Re: [cisco-voip] Cisco's Website and Find on Page (CTRL+F) Issue

[Evgeny Izetov]

Reader View (F9) in Firefox seems to be ok as a workaround until they fix
it.

On Tue, Jan 21, 2020 at 2:12 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Yes, non-visible text is correct, but the code itself is incorrect.  For
> example, if you CTRL+F for body, which is a tag in HTML, there will be no
> results on the page I linked prior.
>
> Also, I tried it in Chrome, FireFox, IE and Edge, all with the same
> problem.
>
> On Tue, Jan 21, 2020 at 12:36 PM Lelio Fulgenzi  wrote:
>
>> I think it has to do with results in non-visible text, i.e. the code
>> itself.
>>
>>
>>
>> But, yes, I have experienced this. Sometimes, using a different browser
>> helps.
>>
>>
>>
>>
>>
>> *From:* cisco-voip  *On Behalf Of 
>> *Anthony
>> Holloway
>> *Sent:* Tuesday, January 21, 2020 11:57 AM
>> *To:* Cisco VoIP Group 
>> *Subject:* [cisco-voip] Cisco's Website and Find on Page (CTRL+F) Issue
>>
>>
>>
>> What's the problem?
>>
>>
>>
>> Go to this page:
>>
>>
>> https://www.cisco.com/c/en/us/support/docs/cloud-systems-management/smart-call-home/119144-config-sch-00.html
>>
>>
>>
>> Then CTRL+F for the search term: cert
>>
>>
>>
>> There should be 21 matches, and not a single one of them are actually
>> rendered on the page.
>>
>>
>>
>> I was trying to figure out which certs on CUCM are for Smart Call Home,
>> because we (and some of you too) have a VeriSign cert expiring next month,
>> and I wanted to double check that it was for smart call home.
>>
>>
>>
>> Anyway, the post is about the website functionality, and not smart call
>> home, you could easily run into this issue on any page and with any search
>> term.  If you're using Chrome, you may have experienced the issue where you
>> can see that there are matches, but you cannot cycle through to any of
>> them; it appears suck.
>>
>>
>>
>> Does anyone else deal with this?  If so, let's get it fixed.
>>
>>
>>
>> I already submitted this via the feedback link at the bottom of the page,
>> but if one of you knows how to work around this issue, that would be great.
>>
>>
>>
>> Through some basic web skills, I was able to determine that the search
>> terms are in the source code for the page, but inside of elements which are
>> hidden via CSS; like the drop down menus.
>>
>>
>>
>> Also, if you can confirm the issue, and then submit the same feedback,
>> maybe through sheer volume, they'll get their web team to fix it sooner
>> rather than later.
>>
>>
>>
>> You just need to click the Feedback link at the very bottom of the page,
>> and I'll even paste some copy here for you, just in case you didn't want to
>> write your own:
>>
>>
>>
>> ---
>>
>>
>>
>> Hello Web Team!
>>
>>
>>
>> When I and several others perform research in the course of performing
>> our job duties, we are often in need of searching within your documentation
>> by way of using the browser's find on page feature (CTRL+F).
>>
>>
>>
>> However, due to the way in which your web page is structured, we are
>> hindered from finding valuable content, because non-visible content is
>> being searched; such as drop down menus.
>>
>>
>>
>> Example: Navigate to:
>>
>>
>>
>>
>> https://www.cisco.com/c/en/us/support/docs/cloud-systems-management/smart-call-home/119144-config-sch-00.html
>>
>>
>>
>> Press CTRL+F, followed by typing: cert
>>
>>
>>
>> You should see that there are 21 matches, but none of them appear
>> anywhere on the page.  That is just one such example, and this happens on
>> many, if not all of your pages.
>>
>>
>>
>> Your consideration for this issue is greatly appreciated.  Thank you.
>>
>>
>>
>> ---
>>
>>
>>
>> Thanks Cisco VoIPers!
>>
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Re: [cisco-voip] Bulk Admin and Jabber

[Evgeny Izetov]

You can use Import/Export configuration with all details. No need for
template and can import different types of devices at the same time.

On Thu, Sep 12, 2019 at 8:24 PM Lelio Fulgenzi  wrote:

>
> We really need to start to minimize the amount of custom scripts /
> programming that we use, not increase it.
>
> It’s a bit of a long story.
>
> So I need to use the built-in tools.
>
> :(
>
> We might just stick with super copy if bulk admin doesn’t work out.
>
> Lelio
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On Sep 12, 2019, at 7:48 PM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> Why not roll your own build tool in Python?  You can ask questions here,
> as lots of us have Python and AXL experience, and then more people get to
> benefit from the collaboration.
>
> Just think how cool it would be to build your very own Jarvis:
>
> 
>
> On Thu, Sep 12, 2019 at 5:22 PM Lelio Fulgenzi  wrote:
>
>>
>> It’s been a while since I’ve used Bulk Admin Tool.
>>
>> Wondering how it might help with our Jabber deployment.
>>
>> I’m thinking, since our requests will come in one at a time, the best
>> approach would be to create a BAT file that has a line for each device
>> type. We’ve decided to load all types so users can switch devices out/in as
>> they wish.
>>
>> Can you bulk load _without_ using a template?
>>
>> What have others done with respect to Jabber bulk loads?
>>
>> Lelio
>>
>> *-sent from mobile device-*
>>
>>
>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>
>> Computing and Communications Services | University of Guelph
>>
>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>> N1G 2W1
>>
>> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>>
>>
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>
>>
>>
>> [image: University of Guelph Cornerstone with Improve Life tagline]
>> ___
>> cisco-voip mailing list
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
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Re: [cisco-voip] Change last Redirect DN in ISR 4451

[Evgeny Izetov]

This will be a diversion header in CUBE. You can change it with "translate
redirect-called" under "voice translation-profile". This is assuming only
one Diversion Header. If the call was forwarded multiple times then there
will be multiple diversion headers, and I'm not sure if the translation
applies to first, last, or all of them.

Make sure to enable Redirecting Diversion Header Delivery - Outbound on the
SIP trunk to CUBE.

On Tue, Jul 30, 2019 at 4:46 PM Reto Gassmann  wrote:

> Hallo Group
>
> Is there a way to change the last Redirect DN Field on a ISR 4451 in the
> SIP Profile?
>
> We have a CUCM 10.5 and UCCE 11.6 and need this field to identify calls
> from the UCCE to CUCM for Voice Recording.
> Now all the Fields (originalcalledpartynumber, finalcalledpartynumber,
> lastredirectDn) I can use for Voice Recording to create a Recording rule
> are the same Value (DN of Agent). I try to change the lastredirectDn to set
> a filter on this number to only record these calls.
>
> Thanks for help
> Regards Reto
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Re: [cisco-voip] Unity Connection Mailbox

[Evgeny Izetov]

That parameter would work but it's global and can break other things. You
could maybe apply a LUA script on the SIP trunk that has the logic for that
exact forwarding scenario...

On Fri, Jul 26, 2019 at 1:20 PM Lelio Fulgenzi  wrote:

> Ok – I’m _*pretty*_ sure this is what helps:
>
>
>
>- Connection Admin > Advanced > Conversations
>- Use Last (Rather than First) Redirecting Number for Routing Incoming
>Call [ ]
>
>
>
> “When this check box is checked, Unity Connection uses the last
> redirecting number for routing incoming calls.”
>
>
>
> I have not (ever) had a chance to test this.
>
>
>
>
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Lelio
> Fulgenzi
> *Sent:* Friday, July 26, 2019 1:10 PM
> *To:* Jason Aarons (Americas) ;
> cisco-voip (cisco-voip@puck.nether.net) 
> *Subject:* Re: [cisco-voip] Unity Connection Mailbox
>
>
>
> There used to be a system wide settings on Unity Connection where you
> could decide to either send it to the originating mailbox. Not sure if it
> was available in the admin GUI or it needed a cisco unity monkey tool to do
> it.
>
>
>
> Will spend a few minutes looking…
>
>
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Jason
> Aarons (Americas)
> *Sent:* Friday, July 26, 2019 12:43 PM
> *To:* cisco-voip (cisco-voip@puck.nether.net) 
> *Subject:* [cisco-voip] Unity Connection Mailbox
>
>
>
>
>
> Jack on PSTN  >   Diane ext 4001  >>   Bobbie ext 4002
>
>
>
> If Jack calls Diane and Diane has call forward all to Bobbie, is there a
> way for Jack to hear Bobbie’s voicemail and not Diane’s?
>
>
>
> TAC said no.  Ask the expert at Cisco Live said no.  is the answer really
> no?
>
>
>
>
>
>
>
>
>
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
> 
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Re: [cisco-voip] How difficult is hybrid call (connector) setup ?

[Evgeny Izetov]

I would highly recommend watching this CiscoLive session: "Cisco Webex
Hybrid Services (formerly known as Spark Hybrid Services) Call Service
Connect Architecture and Design - BRKCOL-2202"

Luca does a great job explaining how it all works.

On Tue, Jul 2, 2019 at 10:39 PM Lelio Fulgenzi  wrote:

> Ok. Thanks. That’s putting things into perspective now.
>
> I’ve got a budget request in for additional compute resources. This now
> puts a solid reason behind it.
>
> And I agree. There really should be an easy way to split and isolate the
> communications paths on the same cluster.
>
> I don’t look forward to managing an additional four servers.
>
>
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On Jul 2, 2019, at 10:25 PM, Ryan Huff  wrote:
>
> It all comes down to what you are using your Expressways for. In a Webex
> Hybrid call connect scenario you could in theory, have calls with source
> uri’s using your domain, which in a pure B2B scenario is usually something
> you’d deny with a call policy.
>
> As I understand it, the dedicated Expressway recommendation (outside of
> pure capacity reasons) is to make it easier to write call policies that
> don’t interfere with other use cases since Expressway doesn’t really have a
> partitioning mechanism (outside of what you can do with search rules).
>
> You really have to go to the Ninja master level with your regular
> expressions in your search rules and call polices to get multiple use cases
> setup and using call policies to reduce toll fraud... and have everything
> work.
>
> ... and this is where Cisco should, in my opinion, step up to the plate a
> little. The best answer really shouldn’t be, “just deploy and use another
> Expressway”.
>
> Yes, that is easier than refining CPL and search rules, but many customers
> run tight compute/storage budgets (Ex. be6k) and cant always spin up more
> Expressways.
>
> -R
>
> On Jul 2, 2019, at 22:06, Lelio Fulgenzi  wrote:
>
> Ok. Thanks.
>
> I think we’re on a low version of expressway. I’ll have to confirm.
>
> My memory is failing, and I can’t find it in my notes, but in one session
> they talked about using a separate pair of C’s and E’s or have a high risk
> of toll fraud.
>
> I believe it was in my Sunday techtorial. I’ll have to reach out to them.
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs
> 
>  |
> @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On Jul 2, 2019, at 9:49 PM, Ryan Huff  wrote:
>
> It’s not that bad (maybe 1 - 2 hours to get a functional test going).
>
> The most unexpected thing I think you may run into is the use/need for
> MTLS (TCP:5062) between the Edge and Control Hub. Also, your Expressway
> version should be 12.5.x (I think 8.11.4 may still work but you’ll get an
> alarm telling you to upgrade if it does work).
>
> Outside of that, it’s a splash of nerd knob turning in the control hub,
> some search rule / traversal & dns zone magic in Expressway C/E and setting
> up the management/call connector in Expressway, you can even re-use a MRA
> (unified communications) traversal client/server (or create a dedicated
> traversal if so inclined).
>
> Here is the guide you’d want to follow for it and it’s pretty complete and
> well written:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html
> 
>
> Thanks,
>
> Ryan
>
> On Jul 2, 2019, at 21:30, Lelio Fulgenzi  wrote:
>
>
> On a scale from 1 to 10, how difficult and/or time consuming is it to
> setup call hybrid connector with WebEx?
>
> I’m just 

Re: [cisco-voip] Multiple native CI/CH WebEx sites in same control hub org

[Evgeny Izetov]

There's this dropdown option on hybrid calendar configuration:
[image: image.png]

And a  "proof" :-)
[image: image.png]

On Thu, May 23, 2019 at 7:30 PM Lelio Fulgenzi  wrote:

>
> Can you share a (redacted) screen shot?
>
> In this case, if hybrid calendar is enabled, which site does it use for
> @webex personal room meetings?
>
> Lelio
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On May 23, 2019, at 7:21 PM, Evgeny Izetov  wrote:
>
> Yes, you can have multiple Control Hub managed Webex sites under one
> Control Hub.
>
> On Thu, May 23, 2019 at 7:09 PM Lelio Fulgenzi  wrote:
>
>>
>> Can anyone confirm they have a control hub org with multiple control hub
>> / common identity managed Webex sites enabled?
>>
>> I’m not talking linked sites. These are WebEx sites that were created
>> from control hub as control hub managed from the get go.
>>
>> There is some scuttlebutt that says this is not possible and it has me
>> concerned.
>>
>> Lelio
>>
>>
>> *-sent from mobile device-*
>>
>>
>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>
>> Computing and Communications Services | University of Guelph
>>
>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>> N1G 2W1
>>
>> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>>
>>
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>
>>
>>
>> [image: University of Guelph Cornerstone with Improve Life tagline]
>> ___
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>>
>
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Re: [cisco-voip] Multiple native CI/CH WebEx sites in same control hub org

[Evgeny Izetov]

Yes, you can have multiple Control Hub managed Webex sites under one
Control Hub.

On Thu, May 23, 2019 at 7:09 PM Lelio Fulgenzi  wrote:

>
> Can anyone confirm they have a control hub org with multiple control hub /
> common identity managed Webex sites enabled?
>
> I’m not talking linked sites. These are WebEx sites that were created from
> control hub as control hub managed from the get go.
>
> There is some scuttlebutt that says this is not possible and it has me
> concerned.
>
> Lelio
>
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
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Re: [cisco-voip] PUT Tool Bootables - what version?

[Evgeny Izetov]

I wonder if TAC also gave up - UltraISO'd it themselves and forgot to add
Bootable_ :-)

On Wed, May 15, 2019 at 3:46 PM Lelio Fulgenzi  wrote:

> I remember when it used to as simple as “format /s”
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On May 15, 2019, at 3:22 PM, Charles Goldsmith 
> wrote:
>
> It's not.  And just in case they changed things, I went and downloaded the
> latest 12.0 and 12.5 of both CUCM and CUC and none of them have the
> bootable part of the ISO.
>
> Simply renaming a file doesn't make it bootable :)
>
>
> On Wed, May 15, 2019 at 1:36 PM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
>> That.  Can't.  Be.  True.  Right?  If so, Brian Meade has been wasting
>> his time with UltraISO.
>>
>> On Wed, May 15, 2019 at 1:26 PM Evgeny Izetov  wrote:
>>
>>> That's good to know. Was it 12.x or 11.x?
>>>
>>> On Wed, May 15, 2019 at 2:19 PM Haas, Neal 
>>> wrote:
>>>
>>>> I had a TAC Call last week, they told me to add BOOTABLE to the name
>>>> (in front) and that was it. They said all ISO’s are now bootable with the
>>>> name change…..
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *From:* cisco-voip  *On Behalf Of 
>>>> *Evgeny
>>>> Izetov
>>>> *Sent:* Wednesday, May 15, 2019 11:17 AM
>>>> *To:* Lelio Fulgenzi 
>>>> *Cc:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
>>>> cisco-voip@puck.nether.net>
>>>> *Subject:* Re: [cisco-voip] PUT Tool Bootables - what version?
>>>>
>>>>
>>>>
>>>> Yeah, CUPS has always been bootable.. CUCM/CUC/CER are still not
>>>>
>>>>
>>>>
>>>> So, what is the proper way to obtaining bootable iso's now? Let's say a
>>>> CUCM 11.5 SU6 needs to be reinstalled, and there's no bootable because it
>>>> was upgraded from an earlier SU. PUT does not have bootable SU6 and neither
>>>> does Enterprise Agreement. Is TAC the only way to get the bootable for a
>>>> specific SU? I believe there used to be a time when everyone was advised
>>>> that TAC is not able to provide bootables?
>>>>
>>>>
>>>>
>>>> On Wed, May 15, 2019 at 12:18 PM Lelio Fulgenzi 
>>>> wrote:
>>>>
>>>>
>>>>
>>>> Same with CUPS if I’m not mistaken.
>>>>
>>>>
>>>>
>>>> ---
>>>>
>>>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>>>
>>>> Computing and Communications Services | University of Guelph
>>>>
>>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>>>> N1G 2W1
>>>>
>>>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>>>
>>>>
>>>>
>>>> www.uoguelph.ca/ccs
>>>> <https://gcc01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7Cnhaas%40fresnocountyca.gov%7Cabd25657699a414cd03a08d6d9618de6%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636935410356509995=jjznrZifNJrZ2fihXdwSkDj6b0FIj9VjvKtFFpDLRDM%3D=0>
>>>> | @UofGCCS on Instagram, Twitter and Facebook
>>>>
>>>>
>>>>
>>>> [image: University of Guelph Cornerstone with Improve Life tagline]
>>>>
>>>>
>>>>
>>>> *From:* cisco-voip  *On Behalf Of 
>>>> *Charles
>>>> Goldsmith
>>>> *Sent:* Wednesday, May 15, 2019 12:09 PM
>>>> *To:* Evgeny Izetov 
>>>> *Cc:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
>>>> cisco-voip@puck.nether.net>
>>>> *Subject:* Re: [cisco-voip] PUT Tool Bootables - what version?
>>>>
>>>>
>>>>
>>>> Plus, UCCX is shipping bootables (filename doesn't reflect it).
>>>>
>>>>
>>>>
>>>> Description :
>>>>
>

Re: [cisco-voip] PUT Tool Bootables - what version?

[Evgeny Izetov]

That's good to know. Was it 12.x or 11.x?

On Wed, May 15, 2019 at 2:19 PM Haas, Neal  wrote:

> I had a TAC Call last week, they told me to add BOOTABLE to the name (in
> front) and that was it. They said all ISO’s are now bootable with the name
> change…..
>
>
>
>
>
>
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Evgeny
> Izetov
> *Sent:* Wednesday, May 15, 2019 11:17 AM
> *To:* Lelio Fulgenzi 
> *Cc:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] PUT Tool Bootables - what version?
>
>
>
> Yeah, CUPS has always been bootable.. CUCM/CUC/CER are still not
>
>
>
> So, what is the proper way to obtaining bootable iso's now? Let's say a
> CUCM 11.5 SU6 needs to be reinstalled, and there's no bootable because it
> was upgraded from an earlier SU. PUT does not have bootable SU6 and neither
> does Enterprise Agreement. Is TAC the only way to get the bootable for a
> specific SU? I believe there used to be a time when everyone was advised
> that TAC is not able to provide bootables?
>
>
>
> On Wed, May 15, 2019 at 12:18 PM Lelio Fulgenzi  wrote:
>
>
>
> Same with CUPS if I’m not mistaken.
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs
> <https://gcc01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7Cnhaas%40fresnocountyca.gov%7Cabd25657699a414cd03a08d6d9618de6%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636935410356509995=jjznrZifNJrZ2fihXdwSkDj6b0FIj9VjvKtFFpDLRDM%3D=0>
> | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Charles
> Goldsmith
> *Sent:* Wednesday, May 15, 2019 12:09 PM
> *To:* Evgeny Izetov 
> *Cc:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] PUT Tool Bootables - what version?
>
>
>
> Plus, UCCX is shipping bootables (filename doesn't reflect it).
>
>
>
> Description :
>
> UCCX 12.0(1) image for fresh install and upgrades.
>
> UCSInstall_UCCX_12_0_1_UCOS_12.0.1.1-24.sgn.iso
>
>
>
>
>
> On Wed, May 15, 2019 at 11:04 AM Evgeny Izetov  wrote:
>
> Wasn't their excuse with not providing bootables that it was based on Red
> Hat? It's CentOS now, and still a struggle..
>
>
>
> On Wed, May 15, 2019 at 11:52 AM Brian Meade  wrote:
>
> I've given up on trying to get bootables.  I haven't had any issues with
> ones made with UltraISO.
>
>
>
> On Wed, May 15, 2019 at 11:39 AM Lelio Fulgenzi  wrote:
>
>
> Just wondering what the Put Tool Bootables are at now? We're planning on
> upgrading to v11.5.1 SU6 due to the field notice and I'd like to have the
> bootable available.
>
> Otherwise it's opening a case with the TAC, etc.
>
> Is it just a matter of submit request and check the filename?
>
> Lelio
>
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca<mailto:le...@uoguelph.ca>
>
> www.uoguelph.ca/ccs
> <https://gcc01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7Cnhaas%40fresnocountyca.gov%7Cabd25657699a414cd03a08d6d9618de6%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636935410356520005=CBMCBtLvabKho%2BCOU5cOazJiBpzZd1eFCZGy2skvifo%3D=0>
> <http://www.uoguelph.ca/ccs
> <https://gcc01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7Cnhaas%40fresnocountyca.gov%7Cabd25657699a414cd03a08d6d9618de6%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636935410356520005=CBMCBtLvabKho%2BCOU5cOazJiBpzZd1eFCZGy2skvifo%3D=0>>
> | @UofGCCS on Instagram, Twitter and Facebook
>
> [University of Guelph Cornerstone with Improve Life tagline]
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> <https://gcc01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip=02%7C01%7Cnhaas%40fresnocountyca.gov%7Cabd25657699a414cd03a08d6d9618de6%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636935410356530009=o85mUFnTglRjSv3cQem5AYirMqUI%2BawatlFqIxnUeYA%3D

Re: [cisco-voip] PUT Tool Bootables - what version?

[Evgeny Izetov]

Yeah, CUPS has always been bootable.. CUCM/CUC/CER are still not

So, what is the proper way to obtaining bootable iso's now? Let's say a
CUCM 11.5 SU6 needs to be reinstalled, and there's no bootable because it
was upgraded from an earlier SU. PUT does not have bootable SU6 and neither
does Enterprise Agreement. Is TAC the only way to get the bootable for a
specific SU? I believe there used to be a time when everyone was advised
that TAC is not able to provide bootables?

On Wed, May 15, 2019 at 12:18 PM Lelio Fulgenzi  wrote:

>
>
> Same with CUPS if I’m not mistaken.
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Charles
> Goldsmith
> *Sent:* Wednesday, May 15, 2019 12:09 PM
> *To:* Evgeny Izetov 
> *Cc:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] PUT Tool Bootables - what version?
>
>
>
> Plus, UCCX is shipping bootables (filename doesn't reflect it).
>
>
>
> Description :
>
> UCCX 12.0(1) image for fresh install and upgrades.
>
> UCSInstall_UCCX_12_0_1_UCOS_12.0.1.1-24.sgn.iso
>
>
>
>
>
> On Wed, May 15, 2019 at 11:04 AM Evgeny Izetov  wrote:
>
> Wasn't their excuse with not providing bootables that it was based on Red
> Hat? It's CentOS now, and still a struggle..
>
>
>
> On Wed, May 15, 2019 at 11:52 AM Brian Meade  wrote:
>
> I've given up on trying to get bootables.  I haven't had any issues with
> ones made with UltraISO.
>
>
>
> On Wed, May 15, 2019 at 11:39 AM Lelio Fulgenzi  wrote:
>
>
> Just wondering what the Put Tool Bootables are at now? We're planning on
> upgrading to v11.5.1 SU6 due to the field notice and I'd like to have the
> bootable available.
>
> Otherwise it's opening a case with the TAC, etc.
>
> Is it just a matter of submit request and check the filename?
>
> Lelio
>
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca<mailto:le...@uoguelph.ca>
>
> www.uoguelph.ca/ccs<http://www.uoguelph.ca/ccs> | @UofGCCS on Instagram,
> Twitter and Facebook
>
> [University of Guelph Cornerstone with Improve Life tagline]
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] PUT Tool Bootables - what version?

[Evgeny Izetov]

Wasn't their excuse with not providing bootables that it was based on Red
Hat? It's CentOS now, and still a struggle..

On Wed, May 15, 2019 at 11:52 AM Brian Meade  wrote:

> I've given up on trying to get bootables.  I haven't had any issues with
> ones made with UltraISO.
>
> On Wed, May 15, 2019 at 11:39 AM Lelio Fulgenzi  wrote:
>
>>
>> Just wondering what the Put Tool Bootables are at now? We're planning on
>> upgrading to v11.5.1 SU6 due to the field notice and I'd like to have the
>> bootable available.
>>
>> Otherwise it's opening a case with the TAC, etc.
>>
>> Is it just a matter of submit request and check the filename?
>>
>> Lelio
>>
>> ---
>> Lelio Fulgenzi, B.A. | Senior Analyst
>> Computing and Communications Services | University of Guelph
>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>> N1G 2W1
>> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram,
>> Twitter and Facebook
>>
>> [University of Guelph Cornerstone with Improve Life tagline]
>>
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Re: [cisco-voip] External call attempts to Expressway E

[Evgeny Izetov]

Check "Expressway Policy Protection" in BRKCOL-2018, it has a good example
for CPL.

But yeah, the best that Expressway can do is to respond with Forbidden to
fraud call attempts instead of Not Found. I wish we could make it
'invisible' like CUBE can be, but per that CiscoLive you'd need to have
some sort of an IPS device in front of it.

On Mon, Apr 29, 2019 at 1:23 PM Brian Meade  wrote:

> Yea, they're hoping you allow inbound calls with source of @yourdomain.com
> or @populardomainhere.com.
>
> Usually they're not targeting Expressways on purpose, they just see it as
> a device responding to SIP messages.  A lot of PBX's on the internet have
> very little security on them so things like changing the source domain will
> work.
>
> I try to do TLS only where possible but TCP is usually needed as well.  I
> always disable UDP SIP on the Expressways which cuts down a little bit.
>
> Then from there, you have to allow more based on destination than source
> if you want to allow B2B calls from any potential source.
>
> On Mon, Apr 29, 2019 at 1:14 PM Pawlowski, Adam  wrote:
>
>> All,
>>
>>
>>
>> I know I’d asked here and elsewhere in the past regarding spam calls and
>> call setup attempts, which seem to be part of the reality of being on the
>> public internet. We see consistent call attempts from our own domain, as
>> well as @google.com . More lately, I see them pop up with the from
>> address of what appears to be another customer’s Expressway E. Not many,
>> but a few.
>>
>>
>>
>> When I set up CPL on our appliances I had referred initially to this blog
>> post:
>>
>>
>>
>>
>> https://ciscoshizzle.blogspot.com/2016/05/hardening-your-cisco-vcs-expressway.html
>>
>>
>>
>> Expressway was new to me and the documentation was (is) not such that you
>> could simply open it and understand how to set it all up end to end without
>> going through the process as the tasks are sort of split between documents.
>> I wanted to note that in this blog they mention that they don’t make any
>> attempt to block routing externally, such that you wouldn’t necessarily
>> care to block calls from the default zone back out across DNS because they
>> weren’t coming to your enterprise. I am assuming that it is possible to
>> configure your search rules to allow this to happen.
>>
>>
>>
>> I don’t understand the point of this, other than perhaps you could
>> attempt calls through known hosts in case they happened to have some sort
>> of trust relationship running, or to try and skirt (or poison) blacklists.
>>
>>
>>
>> Is anyone else seeing that type of call attempt? Do you think it’s worth
>> trying to reach out to groups that appear to be proxying these calls?
>>
>>
>>
>> Best,
>>
>>
>>
>> Adam Pawlowski
>>
>> SUNYAB NCS
>>
>>
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> ___
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>
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Re: [cisco-voip] ESXi 6.5U2 - Edit VM Settings - VM Shuts Down

[Evgeny Izetov]

I also delete starter VMs. The issue was with the freshly deployed OVAs.

On Fri, Apr 12, 2019 at 1:22 PM Schlotterer, Tommy <
tschlotte...@presidio.com> wrote:

> I have done 3 or 4 build outs on ESXi 6.5U2 and I haven’t encounter this
> before.
>
>
>
> I do delete all of the starter VMs that come preinstalled and deploy all
> my own OVA’s so if you use the preinstalled VM’s it could be related to
> that.
>
>
>
> I am curious what you find.
>
>
>
> Thanks
>
> Tommy
>
>
>
> *From:* cisco-voip  *On Behalf Of *Anthony
> Holloway
> *Sent:* Friday, April 12, 2019 8:47 AM
> *To:* Cisco VoIP Group 
> *Subject:* Re: [cisco-voip] ESXi 6.5U2 - Edit VM Settings - VM Shuts Down
>
>
>
> *EXTERNAL EMAIL*
>
>
>
>
>
> I should have mentioned that the unregistering/re-registering of the VM
> does allow me to power on the machine.  Prior to that, the VM will not
> power on.
>
>
>
> On Fri, Apr 12, 2019 at 7:42 AM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> Twice now, on two separate systems, as I was doing my fresh installs of
> CSR 11 apps, I disconnect the bootable ISOs from the VMs, and ESXi throws
> the following Event Log error, followed by shutting down the VM.
>
>
>
> *Issue detected for  on  in ha-datacenter:
> Configuration file has unexpected changes. Virtual machine has been
> terminated. Unregister and re-register virtual machine if the configuration
> file is correct.*
>
>
>
> Granted, both times it happened to me, was during the build out, so at
> least it didn't happen to me during production.  So, beware, if you are
> editing VM settings in the middle of the day, even just changing the DVD
> setting, ESXi 6.5U2 will shutdown your VM.
>
>
> Tommy Schlotterer | Systems Engineer - Collaboration
> Presidio (NASDAQ: PSDO) | presidio.com
> 20 N Saint Clair 3rd Floor, Toledo, OH 43604
> D: 419.214.1415 | C: 419.706.0259 | tschlotte...@presidio.com
>
>
>
>
> *This message w/attachments (message) is intended solely for the use of
> the intended recipient(s) and may contain information that is privileged,
> confidential or proprietary. If you are not an intended recipient, please
> notify the sender, and then please delete and destroy all copies and
> attachments. Please be advised that any review or dissemination of, or the
> taking of any action in reliance on, the information contained in or
> attached to this message is prohibited.*
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Re: [cisco-voip] ESXi 6.5U2 - Edit VM Settings - VM Shuts Down

[Evgeny Izetov]

I had the same experience. It was ESXi 6.5 SU2. It always happened when I
tried to disconnect an ISO from DVD and connect to local host (but didn't
happen when just switching ISOs in datastore). ESXi powers off the VM after
a few seconds and the VM won't boot until unregister/reregister. It seems
to be ok if you shut down the VM first before disconnecting DVD, at least
don't have to unregister/reregister.

On Fri, Apr 12, 2019 at 8:49 AM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I should have mentioned that the unregistering/re-registering of the VM
> does allow me to power on the machine.  Prior to that, the VM will not
> power on.
>
> On Fri, Apr 12, 2019 at 7:42 AM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
>> Twice now, on two separate systems, as I was doing my fresh installs of
>> CSR 11 apps, I disconnect the bootable ISOs from the VMs, and ESXi throws
>> the following Event Log error, followed by shutting down the VM.
>>
>> *Issue detected for  on  in ha-datacenter:
>> Configuration file has unexpected changes. Virtual machine has been
>> terminated. Unregister and re-register virtual machine if the configuration
>> file is correct.*
>>
>> Granted, both times it happened to me, was during the build out, so at
>> least it didn't happen to me during production.  So, beware, if you are
>> editing VM settings in the middle of the day, even just changing the DVD
>> setting, ESXi 6.5U2 will shutdown your VM.
>>
> ___
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>
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Re: [cisco-voip] Cisco TSP Driver & windows 10 (CUCM 11.5)

[Evgeny Izetov]

I installed TSP on my Windows 10 machine recently to demo Attendant
Console. I didn't have any issues with installation or anything.

On Mon, Apr 8, 2019 at 9:34 AM James Dust 
wrote:

> Hi Tommy,
>
>
>
> I am trying to get a piece of software called ‘Identapop’ working with a
> windows 10 machine.
>
>
>
> I am encountering problems though and found this article, which led me to
> believe there might be a general problem deploying TAPI with windows 10:
>
>
>
>
> https://community.cisco.com/t5/call-control/tapi-for-windows-10-x64/td-p/3569914
>
>
>
> perhaps the issue is with the software I am trying to deploy, rather than
> the TAPI driver.
>
>
>
> Thanks for your help.
>
>
>
> James
>
>
>
>
>
>
>
> *From:* Schlotterer, Tommy [mailto:tschlotte...@presidio.com]
> *Sent:* 08 April 2019 14:17
> *To:* James Dust; cisco-voip@puck.nether.net
> *Subject:* RE: Cisco TSP Driver & windows 10 (CUCM 11.5)
>
>
>
> What seems to be the problem?
>
>
>
> CUCM TAPI has had 64bit support since 8.5.
>
>
>
> https://developer.cisco.com/site/tapi/documents/supported-windows-os/
>
>
>
>
>
> Thanks
>
>
> Tommy
>
>
>
>
>
>
>
> * Tommy Schlotterer | Systems Engineer - Collaboration Presidio (NASDAQ:
> PSDO) | presidio.com  20 N Saint Clair 3rd Floor,
> Toledo, OH 43604 D: 419.214.1415 | C: 419.706.0259 |
> tschlotte...@presidio.com *
>
>
>
> *From:* cisco-voip  *On Behalf Of *James
> Dust
> *Sent:* Monday, April 8, 2019 2:27 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Cisco TSP Driver & windows 10 (CUCM 11.5)
>
>
>
> *EXTERNAL EMAIL*
>
>
>
>
>
> Morning all,
>
>
>
> Apologies if this has been touched on before, I just wondered if anyone
> else has a work around for windows 10 x 64 and using the TSP/TAPI driver.
>
>
>
> We have cucm 11.5 and are trying to get a piece of desktop software which
> pops an outlook contact working with a windows 10 install.
>
>
>
> Any help appreciated.
>
>
>
> James
>
>
> *Consider the environment - Think before you print*
>
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Re: [cisco-voip] Application Dial Rule with variable length

[Evgeny Izetov]

Also, even if click-to-dial numbers are not in +E.164 format, I would
probably try to 'globalize' them using translation patterns instead of
ADRs, if at all possible. ADRs are global and affect things that are not
very obvious that they affect them. I remember spending quite a bit of time
troubleshooting Single Number Reach until I figured out that ADRs also
affect SNR (or at least it was in 10.5)

On Thu, Apr 4, 2019 at 12:27 AM Evgeny Izetov  wrote:

> Yes, ideally with the proper +E.164 dial plan, you'd route based on '+'
> route patterns, and then modify the called number with access codes and
> whatnot as the call exists to service provider, i.e. using translations on
> CUBE/VG. I see more and more companies using numbers in +E.164 format in
> Outlook, so click-to-dial is really seamless and does not need any ADRs.
>
> On Wed, Apr 3, 2019 at 11:51 PM Lelio Fulgenzi  wrote:
>
>>
>> Ok. Now I’m intrigued.
>>
>> My goal was to allow people to click on someone’s telephone number that
>> is written with only country codes and have Jabber append the appropriate
>> access code and international access.
>>
>> All this without having to worry about their actual long distance access.
>> If they don’t have the access, the call doesn’t go through.
>>
>> I always thought that was what ADRs were for.
>>
>> Was there another way to accomplish this?
>>
>> Are we talking adding route patterns with the ‘+’ in it?
>>
>>
>>
>> *-sent from mobile device-*
>>
>>
>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>
>> Computing and Communications Services | University of Guelph
>>
>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>> N1G 2W1
>>
>> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>>
>>
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>
>>
>>
>> [image: University of Guelph Cornerstone with Improve Life tagline]
>>
>> On Apr 3, 2019, at 10:59 PM, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>> Huh, I have never noticed this before. I just checked a few different
>> environments I have access to, and none of them address international
>> patterns.  Good catch though.  It seems like you already know it, and Lelio
>> confirmed the solution.  Though, I'd argue that Lelio's configuration is
>> backwards; you should be globalizing the number, not localizing it, but
>> that's none of my business. ;)
>>
>> Could you just avoid using these rules all together and implement your
>> dialing habit support in the CUCM dial plan?  E.g., xlates and xforms?
>>
>> On Wed, Apr 3, 2019 at 7:51 AM Reto Gassmann  wrote:
>>
>>> Hallo group
>>>
>>> I have to configure Application Dial Rules on a CUCM 10.5 to prefix
>>> numbers for Click to call (eg Firefox, Chrome,...) with Jabber 12.5.
>>> Our national numbers are all 10 digits long. So one ADR is enough.
>>> But how about international numbers. They all differ in lenght.
>>> It is true to build a ADR for every length? So that would result in many
>>> rules from 9 to 20 or more Number of Digits.
>>>
>>> Any other ideas, how I could handle the Click 2 Call issue on jabber?
>>> Regard Reto
>>> ___
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
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Re: [cisco-voip] Application Dial Rule with variable length

[Evgeny Izetov]

Yes, ideally with the proper +E.164 dial plan, you'd route based on '+'
route patterns, and then modify the called number with access codes and
whatnot as the call exists to service provider, i.e. using translations on
CUBE/VG. I see more and more companies using numbers in +E.164 format in
Outlook, so click-to-dial is really seamless and does not need any ADRs.

On Wed, Apr 3, 2019 at 11:51 PM Lelio Fulgenzi  wrote:

>
> Ok. Now I’m intrigued.
>
> My goal was to allow people to click on someone’s telephone number that is
> written with only country codes and have Jabber append the appropriate
> access code and international access.
>
> All this without having to worry about their actual long distance access.
> If they don’t have the access, the call doesn’t go through.
>
> I always thought that was what ADRs were for.
>
> Was there another way to accomplish this?
>
> Are we talking adding route patterns with the ‘+’ in it?
>
>
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On Apr 3, 2019, at 10:59 PM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> Huh, I have never noticed this before. I just checked a few different
> environments I have access to, and none of them address international
> patterns.  Good catch though.  It seems like you already know it, and Lelio
> confirmed the solution.  Though, I'd argue that Lelio's configuration is
> backwards; you should be globalizing the number, not localizing it, but
> that's none of my business. ;)
>
> Could you just avoid using these rules all together and implement your
> dialing habit support in the CUCM dial plan?  E.g., xlates and xforms?
>
> On Wed, Apr 3, 2019 at 7:51 AM Reto Gassmann  wrote:
>
>> Hallo group
>>
>> I have to configure Application Dial Rules on a CUCM 10.5 to prefix
>> numbers for Click to call (eg Firefox, Chrome,...) with Jabber 12.5.
>> Our national numbers are all 10 digits long. So one ADR is enough.
>> But how about international numbers. They all differ in lenght.
>> It is true to build a ADR for every length? So that would result in many
>> rules from 9 to 20 or more Number of Digits.
>>
>> Any other ideas, how I could handle the Click 2 Call issue on jabber?
>> Regard Reto
>> ___
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>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
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Re: [cisco-voip] Mask caller ID of some message senders in Unity Connection Single inbox

[Evgeny Izetov]

I'm glad it's working for you but I'm curious how it works. Modifying
calling number on the SIP trunk to CUC will work as you describe for a
forwarded call scenario but will break a direct call when a user pushes
Messages button on the phone. I did testing on 11.5 and it most definitely
was using info on the User Basics page for single inbox. Maybe you have
forward to email instead of Single Inbox? CUC does use the number and name
received over SIP trunk for single inbox but only if it didn't find the
number in the database, like it would be the case for PSTN calls.

On Sat, Mar 30, 2019 at 9:51 AM Ray Maslanka  wrote:

> After some more testing we ultimately found the combination of changing
> labels on the calling device itself and changing the calling party
> transformation patterns to include spoofed numbers rather than restricting
> them entirely is our solution.
>
> I assumed the name and numbers may have been dictated by CUC as Evgeny
> Isetov suggests, but that is not the case at least in this deployment
> running 12.5 and a SIP integration.
>
> Removing the Remote Party Id from the integration SIP trunk also does not
> set the caller to unknown as the design guides suggest.
>
> While setting  the calling party transformation patterns to restrict
> calling ID entirely works great in this environment for handling CLID
> during phone to phone calls, the restriction didn't apply to the CUC trunk.
> The full +E164 DN was being shown in the CUC Single inbox message.
> Changing the calling number rather than restricting it entirely does work
> on the SIP integration trunk though.
>
> As a nice bonus, after convinced the number in the inbox was dictated by
> CUCM rather than CUC, we also found relabeling the DN allowed us control
> the name display in Single Inbox as well.
>
> Hope this helps someone.  Thanks Mark and Evengy for the suggestions
> regardless.
>
>
>
> On Tue, Mar 26, 2019 at 4:03 PM Evgeny Izetov  wrote:
>
>> The following kind of sort of works in my quick testing but it's a bit
>> convoluted.
>>
>> The logic that CUC uses is when a call comes in it tries to match the
>> calling number to an extension in its database. If extension is found, then
>> CUC uses the Display Name and the Extension fields on 'User Basics' page to
>> construct the email for Single Inbox. So you could move the executive's
>> real extension from the User Basics page to Alternate Extension in CUC and
>> configure a fake extension and a generic Display Name on the User Basics
>> page.
>>
>> When the executive leaves a voicemail, CUC determines the target
>> voicemail box using the first diversion header as usual (we are not messing
>> with this part). Then CUC matches the executive's calling number to the
>> alternate extension and uses the information on the 'User Basics' of the
>> executive to construct the "From" information in the email.
>>
>> The obvious caveat is the fake extension still has to be unique in CUC,
>> you can't assign a generic main number to multiple voicemail boxes.
>>
>> -E
>>
>>
>> On Tue, Mar 26, 2019 at 2:51 PM Mark H. Turpin 
>> wrote:
>>
>>> Ray,
>>>
>>>
>>>
>>> I haven’t tested this, but a back of the napkin design might be:
>>>
>>>
>>>
>>> Use a route next hop by calling party translation for your VM pilot.
>>>
>>>
>>>
>>> If a call matches the extensions of executives, then it could goto a
>>> different route pattern, like *.
>>>
>>>
>>>
>>> Then you can mask at the route pattern or RL/RG level.
>>>
>>>
>>>
>>> *From:* cisco-voip  *On Behalf Of *Ray
>>> Maslanka
>>> *Sent:* Tuesday, March 26, 2019 10:38 AM
>>> *To:* cisco-voip@puck.nether.net
>>> *Subject:* [cisco-voip] Mask caller ID of some message senders in Unity
>>> Connection Single inbox
>>>
>>>
>>>
>>> *** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
>>>
>>> Gents,
>>>
>>>
>>>
>>> Running CUCM and a SIP integration to Unity Connection with Single Inbox
>>> to O365.  Normally if a CUC user is forwarded to a voice mailbox of another
>>> CUC user, the Identified User Messaging feature provides the name and
>>> number of the caller in the Outlook message.  The ask is now to have the
>>> numbers of a small subset of users masked in the Outlook message when they
>>> leave voice messages.
>>>
>>>
>>>
>>> The use case is when executives, support staff, call center agents, et

Re: [cisco-voip] Mask caller ID of some message senders in Unity Connection Single inbox

[Evgeny Izetov]

The following kind of sort of works in my quick testing but it's a bit
convoluted.

The logic that CUC uses is when a call comes in it tries to match the
calling number to an extension in its database. If extension is found, then
CUC uses the Display Name and the Extension fields on 'User Basics' page to
construct the email for Single Inbox. So you could move the executive's
real extension from the User Basics page to Alternate Extension in CUC and
configure a fake extension and a generic Display Name on the User Basics
page.

When the executive leaves a voicemail, CUC determines the target voicemail
box using the first diversion header as usual (we are not messing with this
part). Then CUC matches the executive's calling number to the alternate
extension and uses the information on the 'User Basics' of the executive to
construct the "From" information in the email.

The obvious caveat is the fake extension still has to be unique in CUC, you
can't assign a generic main number to multiple voicemail boxes.

-E


On Tue, Mar 26, 2019 at 2:51 PM Mark H. Turpin  wrote:

> Ray,
>
>
>
> I haven’t tested this, but a back of the napkin design might be:
>
>
>
> Use a route next hop by calling party translation for your VM pilot.
>
>
>
> If a call matches the extensions of executives, then it could goto a
> different route pattern, like *.
>
>
>
> Then you can mask at the route pattern or RL/RG level.
>
>
>
> *From:* cisco-voip  *On Behalf Of *Ray
> Maslanka
> *Sent:* Tuesday, March 26, 2019 10:38 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] Mask caller ID of some message senders in Unity
> Connection Single inbox
>
>
>
> *** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
>
> Gents,
>
>
>
> Running CUCM and a SIP integration to Unity Connection with Single Inbox
> to O365.  Normally if a CUC user is forwarded to a voice mailbox of another
> CUC user, the Identified User Messaging feature provides the name and
> number of the caller in the Outlook message.  The ask is now to have the
> numbers of a small subset of users masked in the Outlook message when they
> leave voice messages.
>
>
>
> The use case is when executives, support staff, call center agents, etc.
> want to make calls and leave voicemail but avoid providing direct call back
> information.  Caller ID is masked currently from phone to phone using
> Calling Party Transformation Patterns in CUCM, but this doesn't mask caller
> information when the call reaches Unity Connection.
>
>
>
> The Remote Party Id on the integration SIP trunk can be removed to make
> all callers over that trunk unknown, but we only want a small subset of
> users to be masked.  Alternatively, the Identified User Messaging feature
> can be turned off to prevent name and number in the message, but that is
> system wide.
>
>
>
> Does anyone have a decent solution to mask the caller ID of a relatively
> few users in Unity Connection Single Inbox messages?
>
>
>
>
>
>
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Re: [cisco-voip] IOS SIP Profile match calling number less then 10 digits

[Evgeny Izetov]

Could try something like this:

voice translation-rule 1
rule 1 /^[2-9]..[2-9]..$/ /&/
rule 2 /.*/ /4045551212/

So the logic is, if it's 10 digits then leave it be, anything else gets
masked.

If your extensions are always 4 digits then could just add this as the last
rule:

rule 10 /^$/ /4045551212/

On Wed, Nov 28, 2018, 5:54 PM Jason Aarons (Americas) <
jason.aar...@dimensiondata.com wrote:

>
>
> I have a few phones missing an External Phone Number Mask, was thinking
> about adding a IOS translation rule on outbound dial peer that if less than
> less than 10 digits change calling number to our main number. Any got a
> sample?
>
> SIP carrier is passing some extensions out to PSTN by accident.
>
> Get Outlook for Android 
>
>
>
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
> 
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Re: [cisco-voip] CUCM 11.5.1SU5 from Pub go to Serviceablity and subscriber error?

[Evgeny Izetov]

Just built SU5 in a lab a couple of days ago and noticed the same error
initially. In my case the error went away after enabling feature services
on the subscriber. Not sure what that was about.

On Fri, Oct 5, 2018, 2:03 PM Jason Aarons (Americas) <
jason.aar...@dimensiondata.com> wrote:

>
>
> Any pointers on when Serviceability can't show services status on another
> sub etc? Is that tomcat-trust or ipsec-trust etc?
>
> Seen it on a couple clusters running 11.5.1SU5. Perhaps a known bug.
>
> Db replication status is happy.
>
> Get Outlook for Android 
>
>
>
> This email and all contents are subject to the following disclaimer:
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Re: [cisco-voip] Webex hybrid call service connect unable to find user

[Evgeny Izetov]

They were saying on Cisco Live session that the trunk CSS doesn't take
effect for Teams client calling to PSTN, it would be the line CSS.

>From here:
Cisco Webex Hybrid Services (formerly known as Spark Hybrid Services) Call
Service Connect Architecture and Design - BRKCOL-2202

-E

On Tue, Sep 11, 2018 at 4:50 PM Lelio Fulgenzi  wrote:

> Aha – I knew there was something magical about it. 
>
>
>
> And I like the service parameter too. Make sense to set it to Remote
> Destination Profile + Line Calling Search Space (the other option).
>
>
>
> This is making me think more and more that I don’t need two SIP trunks
> from CUCM to EXP-C/EXP-E, one with onprem access only and one with PSTN
> access.
>
>
>
> Lelio
>
>
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* Brian Meade 
> *Sent:* Tuesday, September 11, 2018 4:34 PM
> *To:* Lelio Fulgenzi 
> *Cc:* Ryan Huff ; Dana Tong ;
> cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] Webex hybrid call service connect unable to
> find user
>
>
>
> Off the top of my head, I think it's similar to Single Number Reach/Mobile
> Voice Access.  So it's trying to match the source SIP URI to the
> Spark/Webex Remote Destination in CUCM.
>
>
>
> There's a CallManager Service Parameter "Inbound Calling Search Space for
> Remote Destination" that controls if it uses the SIP Trunk CSS or the
> Remote Destination CSS.  I think the default it the SIP Trunk CSS.
>
>
>
> On Tue, Sep 11, 2018 at 4:07 PM Lelio Fulgenzi  wrote:
>
> I’m going to be going through that guide over the next couple of days –
> hopefully being able to understand one question I have.
>
>
>
> With MRA, because devices are registering directly to CUCM, they get their
> CSS from CUCM, i.e. ability to make off-campus calls.
>
>
>
> With WebEx/Teams Hybrid setup, if I want those Teams clients to make PSTN
> calls using our PSTN service, is that done automagically like MRA, or is
> there more to it?
>
>
>
> Is there any point that the CSS of the SIP trunk needs to have this
> enabled?
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Ryan
> Huff
> *Sent:* Monday, September 10, 2018 6:07 AM
> *To:* Dana Tong 
> *Cc:* cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] Webex hybrid call service connect unable to
> find user
>
>
>
> The deployment guide does talk about the appropriate search rule needed.
> There is a search rule that must match the first entry in CUCM’s Enterprise
> Parameter, “Fully Qualified Cluster Domain”.
>
>
>
> I would review the sections that describe the configuration of
> Expressway-E and Expressway-C:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide/cmgt_b_ciscospark-hybrid-call-service-config-guide_chapter_0110.html
>
> Sent from my iPhone
>
>
> On Sep 10, 2018, at 04:28, Dana Tong  wrote:
>
> Hi all,
>
>
>
> I have made some progress. I can see the call come in to the Expressway-E
> now but the search history says  x...@cluster1.company.com.au
>
>
>
> My CUCM Enterprise parameter for “Cluster Fully Qualified Domain Name” is
> cluster1.company.com.au so it’s learning this via AXL on the Expressway
> Connector.
>
>
>
> Do I need some kind of search rule / or transform on my Expressway Edge to
> convert this back to x...@company.com.au?
>
>
>
> Cheers
>
> Dana
>
>
>
>
>
> *From:* Brian Meade 
> *Sent:* Friday, 7 September 2018 9:25 PM
> *To:* Dana Tong 
> *Cc:* Ryan Huff ; cisco-voip voyp list <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Webex hybrid call service connect unable to
> find user
>
>
>
> This is when you've got to look through the B2B setup and make sure MTLS
> is setup properly and that the certificates for Webex have been imported
> into your Expressway-E.  If you run diagnostics logging on the Expressway-E
> for a call, do you see anything in the capture for tcp.port==5062?
>
>
>
> On Thu, Sep 6, 2018 at 8:43 PM Dana Tong  wrote:
>
> Yep. I have an enduser in CUCM with a Mail ID. This portion is okay. I
> think it’s more a call setup from Webex to Expressway Edge problem.
>
>
>
> I see no incoming call on the Edge.
>
>
>
>
>
>
>
>
>
>
>
>
>
> *From:* Ryan Huff 
> *Sent:* Friday, 7 September 

Re: [cisco-voip] SIP Carrier Service inside China mainland ?

[Evgeny Izetov]

Also have offices in mainland China and AT says they can't bring SIP
service there.

-E

On Tue, Sep 11, 2018 at 4:19 PM Ed Puzziferri 
wrote:

> Hello Jason,
>
>
>
> We are opening an office in Shanghai and we were also told by carriers in
> China that SIP service is not an option they offer, so we are getting a PRI.
>
>
>
> Ed
>
>
>
> *From:* cisco-voip  *On Behalf Of *Jason
> Aarons (Americas)
> *Sent:* Tuesday, September 11, 2018 3:09 PM
> *To:* cisco-voip (cisco-voip@puck.nether.net) 
> *Subject:* [cisco-voip] SIP Carrier Service inside China mainland ?
>
>
>
>
>
> Anyone have an office with a carrier SIP service in mainland China
> providing DIDs?  My customer has several offices and China Telecom seems to
> indicate SIP Trunks are not available.
>
>
>
> -jason
>
>
>
>
>
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer;
> 
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Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

[Evgeny Izetov]

Could you get an actual pcap from the CUBE so we can look at it in
Wireshark?

On Fri, Aug 10, 2018, 10:35 PM Benjamin Turner 
wrote:

> I did and didnt find any..
>
> I tried almost everything before reaching out to the team.
>
> Get Outlook for Android 
>
> --
> *From:* Ryan Huff 
> *Sent:* Friday, August 10, 2018 10:31:57 PM
> *To:* Benjamin Turner
> *Cc:* Loren Hillukka; cisco-voip voyp list
> *Subject:* Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>
> Have you looked at the stats on 0/0/0? See any CRC errors or giants?
>
> Sent from my iPhone
>
> On Aug 10, 2018, at 22:28, Benjamin Turner  wrote:
>
> IOS Bug?
>
> Get Outlook for Android 
>
> --
> *From:* Benjamin Turner 
> *Sent:* Friday, August 10, 2018 10:27:31 PM
> *To:* Ryan Huff
> *Cc:* Loren Hillukka; cisco-voip voyp list
> *Subject:* Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>
> This is strange. The monitor capture on the cube sees the 1xx responses
> but my debug does not.
>
> Get Outlook for Android 
>
> --
> *From:* Benjamin Turner 
> *Sent:* Friday, August 10, 2018 10:25:16 PM
> *To:* Ryan Huff
> *Cc:* Loren Hillukka; cisco-voip voyp list
> *Subject:* Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>
> Acl is not blocking and I tried to set binding on outbound dial-peer to
> outbound interface and inbound dial-peer to inbound interface
>
> Get Outlook for Android 
>
> --
> *From:* Ryan Huff 
> *Sent:* Friday, August 10, 2018 10:23:00 PM
> *To:* Benjamin Turner
> *Cc:* Loren Hillukka; cisco-voip voyp list
> *Subject:* Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>
> I would suggest a couple of things to start with;
>
> 1.) Ditch the global bindings on 0/0/0 and bind your individual dial-peers
> to the appropriate interfaces
>
> 2.) Verify your ACL 101 isn’t interfering (you didn’t include your access
> lists so I can’t tell)
>
> Sent from my iPhone
>
> On Aug 10, 2018, at 22:15, Benjamin Turner  wrote:
>
> Very basic:
>
>
>
>
>
>
>
> version 15.5
>
>
>
> voice service voip
>
> ip address trusted list
>
>   ipv4 67.231.8.75
>
>   ipv4 67.231.12.12
>
>   ipv4 192.168.1.20
>
>   ipv4 162.245.36.90
>
> address-hiding
>
> mode border-element license capacity 100
>
> allow-connections sip to sip
>
> no supplementary-service sip moved-temporarily
>
> no supplementary-service sip refer
>
> fax protocol pass-through g711ulaw
>
> sip
>
>   bind control source-interface GigabitEthernet0/0/0
>
>   bind media source-interface GigabitEthernet0/0/0
>
>   registrar server
>
>   no update-callerid
>
>   early-offer forced
>
>   midcall-signaling passthru
>
>   pass-thru content sdp
>
> !
>
> voice class codec 1
>
> codec preference 1 g711ulaw
>
> codec preference 2 g729r8
>
> !
>
> !
>
> voice class sip-profiles 30
>
> request ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
>
> !
>
> !
>
> !
>
> !
>
> !
>
> voice translation-rule 2
>
> rule 1 /\+1\([2-9].\)/ /\1/
>
> !
>
> voice translation-rule 11
>
> rule 1 /\([2-9]..[2-9]..$\)/ /+1\1/
>
> rule 2 /\(.*\)/ /+\1/
>
> !
>
> voice translation-rule 22
>
> rule 1 /9\(1[2-9]..[2-9]..\)$/ /\1/
>
> rule 2 /9\(911\)$/ /\1/
>
> rule 3 /9\([2-8]11\)$/ /\1/
>
> !
>
> !
>
> voice translation-profile 10DigitTo+1
>
> translate calling 11
>
> translate called 11
>
> !
>
> voice translation-profile LOCALIZE
>
> translate calling 2
>
> !
>
> voice translation-profile OutgoingToBandwidthSIP
>
> translate calling 2
>
> translate called 11
>
> !
>
> !
>
> !
>
> !
>
> voice-card 0/4
>
> dsp services dspfarm
>
> no watchdog
>
> !
>
> !
>
> interface GigabitEthernet0/0/0
>
> description WAN SIP TRUNK TO BANDWIDTH
>
> ip address 162.245.36.90 255.255.255.248
>
> ip nat outside
>
> ip access-group 101 in
>
> ip access-group 101 out
>
> negotiation auto
>
> no cdp enable
>
> !
>
> interface GigabitEthernet0/0/1
>
> description LAN
>
> ip address 192.168.1.1 255.255.255.0
>
> ip nat inside
>
> negotiation auto
>
> !
>
> interface GigabitEthernet0/0/2
>
> no ip address
>
> negotiation auto
>
> !
>
> interface Service-Engine0/4/0
>
> !
>
> interface GigabitEthernet0
>
> vrf forwarding Mgmt-intf
>
> no ip address
>
> negotiation auto
>
> !
>
> !
>
> ip nat inside source list 100 interface GigabitEthernet0/0/0 overload
>
> !
>
> !
>
> !
>
> dial-peer voice 1000 voip
>
> description BANDWIDTH Incoming SIP and VOIP calls
>
>  session protocol sipv2
>
> session target sip-server
>
> incoming called-number .
>
> voice-class codec 1
>
>   dtmf-relay rtp-nte sip-notify
>
> no vad
>
> !
>
> dial-peer voice 4000 voip
>
> description Local Dialing (10-Digit) to Bandwidth SIP Trunk
>
> translation-profile outgoing OutgoingToBandwidthSIP
>
> destination-pattern [2-9]..[2-9]..$
>
> session protocol sipv2
>
> session target ipv4:67.231.8.75
>
> voice-class codec 1
>
>  

Re: [cisco-voip] SIP header modification

[Evgeny Izetov]

On second look, I don't think I got that right, so disregard :-)

On Mon, Jun 11, 2018, 12:53 PM Evgeny Izetov  wrote:

> Looks like the variable in the replace string doesn't have brackets for
> it. The only brackets are taken by the \1 variable. You'd need a second set
> of brackets to insert the \u01 variable.
>
> On Mon, Jun 11, 2018, 12:29 PM James Buchanan 
> wrote:
>
>> Hello,
>>
>> I am trying to take a REFER header, copy it to a copy list, and use it to
>> modify the To: header.
>>
>> I have an inbound dial peer using:
>>
>> voice class sip copy-list 1
>> sip-header Refer-To
>>
>>
>> and I have the following SIP profile in place for the outgoing dial peer:
>>
>> voice class sip-profiles 1
>>  request ANY sip-header Refer-To copy 
>> "Refer-To:)"
>> u01
>>
>>  request INVITE sip-header To modify "(To: " "\1\u01"
>>
>>
>> Whenever I run this through the SIP profile test tool (at
>> https://cway.cisco.com/tools/SipProfileTest/) it give me the properly
>> modified To: header. However, whenever, I run it through CUBE, I get this:
>>
>> To: >
>> I'm sure there's something simple I'm missing. Any suggestions?
>>
>> Thanks,
>>
>> James
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
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Re: [cisco-voip] SIP header modification

[Evgeny Izetov]

Looks like the variable in the replace string doesn't have brackets for it.
The only brackets are taken by the \1 variable. You'd need a second set of
brackets to insert the \u01 variable.

On Mon, Jun 11, 2018, 12:29 PM James Buchanan 
wrote:

> Hello,
>
> I am trying to take a REFER header, copy it to a copy list, and use it to
> modify the To: header.
>
> I have an inbound dial peer using:
>
> voice class sip copy-list 1
> sip-header Refer-To
>
>
> and I have the following SIP profile in place for the outgoing dial peer:
>
> voice class sip-profiles 1
>  request ANY sip-header Refer-To copy 
> "Refer-To:)"
> u01
>
>  request INVITE sip-header To modify "(To: " "\1\u01"
>
>
> Whenever I run this through the SIP profile test tool (at
> https://cway.cisco.com/tools/SipProfileTest/) it give me the properly
> modified To: header. However, whenever, I run it through CUBE, I get this:
>
> To: 
> I'm sure there's something simple I'm missing. Any suggestions?
>
> Thanks,
>
> James
> ___
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> cisco-voip@puck.nether.net
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>
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Re: [cisco-voip] FXO port ignore incoming

[Evgeny Izetov]

Under the FXO port try "ring number x" where x is how many times the port
should ring before VG answers it. I don't have a VG at hand right now to
see the max value of rings possible but it could accomplish what you need.

-E

On Dec 7, 2017 3:53 PM, "Norton, Mike"  wrote:

> Unfortunately the gateway in question doesn’t have any FXS ports in it. If
> I have to resort to doing plar opx to somewhere that never answers, ideally
> the “somewhere” should not require any hardware.
>
>
>
> During normal operation, the analog line is just for outgoing local 911
> calls. Incoming calls come via WAN from the PRI at another site. But the
> telco rings the local analog line simultaneously with the PRI. During power
> outages, staff plug in a POTS phone in order to get the incoming calls via
> the analog line. During normal operation I need to ignore the ringing on
> the analog line so that calls don’t all show up at the PBX twice.
>
>
>
> -mn
>
>
>
>
>
> *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
> Meade
> *Sent:* December 7, 2017 1:08 PM
> *To:* Norton, Mike 
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] FXO port ignore incoming
>
>
>
> OPX to an FXS port maybe?
>
>
>
> I'm curious what the use case here if you never want it to answer.
>
>
>
> On Thu, Dec 7, 2017 at 2:43 PM, Norton, Mike 
> wrote:
>
> I need to make an FXO port ignore incoming calls. I.e., don’t take the
> trunk off-hook, and don’t make any phones ring. Just let the trunk keep
> ringing indefinitely.
>
>
>
> What is the best way to accomplish this? I could swear I’ve done this
> before, but I’m drawing a blank at the moment... connection plar opx to
> somewhere that never answers? There must be a better way than that.
>
>
>
> -mn
>
>
> ___
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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>
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Re: [cisco-voip] T1 PRI configuration on 2921 not matching startup config

[Evgeny Izetov]

I wonder if you only had enough DSPs for those 2 and 1/3 PRIs.

On Oct 2, 2017 4:02 PM, "Lelio Fulgenzi"  wrote:



Well, apparently, CUCM enabling of ports overrides the router
configuration.



I did have all four ports configured. After deleting those two ports, the
running now matches the startup.



This is very weird. Is there a configuration command that says only
download configuration for the ports I’ve enabled?







---

Lelio Fulgenzi, B.A.

Senior Analyst, Network Infrastructure

Computing and Communications Services (CCS)

University of Guelph



519-824-4120 Ext 56354 <(519)%20824-4120>

le...@uoguelph.ca

www.uoguelph.ca/ccs

Room 037, Animal Science and Nutrition Building

Guelph, Ontario, N1G 2W1



*From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
Of *Lelio Fulgenzi
*Sent:* Monday, October 02, 2017 2:29 PM
*To:* voyp list, cisco-voip (cisco-voip@puck.nether.net)
*Subject:* [cisco-voip] T1 PRI configuration on 2921 not matching startup
config





OK – Having a weird issue. I was pretty sure that the router config took
precedence over CUCM (with respect to enabling ports).



I’ve got four of  the four ports configured on CUCM. So not sure why three
are being configured on the router.



Very strange!



My startup looks something like this:



controller T1 0/0/0

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/1

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/2

shutdown

cablelength long 0db

!

controller T1 0/0/3

shutdown

cablelength long 0db

!



!

interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

isdn bchan-number-order ascending

no cdp enable



voice-port 0/0/0:23

echo-cancel coverage 64

!

voice-port 0/0/1:23

!







But after a reload, my running config looks something like this:



controller T1 0/0/0

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/1

cablelength long 0db

pri-group timeslots 1-24 service mgcp

!

controller T1 0/0/2

cablelength long 0db

pri-group timeslots 1-8,24 service mgcp

!

controller T1 0/0/3

cablelength long 0db

!



interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn protocol-emulate network

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

interface Serial0/0/2:23

no ip address

encapsulation hdlc

isdn switch-type primary-dms100

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!



!

voice-port 0/0/0:23

echo-cancel coverage 64

!

voice-port 0/0/1:23

echo-cancel coverage 64

!

voice-port 0/0/2:23

echo-cancel coverage 64



---

Lelio Fulgenzi, B.A.

Senior Analyst, Network Infrastructure

Computing and Communications Services (CCS)

University of Guelph



519-824-4120 Ext 56354 <(519)%20824-4120>

le...@uoguelph.ca

www.uoguelph.ca/ccs

Room 037, Animal Science and Nutrition Building

Guelph, Ontario, N1G 2W1



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Re: [cisco-voip] vg224 registered to two different Cm's

[Evgeny Izetov]

Also check your DNS records several times for every FQDN both forward and
reverse. I've seen something similar before where devices showed registered
in one node and unregistered in another, and it turned out DNS was
misconfigured (for the same query it was returning different results when
you ran it against nslookup multiple times).

On Tue, Jan 17, 2017 at 2:48 PM, Brian Meade <bmead...@vt.edu> wrote:

> Try looking at the RISDB on the sub: "show ridb query gateway" and see if
> it shows Registered on the Sub. If so, RIS DC restart should fix it so it
> shows right on the pub webpage.
>
> On Tue, Jan 17, 2017 at 2:44 PM, Scott Voll <svoll.v...@gmail.com> wrote:
>
>> sh ccm shows they resolve and are registered to the sub and backup to the
>> pub.
>>
>> That is the wierd thing.  as some voice ports show the sub and some the
>> pub
>>
>> we are looking at restarting the RIS data collector
>>
>> Scott
>>
>>
>> On Tue, Jan 17, 2017 at 11:09 AM, Evgeny Izetov <eize...@gmail.com>
>> wrote:
>>
>>> Can those VGs resolve the FQDNs? My guess they wouldn't even register if
>>> they couldn't but maybe they are keeping some sort of 'last known good'
>>> configuration.
>>>
>>> On Tue, Jan 17, 2017 at 1:59 PM, Jeffrey McHugh <jmch...@fidelus.com>
>>> wrote:
>>>
>>>> Sometimes restarting RIS data collector can clear registration
>>>> anomalies
>>>>
>>>>
>>>>
>>>> *From:* Scott Voll [mailto:svoll.v...@gmail.com]
>>>> *Sent:* Tuesday, January 17, 2017 1:58 PM
>>>> *To:* Brian Meade <bmead...@vt.edu>
>>>> *Cc:* Jeffrey McHugh <jmch...@fidelus.com>; cisco-voip@puck.nether.net
>>>> *Subject:* Re: [cisco-voip] vg224 registered to two different Cm's
>>>>
>>>>
>>>>
>>>> yup but did changed from IP to FQDN on the cluster..
>>>>
>>>>
>>>>
>>>> On Tue, Jan 17, 2017 at 10:57 AM, Brian Meade <bmead...@vt.edu> wrote:
>>>>
>>>> Same Device Pool on all the ports?
>>>>
>>>>
>>>>
>>>> On Tue, Jan 17, 2017 at 1:55 PM, Scott Voll <svoll.v...@gmail.com>
>>>> wrote:
>>>>
>>>> sorry, mgcp.. did a no mgcp and mgcp.  no change
>>>>
>>>>
>>>>
>>>> reset from CM and all registered to cmsub reset but the once on the pub
>>>> didn't.
>>>>
>>>>
>>>>
>>>> CM shows good replication on the DB's.  it's the wierdest thing I've
>>>> seen.
>>>>
>>>>
>>>>
>>>> Scott
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jan 17, 2017 at 10:17 AM, Brian Meade <bmead...@vt.edu> wrote:
>>>>
>>>> Try resetting all the ports from CUCM then maybe bounce SCCP on the VG
>>>> if that still didn't work.
>>>>
>>>>
>>>>
>>>> On Tue, Jan 17, 2017 at 12:45 PM, Scott Voll <svoll.v...@gmail.com>
>>>> wrote:
>>>>
>>>> All are in the same DP.  we just changed from IP to FQDN over the
>>>> weekend.  Any other thoughts?
>>>>
>>>>
>>>>
>>>> Scott
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jan 17, 2017 at 9:41 AM, Jeffrey McHugh >>> <jmch...@fidelus.com>> wrote:
>>>>
>>>> Pretty sure you can assign device pools to the specific ports,
>>>> different DP’s may have diff. call mgr groups
>>>>
>>>>
>>>>
>>>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>>>> Behalf Of *Scott Voll
>>>> *Sent:* Tuesday, January 17, 2017 12:38 PM
>>>> *To:* cisco-voip@puck.nether.net
>>>> *Subject:* [cisco-voip] vg224 registered to two different Cm's
>>>>
>>>>
>>>>
>>>> How can some ports on a VG register to the pub and some to the Sub on
>>>> the same VG?
>>>>
>>>>
>>>>
>>>> totally confused on this one.
>>>>
>>>>
>>>>
>>>> CM 10.5.2 /  VG vg224-i6k9s-mz.151-4.M9.bin
>>>>
>>>>
>>>>
>>>> Scott
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>

Re: [cisco-voip] vg224 registered to two different Cm's

[Evgeny Izetov]

Can those VGs resolve the FQDNs? My guess they wouldn't even register if
they couldn't but maybe they are keeping some sort of 'last known good'
configuration.

On Tue, Jan 17, 2017 at 1:59 PM, Jeffrey McHugh  wrote:

> Sometimes restarting RIS data collector can clear registration anomalies
>
>
>
> *From:* Scott Voll [mailto:svoll.v...@gmail.com]
> *Sent:* Tuesday, January 17, 2017 1:58 PM
> *To:* Brian Meade 
> *Cc:* Jeffrey McHugh ; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] vg224 registered to two different Cm's
>
>
>
> yup but did changed from IP to FQDN on the cluster..
>
>
>
> On Tue, Jan 17, 2017 at 10:57 AM, Brian Meade  wrote:
>
> Same Device Pool on all the ports?
>
>
>
> On Tue, Jan 17, 2017 at 1:55 PM, Scott Voll  wrote:
>
> sorry, mgcp.. did a no mgcp and mgcp.  no change
>
>
>
> reset from CM and all registered to cmsub reset but the once on the pub
> didn't.
>
>
>
> CM shows good replication on the DB's.  it's the wierdest thing I've seen.
>
>
>
> Scott
>
>
>
>
>
> On Tue, Jan 17, 2017 at 10:17 AM, Brian Meade  wrote:
>
> Try resetting all the ports from CUCM then maybe bounce SCCP on the VG if
> that still didn't work.
>
>
>
> On Tue, Jan 17, 2017 at 12:45 PM, Scott Voll  wrote:
>
> All are in the same DP.  we just changed from IP to FQDN over the
> weekend.  Any other thoughts?
>
>
>
> Scott
>
>
>
>
>
> On Tue, Jan 17, 2017 at 9:41 AM, Jeffrey McHugh  > wrote:
>
> Pretty sure you can assign device pools to the specific ports, different
> DP’s may have diff. call mgr groups
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Scott Voll
> *Sent:* Tuesday, January 17, 2017 12:38 PM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] vg224 registered to two different Cm's
>
>
>
> How can some ports on a VG register to the pub and some to the Sub on the
> same VG?
>
>
>
> totally confused on this one.
>
>
>
> CM 10.5.2 /  VG vg224-i6k9s-mz.151-4.M9.bin
>
>
>
> Scott
>
>
>
>
>
>
>
>
> *Jeffrey McHugh* | Sr. Collaboration Consulting Engineer | VCP-DCV, CCNP
> Collaboration
> [image: Company_Logo_Image] 
> *Fidelus Technologies, LLC*
> Named Best UC Provider in the USA
> 
> 240 West 35th Street, 6th Floor, New York, NY 10001
> *+1-212-616-7801 <(212)%20616-7801>* office | *+1-212-616-7850
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> 
>
>
> Disclaimer - January 17, 2017
>
> This email and any files transmitted with it are confidential and
> intended solely for *Scott Voll,Brian Meade,cisco-voip@puck.nether.net
> *. If you are not the named addressee you
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>
>
>
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>
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Re: [cisco-voip] Caller ID Manipulation

[Evgeny Izetov]

It still works with SIP. It is controlled by the "Transmit UTF-8 for
Calling Party Name" checkbox on a SIP trunk. When unchecked (default),
ASCII Display Name gets sent. If you check it, then Display Name is sent.
If ASCII Display Name is blank, then I believe just the number is going to
be sent out.

On Tue, Jan 17, 2017 at 12:44 PM, Norton, Mike <mikenor...@pwsd76.ab.ca>
wrote:

> Oh yes, that is a good trick I used to use back when my PRI gateways were
> MGCP. Does it work with SIP too? I guess I just always assumed SIP would
> take Unicode names.
>
> Not actually using CUCM any longer, which is why I do the names on the
> gateway now and forgot about the ASCII Name trick.
>
> -mn
> ------
> From: Evgeny Izetov
> Sent: 17/01/2017 10:31 AM
> To: Norton, Mike
> Cc: Alan Libbee; Cisco VoIP Group; Ben Amick
>
> Subject: Re: [cisco-voip] Caller ID Manipulation
>
> You could also look into changing the ASCII Display Name on line
> appearances as that's what gets sent to trunks by default.
>
> On Jan 17, 2017 12:18 PM, "Norton, Mike" <mikenor...@pwsd76.ab.ca> wrote:
>
> Alan, I think the CNAM database lookup is only U.S. thing. In Canada and
> presumably elsewhere, the calling name is set by the caller (if ISDN or
> SIP, otherwise by their CO switch if analog POTS) and is passed all the way
> through the PSTN to the callee.
>
>
>
> I’ve never understood why the CUCM web interface calls the field “internal
> caller ID” because there is nothing internal about it. The PSTN in some
> countries, such as U.S.A., ignores it, but that shouldn’t be a license for
> Cisco to misname the field, leading to people getting surprised when they
> discover that the name is not kept internal.
>
>
>
> Likewise, there is nothing about Ben’s SIP provider that I would agree is
> “very lax.” To me, it sounds very perfectly normal, because that is the way
> the PSTN works for me.
>
>
>
> On my PRI gateways, I modify the outgoing names by using an inbound SIP
> profile on the incoming SIP dial-peer. This allows me to alter the name in
> SIP From field based on what number is in the From field. So I can put a
> generic name for certain number ranges, special names for specific numbers,
> no change for incoming PSTN calls that are forwarded back out to the PSTN,
> etc. I think you should be able to do the same on a CUBE. Here’s some
> snippets of it as an example:
>
>
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+17808643696
> <(780)%20864-3696>@)" "\1: \"SPRT RVR REG AC\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+17805682265
> <(780)%20568-2265>@)" "\1: \"TP CREEK SCHOOL\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+17807662294
> <(780)%20766-2294>@)" "\1: \"WMBLY ELEM SCHL\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+17805328133
> <(780)%20532-8133>@)" "\1: \"PWSD76\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+178035752[01][0-9]@)"
> "\1: \"PWSD76\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+1780357522[0-4]@)"
> "\1: \"PWSD76\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+178083130[5-9][0-9]@)"
> "\1: \"PWSD76\" \2"
>
> request ANY sip-header From modify "^(.+):.*(<sip:\+17808643741
> <(780)%20864-3741>@)" "\1: \"PWSD76\" \2"
>
>
>
> Note that inbound SIP profiles has to be turned on under “voice service
> voip” in order to work. But I guess you could probably do it on the
> outgoing dial-peer to your SIP provider if you wanted. (I have to do it on
> incoming because my outgoing is PRI.)
>
>
>
> -mn
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Alan Libbee
> *Sent:* Tuesday, January 17, 2017 8:02 AM
> *To:* Ben Amick <bam...@humanarc.com>
> *Cc:* Cisco VOIP <cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Caller ID Manipulation
>
>
>
> Ben,
>
>
>
> You can set calling name on the device or trunk level. Most landlines and
> businesses not using SIP will only receive the calling number, the name is
> looked up through the cnam database. You can query the cnam database here:
> https://www.opencnam.com. The cnam database should be updated to reflect
> your business name, I recommend spot checking your DIDs to see what is
> displayed in cnam. I have access to some different sip carriers if you wan

Re: [cisco-voip] Jabber/CIPC and QoS

[Evgeny Izetov]

Ben, here's the link to the site and the session video:
https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=89103=true

On Wed, Jan 4, 2017 at 11:44 AM, Ryan Huff <ryanh...@outlook.com> wrote:

> Yes, TRP does have some drawbacks; video, binary floor control BUT, works
> great for voice media. It's a heavy overhead and isn't a complete solution
> but works in a pinch if you're dealing with some C Level users that "just
> want the computer phone to work".
>
> I have also been known to swap out the network card in user pcs for dual
> interface cards, then use a persistent route in the PC to force the soft
> phone's traffic to its call control server out of one interface that is on
> the voice network (leaving the other interface on the data network).
>
> A crude solution, but it worked well in a situation where the networking
> gear wouldn't have supported what we would've needed to do with QOS. Dual
> port PC network cards, even in bulk, are a heck of a lot cheaper than new
> networking gear.
>
> Yikes, giving myself flashbacks from rehashing all these memories of being
> a network admin for a nonprofit  need some coffee 
>
> On Jan 4, 2017, at 11:27 AM, Lelio Fulgenzi <le...@uoguelph.ca> wrote:
>
> I would have loved to do MTP resources across the board... helps with
> security as well, less holes to open up. But I found a few features that
> wouldn't work, like desktop sharing, etc. If they supported all features
> with MTP, I'd would have likely been able to justify a couple of routers to
> do it.
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst, Network Infrastructure
> Computing and Communications Services (CCS)
> University of Guelph
>
> 519-824-4120 Ext 56354 <(519)%20824-4120>
> le...@uoguelph.ca
> www.uoguelph.ca/ccs
> Room 037, Animal Science and Nutrition Building
> Guelph, Ontario, N1G 2W1
>
>
>
> --
> *From:* cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Ben
> Amick <bam...@humanarc.com>
> *Sent:* Wednesday, January 4, 2017 11:18 AM
> *To:* Evgeny Izetov; Ryan Huff
> *Cc:* Cisco VoIP Group
> *Subject:* Re: [cisco-voip] Jabber/CIPC and QoS
>
>
> Evgeny,
>
> That’s great, and I was able to find the PDF from the session but I can’t
> seem to remember how to find the site that has the recordings of the
> sessions – could you provide a link to that?
>
>
>
> Ryan,
>
> That sounds like a solid idea for when QoS is absolutely absolutely
> necessary, but I have nowhere near enough MTP resources to do that for all
> the softphones in my org.
>
>
>
> *Ben Amick*
>
> Telecom Analyst
>
>
>
> *From:* Evgeny Izetov [mailto:eize...@gmail.com <eize...@gmail.com>]
> *Sent:* Tuesday, January 03, 2017 10:15 PM
> *To:* Ryan Huff <ryanh...@outlook.com>
> *Cc:* Ben Amick <bam...@humanarc.com>; Cisco VoIP Group <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Jabber/CIPC and QoS
>
>
>
> I saw a CiscoLive! session recently that seemed to recommend the ports and
> access-lists approach. The idea is that you can now specify separate port
> ranges for audio and video in SIP Profile. The session goes quite in depth
> and is worth the watch:
>
> BRKCOL-2616 - QoS Strategies and Smart Media Techniques for Collaboration
> Deployments (2016 Berlin) - 2 Hours
>
>
>
> On Tue, Jan 3, 2017 at 9:49 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>
> I see; while this is by no means a complete solution, it may help. I'm
> assuming Cisco based soft phones (CIPC, CSF, BOT, TAB ... etc).
>
>
>
> You may try Trusted Relay Points (set in the device level configuration).
> This does rely and depend on your media resource architecture and design;
> i.e. you'll need to have media resources that support TRP available.
>
>
>
> Using TRP on the device config for a soft phone will cause CUCM to
> dynamically insert an MTP in the call flow which will allow for adherence
> to QOS trust policies and offer a predetermined network path for call flows
> in an otherwise untrusted network (presumably, the data network).
>
> -Ryan
>
>
>
>
>
> Sent from my iPhone
>
> On Jan 3, 2017, at 9:30 PM, Ben Amick <bam...@humanarc.com> wrote:
>
> Only for softphones. Currently most of our servers live on the same LAN as
> end users, so yeah. Hardphones have their own VLAN so its not as bad. In
> the future it won’t be that way but for the time being it is.
>
>
>
> *Ben Amick*
>
> Telecom Analyst
>
>
>
> *From:* Ryan Huff [mailto:ryanh...@outlook.com <ryanh...@outlook.com>]
> *Sent:* Tuesday, January 03, 2017

Re: [cisco-voip] Jabber/CIPC and QoS

[Evgeny Izetov]

I saw a CiscoLive! session recently that seemed to recommend the ports and
access-lists approach. The idea is that you can now specify separate port
ranges for audio and video in SIP Profile. The session goes quite in depth
and is worth the watch:

BRKCOL-2616 - QoS Strategies and Smart Media Techniques for Collaboration
Deployments (2016 Berlin) - 2 Hours


On Tue, Jan 3, 2017 at 9:49 PM, Ryan Huff  wrote:

> I see; while this is by no means a complete solution, it may help. I'm
> assuming Cisco based soft phones (CIPC, CSF, BOT, TAB ... etc).
>
> You may try Trusted Relay Points (set in the device level configuration).
> This does rely and depend on your media resource architecture and design;
> i.e. you'll need to have media resources that support TRP available.
>
> Using TRP on the device config for a soft phone will cause CUCM to
> dynamically insert an MTP in the call flow which will allow for adherence
> to QOS trust policies and offer a predetermined network path for call flows
> in an otherwise untrusted network (presumably, the data network).
>
> -Ryan
>
>
>
> Sent from my iPhone
> On Jan 3, 2017, at 9:30 PM, Ben Amick  wrote:
>
> Only for softphones. Currently most of our servers live on the same LAN as
> end users, so yeah. Hardphones have their own VLAN so its not as bad. In
> the future it won’t be that way but for the time being it is.
>
>
>
> *Ben Amick*
>
> Telecom Analyst
>
>
>
> *From:* Ryan Huff [mailto:ryanh...@outlook.com ]
> *Sent:* Tuesday, January 03, 2017 9:18 PM
> *To:* Ben Amick 
> *Cc:* NateCCIE ; Cisco VoIP Group <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Jabber/CIPC and QoS
>
>
>
> Ben,
>
>
>
> By flat network; I am to assume that there is no layer 2 partition between
> rtp/signaling and general data traffic?
>
>
> On Jan 3, 2017, at 9:15 PM, Ben Amick  wrote:
>
> Yeah, I have the luck of having MPLS right now, and I don’t see us going
> iWAN for a while for various reasons. QoS on the WAN right now even isn’t
> my issue, it’s QoS on the LAN. Right now we have a relatively flat network,
> and certain segments of our troupe **cough**developers**cough** seems to
> have made our internal traffic ugly, to the point that I may have to do an
> analysis of it, as we’re having just random periods here and there where
> calls just have horrible quality, of the type you normally see fixed by QoS
>
>
>
> *Ben Amick*
>
> Telecom Analyst
>
>
>
> *From:* Ryan Huff [mailto:ryanh...@outlook.com ]
> *Sent:* Tuesday, January 03, 2017 8:40 PM
> *To:* NateCCIE 
> *Cc:* Ben Amick ; Cisco VoIP Group <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Jabber/CIPC and QoS
>
>
>
> It's a shame really ... MPLS is far superior IMO, for many reasons. Call
> it iWAN, DMVPN, AutoVPN  whatever, it is still as Nate says, public
> Internet.
>
>
>
> Try getting a 30 or 60 minute SLA with escalation after 15 minutes from a
> public Comcast or Time Warner/Charter package.
>
>
> On Jan 3, 2017, at 7:53 PM, NateCCIE  wrote:
>
> Or take the most approach of do nothing.
>
>
>
> My personal favorite is to use codecs where QoS matters less, like iLBC,
> OPUS, etc.
>
>
>
> So many business are getting rid of the QoS capable WAN and just doing
> VPNs, even if they have fancy names that make it sound better than public
> internet.
>
> Sent from my iPhone
>
>
> On Jan 3, 2017, at 2:25 PM, Ben Amick  wrote:
>
> So, I know this is an age old question that’s debated, but I’ve been
> wondering if anyone here has a perspective here in regards to QoS for
> softphones. Obviously, with hardphones, you usually partition a separate
> VLAN with AutoQoS/DSCP tags, but that isn’t applicable with softphones.
>
>
>
> I’ve heard of three different options in the past, neither of which seem
> to be very simple to deploy, but all seem to be Jabber-centric.
>
> 1.  Configuring windows to perform DSCP tagging, and do DSCP QoS on
> the switches they are connected to, as well as trusting the device.
> Problems: Requires users to be local admins, openings for abuse and network
> impact due to blind PC trust.
>
> 2.  Configuring your switches with an access list that recognizes the
> ports Jabber does outbound to attach DSCP tags to them. Problems: Other
> programs could theoretically use those ports
>
> 3.  Installing Medianet services on all jabber clients; Configure all
> switches for medianet tagging. Problem: (I think?) Requires newer switches
> to use, maybe needs an additional server (I vaguely remember possibly
> needing prime collab?)?
>
>
>
> Maybe I’m missing some things, but what approach have you guys taken for
> softphone/Jabber QoS? And on top of that, what options are there for CIPC
> (I know there’s the auto qos trust cisco-softphone for cisco switches, 

Re: [cisco-voip] SRST for SIP phones on 4000 Series

[Evgeny Izetov]

I believe when you do 'voice register global' the mode is SRST by default.
So you are not seeing the option of 'mode srst' because you are already in
that mode.

On Sep 14, 2016 1:03 AM, "Sreekanth"  wrote:

> Mode esrst does not mandate the use of the SRST manager. You can configure
> mode esrst and then configure the router using the CLI.
> Esrst provides many more features to the phones over traditional SRST.
>
> Are you not getting the 'id network' option under voice register pool for
> mode esrst?
>
> Thanks
>
> On Sep 13, 2016 9:34 PM, "Jonathan Charles"  wrote:
>
>> Is there a way to deploy conventional SRST on the new 4000 series routers
>> without the SRST manager?
>>
>> It only gives me the option of mode cme or esrst, and when I try to
>> configure the pool it asks for a mac or device-id...
>>
>> Is automatic SRST no longer a thing on the 4000 series?
>>
>>
>>
>> Jonathan
>>
>>
>>
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] Changing vCPU Quantities

[Evgeny Izetov]

The latest ova readme files for CUC do have a one-liner about 1 cpu
restrictions for Unified Messaging under the 1000 users template
information, but yeah, it's pretty easy to overlook.

-E

On Wed, Jul 27, 2016 at 6:41 PM, Erick Wellnitz 
wrote:

> Based on that caveat, I'd think that's the best bet.  Same with running
> out of storage space in anything other than the common partition.  Too bad
> PCD doesn't do CUC migrations.
>
> If I remember correctly this caveat is due to an application limitation
> and not the OS.
>
> On Jul 27, 2016 11:17 AM, "Anthony Holloway" <
> avholloway+cisco-v...@gmail.com> wrote:
>
> >
>
> > Hi Everyone,
> >
> > I have been telling people on this list
>  for a while now,
> and for that matter, in person too, that you can just shutdown the VM and
> modify the vCPU count per VM and then power it back on.
> >
> > In fact, I know I've done this a half a dozen times in the past.
> However, for CUC, a colleague of mine just uncovered the following bullet
> point:
> >
> *> "Adding vCPU is supported for all apps except Unity and Unity
> Connection, but requires VM to be shutdown first."*
> >
> > Source:
> http://docwiki.cisco.com/wiki/Unified_Communications_VMware_Requirements#Resize_Virtual_Machine
> >
> > So, with CUC being the exception, it would stand to reason that CUC
> requires a DRS backup, new OVA deployment, then DRS restore to change the
> vCPU count, correct?
> >
> > To add to the confusion, PDI is even stating
> 
> in certain cases online that changing the vCPU count without a DRS
> backup.restore is support.
> >
> > So, what do you know about this?  Is CUC really different from the other
> apps, or is this a case of bad documentation?
> >
> > Thanks.
> >
> > ___
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net 
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> >
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] UCCX 9 prompts gibberish

[Evgeny Izetov]

I believe when you upload an MoH file to CUCM it stores it in G729 as one
of the formats. Then you could try retrieving it via CLI. Not sure if it
will work, haven't tried.

http://www.netcraftsmen.com/retrieving-music-on-hold-moh-files-from-cucm/

On Tue, Feb 16, 2016 at 11:17 AM, James Buchanan 
wrote:

> No, just for G711. I don't have a solution for G729. Hopefully someone
> else does.
>
> On Tue, Feb 16, 2016 at 11:12 AM, abbas Wali  wrote:
>
>> are we talking about converting them to G729?
>> i took all the steps few times now and everytime its the same.
>> even the file show it is been saved in 64kbps instead of 8.
>>
>> [image: Inline images 1]
>>
>> On 16 February 2016 at 16:03, James Buchanan 
>> wrote:
>>
>>> You can do it in Audacity under Other Formats when you export the file.
>>> However, I've never seen an option for G729.
>>>
>>> On Tue, Feb 16, 2016 at 10:17 AM, abbas Wali  wrote:
>>>
 ​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  wrote:

> Even to g729?
>
> On 16 Feb 2016, at 10:11 AM, Haas, Neal  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 
> *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 <
> 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  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.
>
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
>
> --
>
> *Abbas Wali*
>
>


 --
 *Abbas Wali*

>>>
>>>
>>
>>
>> --
>> *Abbas Wali*
>>
>
>
> ___
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>
>
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