:35:44 AM
Subject: [cisco-voip] 8 channel GSM gateways?


One of our offices (in Brazil) wants to deploy 8 cell phones to make
calls to cellular destinations cheaper. (Cell provider has cheap
on-net calls)

We've had bad experiences with analog gateways in the past (disconnect
supervision problems).

Does anyone have any recommendations for an 8-port GSM gateway (either
PRI or H.323/SIP interface) which can be used in Brazil?

If we do manage to find such a box, do you just pull the SIM cards
from 8 cell phones and plug them into the box?

Thanks.
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Re: [cisco-voip] 'buzz' secretary's phone?

[Haas, Neal]

Yes InformaCast can do this.

But what we did was just add a speed dial on the Supervisors phones and then 
add a NON-DID to the secretary's phone that says Boss is calling (not really 
but tell them that any way). Now the boss has a direct line and the secretary 
has a line that she knows she needs to answer and that does not go to voicemail.


Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Erick 
Wellnitz
Sent: Monday, February 24, 2014 1:35 PM
To: cisco-voip
Subject: [cisco-voip] 'buzz' secretary's phone?

I have a small group of users who were fond of the old PBX feature where they 
could 'buzz' their secretary.  I think this was some sort of signaling key in 
the Nortel world.

Does anyone know if there is an XML service that might accomplish this?  Would 
the basic Informacast product be able to accomplish this or would we be better 
off finding some sort of buzzer box that we could crossconnect to the 
secretary's location?

Thanks in advance!

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Re: [cisco-voip] 'buzz' secretary's phone?

[Haas, Neal]

Well now, have him use email, that might be more efficient. Nortel went the way 
of telegraph. :

You might have him select a certain speed dial on his phone, that corresponds 
to a specific number on the secretaries phone, she doesn't need to pick it up 
just read what  it means on the phonemore Coffee, bring me a 
doughnut.

Good luck

Neal Haas

From: Erick Wellnitz [mailto:ewellnitzv...@gmail.com]
Sent: Monday, February 24, 2014 1:59 PM
To: Haas, Neal
Cc: cisco-voip
Subject: Re: [cisco-voip] 'buzz' secretary's phone?

We already have that set up.  The problem this user has with that is it puts 
their current call on hold.

He uses it almost like a teelegraph.  1 buzz means x.  2 buzz means y.  3 menas 
z and so on.  It's really an antiquated methodology but it does have it's 
merits.

He likes it because it doesn't interrupt a call in progress and the secretary 
doesn't need to be looking at the phone or the PC to get the 'message'.

On Mon, Feb 24, 2014 at 3:47 PM, Haas, Neal 
nh...@co.fresno.ca.usmailto:nh...@co.fresno.ca.us wrote:
Yes InformaCast can do this.

But what we did was just add a speed dial on the Supervisors phones and then 
add a NON-DID to the secretary's phone that says Boss is calling (not really 
but tell them that any way). Now the boss has a direct line and the secretary 
has a line that she knows she needs to answer and that does not go to voicemail.


Neal Haas

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.netmailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Erick Wellnitz
Sent: Monday, February 24, 2014 1:35 PM
To: cisco-voip
Subject: [cisco-voip] 'buzz' secretary's phone?

I have a small group of users who were fond of the old PBX feature where they 
could 'buzz' their secretary.  I think this was some sort of signaling key in 
the Nortel world.

Does anyone know if there is an XML service that might accomplish this?  Would 
the basic Informacast product be able to accomplish this or would we be better 
off finding some sort of buzzer box that we could crossconnect to the 
secretary's location?

Thanks in advance!


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