Re: [cisco-voip] [External] Re: No Custom Background Images after CUCM 14

2023-03-27 Thread Bill Talley
You can download the files using the command below:file get tftp Desktops/800x480x24/*Just replace the directory name matching the resolution for any other appropriate phone models and that will download those files to the sftp server and directory you designate after entering that command… assuming you’re talking about downloading the original .png image files and List.xml file. Similarly, the following command will list all of the files in that directory…file list tftp Desktops/800x480x24/*And on the inactive partition file list inactivelog cm/tftpdata/Desktops/800x480x24/*Or to download:file get inactivelog cm/tftpdata/Desktops/800x480x24/*Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.On Mar 27, 2023, at 2:42 PM, JASON BURWELL  wrote:







That for all the responses so far. All of the image files and List.xml appear to have ”survived” the upgrade although since you can’t download files to view them, I have no idea if they are uncorrupted. The issue is that I now have a bunch
 of default images in a new tftpsgn directory. All the images are .sgn and there is also a List.xml.sgn. According to TAC this is a known issues whereas the phones now look for the secure file. They are telling me as soon as all the files in the tftpsgn directory
 are deleted, the phone will revert back to looking in the standard Desktops/800x480x24 directory instead of the tftpsgn/Desktops/800x480x24 I may try to reupload the List.xml but if what they are telling me is correct, it will never work until all the tftpsgn
 files are deleted, which so far we have been unable to do.
 
Jason
 
 

From: Hunter Fuller  
Sent: Monday, March 27, 2023 10:35 AM
To: Bill Talley 
Cc: JASON BURWELL ; cisco-voip@puck.nether.net
Subject: Re: [External] Re: [cisco-voip] No Custom Background Images after CUCM 14

 

Cisco’s upgrade procedure indicates that custom tftp files are never expected to survive an upgrade. I did notice the files not deleting on my system when I went
 to replace them after upgrade to 12. 5, however, you can upload “on top of” the 






Cisco’s upgrade procedure indicates that custom tftp files are never expected to survive an upgrade. 


 


I did notice the files not deleting on my system when I went to replace them after upgrade to 12.5, however, you can upload “on top of” the existing files then delete the .sgn and that seems to work. Unless that changed in 14. 


 


On Mon, Mar 27, 2023 at 09:30 Bill Talley <btal...@gmail.com> wrote:



I assume you’re saying the images don’t appear under Wallpapers on the phones?  Have you verified the appropriate Desktops directory on the tftp server?  I assume yes to these questions below since TAC appears to directing you towards the
 sgn files.  Explanation for these questions is below. 

 


- List.xml is present and contains all of the correct file names/references?


- Background images and thumbnail files are present?


 


I recently did a standard upgrade from cucm 12.5 to 14Su2 and on the pub, the upgrade didn’t copy the custom images to the new partition and List.xml was replaced with the default file so only the default images appeared under wallpapers.
    On the sub, the List.xml was copied during the upgrade, but the images were not.   I ended up downloading the original files from the tftp directory on the  inactive  partition the uploading them via OS Admin and restarting tftp.   This happened for two
 different phone models but not all phone models. 


 

 


 



Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.







On Mar 27, 2023, at 7:26 AM, JASON BURWELL <jason.burw...@foundersfcu.com> wrote:




 









After we upgraded to CUCM 14 with PCD, our 88XX phones no longer have custom wallpaper images available. The solution from TAC was to delete all the tftpsgn files from the TFTP
 directory on each node and restart tftp services however when we delete the files using the GUI or CLI, they remain and do not delete. TAC has tried twice unsuccessfully to access the directory from root and delete the files but they have not been able to
 do that. Surely others have encountered this issue and I’m wondering what everyone else has done.  I can’t believe it’s really this complicated of an issue to resolve. Thanks Jason
 

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-- 






--
Hunter Fuller (they)
Router Jockey
VBH M-1C
+1 256 824 5331

Office of Information Technology
The University of Alabama in Huntsville
Network Engineering








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Re: [cisco-voip] [External] Re: No Custom Background Images after CUCM 14

2023-03-27 Thread Bill Talley
Do you have a reference link for that?  I’ve never experienced that, let alone read that in upgrade documentation.   I’ve had that problem with earlier versions of PCD but never on a direct standard or refresh upgrade. The upgrade documentation says custom files aren’t contained in DRS backup files.  Maybe I’ve just gotten lucky and the background images have all be accidentally copied 路‍♂️Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.On Mar 27, 2023, at 9:35 AM, Hunter Fuller  wrote:Cisco’s upgrade procedure indicates that custom tftp files are never expected to survive an upgrade. I did notice the files not deleting on my system when I went to replace them after upgrade to 12.5, however, you can upload “on top of” the existing files then delete the .sgn and that seems to work. Unless that changed in 14. On Mon, Mar 27, 2023 at 09:30 Bill Talley <btal...@gmail.com> wrote:I assume you’re saying the images don’t appear under Wallpapers on the phones?  Have you verified the appropriate Desktops directory on the tftp server?  I assume yes to these questions below since TAC appears to directing you towards the sgn files.  Explanation for these questions is below. - List.xml is present and contains all of the correct file names/references?- Background images and thumbnail files are present?I recently did a standard upgrade from cucm 12.5 to 14Su2 and on the pub, the upgrade didn’t copy the custom images to the new partition and List.xml was replaced with the default file so only the default images appeared under wallpapers.    On the sub, the List.xml was copied during the upgrade, but the images were not.   I ended up downloading the original files from the tftp directory on the  inactive  partition the uploading them via OS Admin and restarting tftp.   This happened for two different phone models but not all phone models. Sent from my iPhone with bery tiny touchscreeb input keys, please excude my typtos.On Mar 27, 2023, at 7:26 AM, JASON BURWELL <jason.burw...@foundersfcu.com> wrote:







After we upgraded to CUCM 14 with PCD, our 88XX phones no longer have custom wallpaper images available. The solution from TAC was to delete all the tftpsgn files from the TFTP directory on each node and restart tftp services however when
 we delete the files using the GUI or CLI, they remain and do not delete. TAC has tried twice unsuccessfully to access the directory from root and delete the files but they have not been able to do that. Surely others have encountered this issue and I’m wondering
 what everyone else has done.  I can’t believe it’s really this complicated of an issue to resolve. Thanks Jason
 



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Re: [cisco-voip] No Custom Background Images after CUCM 14

2023-03-27 Thread Bill Talley
I assume you’re saying the images don’t appear under Wallpapers on the phones?  
Have you verified the appropriate Desktops directory on the tftp server?  I 
assume yes to these questions below since TAC appears to directing you towards 
the sgn files.  Explanation for these questions is below. 

- List.xml is present and contains all of the correct file names/references?
- Background images and thumbnail files are present?

I recently did a standard upgrade from cucm 12.5 to 14Su2 and on the pub, the 
upgrade didn’t copy the custom images to the new partition and List.xml was 
replaced with the default file so only the default images appeared under 
wallpapers.On the sub, the List.xml was copied during the upgrade, but the 
images were not.   I ended up downloading the original files from the tftp 
directory on the  inactive  partition the uploading them via OS Admin and 
restarting tftp.   This happened for two different phone models but not all 
phone models. 



Sent from my iPhone with bery tiny touchscreeb input keys, please excude my 
typtos.

> On Mar 27, 2023, at 7:26 AM, JASON BURWELL  
> wrote:
> 
> 
> After we upgraded to CUCM 14 with PCD, our 88XX phones no longer have custom 
> wallpaper images available. The solution from TAC was to delete all the 
> tftpsgn files from the TFTP directory on each node and restart tftp services 
> however when we delete the files using the GUI or CLI, they remain and do not 
> delete. TAC has tried twice unsuccessfully to access the directory from root 
> and delete the files but they have not been able to do that. Surely others 
> have encountered this issue and I’m wondering what everyone else has done.  I 
> can’t believe it’s really this complicated of an issue to resolve. Thanks 
> Jason
>  
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Re: [cisco-voip] Call Flow data via transfers

2022-09-20 Thread Bill Talley
Not to hijack the thread, but which version did Cisco enable CDRs by default in 
CUCM?  I’ve always had to go to cucm service parameters and enable CDRs on 
every node…

Sent from my iPhone with bery tiny touchscreeb input keys, please excude my 
typtos.

> On Sep 20, 2022, at 9:44 AM, Wes Sisk (wsisk)  wrote:
> 
>  Generally speaking there should be adequate info in CDR to track the 
> multiple call legs. There are examples of tracking call flows here:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1_SU7/cucm_b_reporting-billing-administration-guide-1151su7/cucm_b_reporting-billing-administration-guide-1151su7_chapter_01000.html
> 
> Might be interesting to hear from ISI how they interpret the data provided 
> and what gaps exist.
> 
> -w
> 
> On Sep 20, 2022, at 10:27 AM, Scott Voll  wrote:
> 
> So we are on CM 12.5  also using contact center express.
> 
> we use isi for our CDR's.
> 
> is there a way to track call flow from call coming into our contact center 
> and being transferred to hunt groups, or other people, vm - then zero outs, 
> and to other places?
> 
> seems to be a large hole in our data.  Just wondering if there is something 
> i'm missing, or a better application to get this data from?
> 
> Not opposed to changing up applications if needed.  maybe a reception console 
> app could provide this?  I don't think our UCCx is the correct app for our 
> reception desk anyway.
> 
> Thanks
> 
> Scott
> PS. does M$ Teams have this kind of issue also? 
> 
> 
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Re: [cisco-voip] Cisco UCCX Roadmap

2022-09-06 Thread Bill Talley
Haven’t looked into it much, but have quite a few customers who are avoiding 
WxCC due to existing backend integrations with on-prem SQL servers, mainframes, 
and http/REST integrations….  Those are probably being marketed as front end 
widget/gadget integrations for WxCC I’m guessing.  

First I’ve heard that UCCX development is nearly dead, but good to know, 
especially from a personal development perspective lol.

> On Sep 6, 2022, at 7:12 PM, Matthew Loraditch 
>  wrote:
> 
> Wasn’t at Cisco live but at least as far as what’s available to partners, 
> there is nothing that I can’t talk about and publicly there is nothing.
>  
> The on prem software development is quite nearly dead.
>  
>   
> 
> Matthew Loraditch​
> Sr. Network Engineer
> direct: 443.541.1518 
> e: mloradi...@heliontechnologies.com 
> 
>  
>     
>  
>    
> www.heliontechnologies.com 
>  
> From: cisco-voip  > On Behalf Of JASON BURWELL
> Sent: Tuesday, September 6, 2022 7:51 PM
> To: 'cisco-voip@puck.nether.net ' 
> mailto:cisco-voip@puck.nether.net>>
> Subject: [cisco-voip] Cisco UCCX Roadmap
>  
> [EXTERNAL]
>  
> Does anyone that attended Cisco Live have any info on what the future 
> development plans are for UCCX? Is there a v14 and beyond on the roadmap? Any 
> idea how rapidly they are trying to push people to Webex Contact Center? Is 
> anyone on here using Webex Contact Center with 150-200 agents?
>  
>  
> Thanks!!
> Jason
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Re: [cisco-voip] Fun message on CER after power outage...

2022-02-22 Thread Bill Talley
It’s no different than it’s ever been except now Cisco sets a flag and TAC 
looks for that flag. It’s the most closed minded, short sighted policy change I 
ever seen from a technology company.



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

> On Feb 22, 2022, at 3:14 PM, Nick Russo via cisco-voip 
>  wrote:
> 
> 
> This is my favorite new Cisco "feature".  All the 14.x platforms have this.  
> I think they implemented in 12.5(su4 or 5), but if the system goes down 
> dirty, the official Cisco policy is to do a restore from backup.  So far, I 
> haven't had any actual issues after this happens, but if your system is in 
> this state, TAC won't help you until you've rebuilt.  There are a few options 
> for graceful shutdown in VMWare, but your server needs to be on a UPS for 
> them to work.
> 
> On Tuesday, February 22, 2022, 01:01:39 PM PST, Adam Pawlowski 
>  wrote:
> 
> 
> I need to find it, but of course can’t right now, but I swear this is in the 
> ER documentation somewhere, some version, that the system must be 
> re-installed if it shuts down unexpectedly. Outside of this message.
> 
>  
> 
> I got called out on it by TAC, and they have documentation way preceding that 
> message appearing visibly to point out where unexpected shutdowns have 
> occurred.
> 
>  
> 
> There was a back and forth in the community on this, with a lot of people 
> more interesting in making the message go away than trying to figure out if 
> the system was okay or rebuild it.
> 
> I guess in some places it is normal to operate the infrastructure on car 
> batteries or what have you. I really must be missing something as this feels 
> like the old “put tape over the check engine light” .
> 
>  
> 
> Same thing I said there – personally, rebuild the thing when you can on your 
> time table, instead of having it fail later, or during an upgrade.
> 
> CER seems to be the easiest thing to re-install – if you have exports you can 
> replace the system functionality rather quickly even when DRF backups are no 
> good.
> 
>  
> 
> Adam
> 
>  
> 
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Tuesday, February 22, 2022 3:07 PM
> To: Jonathan Charles ; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Fun message on CER after power outage...
> 
>  
> 
> Yes – those warnings were making their way around another forum. It all has 
> to do with reviewing the log files for a corresponding manual shutdown for 
> any bootup sequences.
> 
>  
> 
> There is a fix (?) for the core apps found here. Normally these should work 
> on CER but it’s not listed. Not sure why.
> 
>  
> 
> https://www.cisco.com/web/software/286319173/139477/ciscocm.add_utils_ungraceful_warn_disable_v1.0.cop-README.pdf
> 
>  
> 
> From the readme (above):
> 
>  
> 
> The fix is included natively in 12.5.1SU6 (12.5.1.16900-x) and higher and 
> 14SU2 (14.0.1.12900-x) and higher.
> 
>  
> 
> Note: I don’t think this fixes the issue, just removes the warning. If TAC 
> reviews the logs and finds an ungraceful shutdown, they can start asking you 
> to rebuild.
> 
>  
> 
> PS I don’t see why a system can’t recover without a rebuild either.
> 
>  
> 
> From: cisco-voip  On Behalf Of Jonathan 
> Charles
> Sent: Tuesday, February 22, 2022 2:58 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Fun message on CER after power outage...
> 
>  
> 
> CAUTION: This email originated from outside of the University of Guelph. Do 
> not click links or open attachments unless you recognize the sender and know 
> the content is safe. If in doubt, forward suspicious emails to 
> ith...@uoguelph.ca
> 
>  
> 
> Only on the CLI...
> 
>  
> 
> WARNING: Ungraceful shutdown detected - A rebuild of this node is highly 
> recommended to ensure no negative impact(such as configuration or file system 
> corruption). For rebuild instructions, see the installation guide.
> 
>  
> 
> The bugs suggest it should only be on 12.5 and early... this is on CER 14...
> 
>  
> 
> Any recommendations? This seems a bit extreme...
> 
>  
> 
> The system appears to be working normally.
> 
>  
> 
>  
> 
>  
> 
> Jonathan
> 
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Re: [cisco-voip] UCCX 12.5 Real Time Reporting Tool

2021-11-18 Thread Bill Talley
I was stuck on using JRE 1.7 because of the historical reliance by Cisco on 
that specific version.  After installing JRE 1.8, adding the security exception 
in Java, and restoring security prompts in Java, I was able to reconnect to CCX 
12.5 using the RTR utility you’re asking about.  The specific JRE version I’ve 
been using is 1.8.0_291.



> On Nov 18, 2021, at 12:58 PM, Johnson, Tim  wrote:
> 
> Is anyone using UCCX 12.5 SU1 ES01, and able to use the Real Time Reporting 
> Tool (not RTMT)? I find it to be handy in troubleshooting on rare occasions, 
> but as soon as I upgraded to 12.5 from 12.0 back in August, it stopped 
> working for me on multiple clients. It launches, but whenever you attempt to 
> open any of the reports it just hangs with a Windows spinning wheel. Once or 
> twice, I’ve been able to load one report after I launch the tool but it will 
> stop working if I try to look at a different report.
>  
> I opened a TAC case a little while ago but the engineer took me down all 
> sorts of rabbit holes. I had completely reinstalled Java and it ended up 
> loading but then stopped working again within a day after. I didn’t have the 
> time or patience then to continue with TAC.
>  
> Tim Johnson
> Voice & Video Engineer
> Central Michigan University
> Call me: +19897744406 
> Video Call me: johns...@cmich.edu 
> Fax me: +19897795900
> Meet me: http://cmich.webex.com/meet/johns10t 
> 
>  
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Re: [cisco-voip] E164 Alt Ext and Cisco Unity

2021-10-05 Thread Bill Talley
Yeah it definitely doesn’t scale well, adds administrative overhead, and won’t 
work for direct transfers to voicemail (e.g. user dials * then the 5-digit 
extension to transfer a caller directly to VM) as you’d need a CTI route point 
in each offices local partition assigned to its that specific voicemail profile 
for that, but those are the complications with overlapping dialplans and shared 
voicemail.Sooo many potential complications….   Good luck. 

Hopefully someone else has a cleaner suggestion. 

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

> On Oct 5, 2021, at 4:28 PM, Austin Williams  
> wrote:
> 
> 
> I mis-read your email, I thought you meant to create a VM box. That does work 
> but ya i'm going to have to build one for each site, and some sites are going 
> to have multiple ranges but this will work. I'd still love to see if there is 
> an overall setting that can fix this but this is a great workaround.
> 
>> On Tue, Oct 5, 2021 at 4:16 PM Bill Talley  wrote:
>> My bad, the voicemail box mask field on the voicemail profile in CUCM 
>> ctually won’t allow a \ so it would just be entered on the profile as 
>> +150259X.  
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Oct 5, 2021, at 3:47 PM, Austin Williams  
>>>> wrote:
>>>> 
>>> 
>>> No unity won't take the \ in \+1. we built them with +1 and strip the \ on 
>>> the way over to unity. If you dial the full E164 number of each DN it 
>>> locates the box perfectly and works correctly. Our issue is when you use 
>>> the 5 or 10 digit alternates created on the line. According to Cisco even 
>>> though you build the E164 Alt and Ent Alt directly on the DN in Call 
>>> Manager, Call Manager considers them as unique seperate numbers. So when 
>>> you dial 5 digits that's what gets passed to unity even though it's built 
>>> as an E164 Ent Alt on a DN that is a full \+1 DN. 
>>> 
>>>> On Tue, Oct 5, 2021 at 3:29 PM Bill Talley  wrote:
>>>> Never tried this but can you create voicemail profile with \+1502596X 
>>>> as the voicemail box mask?  If that works you’d have to create one for 
>>>> each site and/or DID/e164 range and assign that to the respective 
>>>> directory numbers.   
>>>> 
>>>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>>>> Please excude my typtos.
>>>> 
>>>>>> On Oct 5, 2021, at 3:19 PM, Austin Williams  
>>>>>> wrote:
>>>>>> 
>>>>> 
>>>>> Hi All, 
>>>>> 
>>>>> We are running into an issue and have sought help from Cisco TAC who has 
>>>>> not been able to help us on this issue. Currently we are running a Full 
>>>>> +E164 numbering Plan with both 5 digit Enterprise Alternate numbers and 
>>>>> +E.164 Alternate numbers. So to give an example of a Test phone I have 
>>>>> set up:
>>>>> 
>>>>> Directory number: \+150259663629
>>>>> Enterprise Alt
>>>>> Number Mask: X
>>>>> Alternate Number: 63629
>>>>> Route Partition: -EALT-PT
>>>>> +E.164 Alternate Number
>>>>> Number Mask: XX  
>>>>> Alternate Number: 5025966329
>>>>> Route Partition: 0-KHC-EntAltExt
>>>>> 
>>>>> When this phone is dialed by either the 5 digit or 10 digit and forwards 
>>>>> to voicemail the redirected and last redirected number that Unity 
>>>>> Receives is the Dialed number, either the 5 digit or 10 digit, and as we 
>>>>> have not configured ALT extensions on the mailboxes it fails to find the 
>>>>> correct box. If I set the Alt ext on each Unity mail box it does function 
>>>>> and I can do that for the 10 digit, I am going to run into an Issue 
>>>>> setting the 5 digit as an Alt extension on the mailboxes as we are going 
>>>>> to overlap on multiple sites. 
>>>>> 
>>>>> What we are trying to do is find a way to tell the call manager to send 
>>>>> the full +1XX as the Last Redirected Number when either the 11 
>>>>> digit full, 10 digit or 5 digit number is dialed. We have tried a Called 
>>>>> Number Transformation to no result and have messed with the device pool 
>>>>> setting  “Redirected transformation CSS” on a test DP to no effect 
>>>>> either. 
>>>>> 
>>>>> Does anyone have any suggestions
>>>>> 
>>>>> CUCM version: 12.5.1.14900-63
>>>>> Unity Version: 12.5.1.14900-45
>>>>> 
>>>>> Our Unity mailboxes are built with +1XX as their D
>>>>> 
>>>>> Thanks,
>>>>> 
>>>>> Austin John Williams
>>>>> 
>>>>> Email: austinwi...@gmail.com
>>>>> 
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Re: [cisco-voip] E164 Alt Ext and Cisco Unity

2021-10-05 Thread Bill Talley
My bad, the voicemail box mask field on the voicemail profile in CUCM ctually 
won’t allow a \ so it would just be entered on the profile as +150259X.  

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

> On Oct 5, 2021, at 3:47 PM, Austin Williams  
> wrote:
> 
> 
> No unity won't take the \ in \+1. we built them with +1 and strip the \ on 
> the way over to unity. If you dial the full E164 number of each DN it locates 
> the box perfectly and works correctly. Our issue is when you use the 5 or 10 
> digit alternates created on the line. According to Cisco even though you 
> build the E164 Alt and Ent Alt directly on the DN in Call Manager, Call 
> Manager considers them as unique seperate numbers. So when you dial 5 digits 
> that's what gets passed to unity even though it's built as an E164 Ent Alt on 
> a DN that is a full \+1 DN. 
> 
>> On Tue, Oct 5, 2021 at 3:29 PM Bill Talley  wrote:
>> Never tried this but can you create voicemail profile with \+1502596X as 
>> the voicemail box mask?  If that works you’d have to create one for each 
>> site and/or DID/e164 range and assign that to the respective directory 
>> numbers.   
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Oct 5, 2021, at 3:19 PM, Austin Williams  
>>>> wrote:
>>>> 
>>> 
>>> Hi All, 
>>> 
>>> We are running into an issue and have sought help from Cisco TAC who has 
>>> not been able to help us on this issue. Currently we are running a Full 
>>> +E164 numbering Plan with both 5 digit Enterprise Alternate numbers and 
>>> +E.164 Alternate numbers. So to give an example of a Test phone I have set 
>>> up:
>>> 
>>> Directory number: \+150259663629
>>> Enterprise Alt
>>> Number Mask: X
>>> Alternate Number: 63629
>>> Route Partition: -EALT-PT
>>> +E.164 Alternate Number
>>> Number Mask: XX  
>>> Alternate Number: 5025966329
>>> Route Partition: 0-KHC-EntAltExt
>>> 
>>> When this phone is dialed by either the 5 digit or 10 digit and forwards to 
>>> voicemail the redirected and last redirected number that Unity Receives is 
>>> the Dialed number, either the 5 digit or 10 digit, and as we have not 
>>> configured ALT extensions on the mailboxes it fails to find the correct 
>>> box. If I set the Alt ext on each Unity mail box it does function and I can 
>>> do that for the 10 digit, I am going to run into an Issue setting the 5 
>>> digit as an Alt extension on the mailboxes as we are going to overlap on 
>>> multiple sites. 
>>> 
>>> What we are trying to do is find a way to tell the call manager to send the 
>>> full +1XX as the Last Redirected Number when either the 11 digit 
>>> full, 10 digit or 5 digit number is dialed. We have tried a Called Number 
>>> Transformation to no result and have messed with the device pool setting  
>>> “Redirected transformation CSS” on a test DP to no effect either. 
>>> 
>>> Does anyone have any suggestions
>>> 
>>> CUCM version: 12.5.1.14900-63
>>> Unity Version: 12.5.1.14900-45
>>> 
>>> Our Unity mailboxes are built with +1XX as their D
>>> 
>>> Thanks,
>>> 
>>> Austin John Williams
>>> 
>>> Email: austinwi...@gmail.com
>>> 
>>> ___
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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[cisco-voip] CCX 12.5 upgrade breaks custom Finesse gadget

2021-06-28 Thread Bill Talley
Hi all,

Has anyone had any experiences in which upgrading from CCX 11.6 (or any 
version) to CCX 12.5 causes any of their Finesse gadgets to stop rendering?  
We’ve encountered this problem.  The same gadget renders just fine on every 
11.6 instance I’ve installed it on, but on every 12.5 instance it fails to 
render in the browser.  We’ve done all the typical upgrade prep tasks, clearing 
the browser cache, etc, with no impact.

Here’s an interesting tidbit, if after agents login, either they clear their 
browser cache while actively logged in, OR we restart the Finesse Tomcat 
service on the CCX server (single node cluster), this gadget immediately 
renders in the agent browser in both scenarios and continues to work for their 
current session.  This process has to be repeated every time the agent logs in. 

Just curious if anyone has seen this, or might be aware of any gadget, XML or 
security changes needed on 12.5 to load gadgets?  I didn’t find any changes 
noted in the 12.5 install or developer docs, and have been unable to find any 
specific 12.5 sample gadgets online in the repository.

TIA everyone,
Bill

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Re: [cisco-voip] Bat.xlt for CUCM 12.5.1

2021-03-30 Thread Bill Talley
Ariel,


Not sure if you received this or not but I unicasted it to you.  Hopefully it 
makes it though the spam/av filters.

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

> On Mar 30, 2021, at 3:30 PM, ROZA, Ariel via cisco-voip 
>  wrote:
> 
> 
> Hi, Guys and gals!
>  
> Does anyone has a bat.xlt file for a cucm 12.5? I am working on a cluster 
> that seem to have it´s file deleted and I am pinging everyone I know before 
> resorting to install whole server for it.
>  
> Regards,
>  
> Ariel
>  
>  
> Ariel Pablo Roza
> Post Sales UC Engineer / Southern Cone
> 
>  
> t: (011) 5282-0458
> m: (011) 5017-4417
> ariel.r...@la.logicalis.com
> Av. Belgrano 955 – Piso 20 - C1092AAJ
> CABA – Buenos Aires - Argentina
> www.la.logicalis.com
>  
> 
> 
>  
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Re: [cisco-voip] List still active?

2020-12-25 Thread Bill Talley
Thanks for the confirmation Ryan.  Are you also seeing a significant decline in 
volume from the group?   

Hope all the usual (and even casual) participants are staying healthy and 
employed.  Hope those aren’t reasons for the decline in forum usage. 

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

> On Dec 25, 2020, at 1:28 PM, Ryan Huff  wrote:
> 
> I still see you.
> 
> Sent from my iPhone
> 
>> On Dec 25, 2020, at 14:28, Bill Talley  wrote:
>> 
>> I stopped receive list emails.   Is the list dead or was I banned? 
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> ___
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Re: [cisco-voip] SFTP on CUCM 11.5.su8

2020-09-04 Thread Bill Talley
I have seen issues after upgrades, matching your symptoms, that are resolved by 
moving the existing backup files to a different folder, so the destination 
directory is empty, then running a new clean backup from the source.

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

> On Sep 4, 2020, at 10:32 AM, Jason Aarons  wrote:
> 
> 
> I am seeing more DRF failures (DRS backups) with 12.5.1SU3 then in the past. 
> My Unity Connection 12.5.1SU3 is getting 50% success. I would suspect a bug. 
> 
>> On Fri, Sep 4, 2020, 11:06 AM NateCCIE  wrote:
>> Have you tried changing the os to permissive?  I think it’s utils os secure 
>> permissive. 
>> 
>> Sent from my iPhone
>> 
 On Sep 4, 2020, at 4:49 AM, Louis Koekemoer (MEA) 
  wrote:
 
>>>  
>>> 
>>> Was wondering if anyone else was experiencing issues with SFTP on 11.5su8. 
>>> I have 5 clusters that was recently upgraded to 12.5su8 and when I want to 
>>> collect any files from them via SFTP it fails. I have done this numerous 
>>> times in my life and also tested with a 12.5su3 instance I have and it 
>>> works, but none of the 11.5su8 servers allows me. I used various different 
>>> servers/PC with FreeFTPD, Solarwinds and Mini SFTP.
>>> 
>>>  
>>> 
>>> Example would be to collect MOH files.
>>> 
>>> 12.5su3
>>> 
>>> admin:file get activelog mohprep/*
>>> 
>>> Please wait while the system is gathering files info ...
>>> 
>>> Get file: active/mohprep/CiscoMOHSourceReport.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.xml
>>> 
>>> done.
>>> 
>>> Sub-directories were not traversed.
>>> 
>>> Number of files affected: 16
>>> 
>>> Total size in Bytes: 18537609
>>> 
>>> Total size in Kbytes: 18103.133
>>> 
>>> Would you like to proceed [y/n]? y
>>> 
>>> SFTP server IP: 23.240.48.250
>>> 
>>> SFTP server port [22]: 21
>>> 
>>> User ID: ccmadmin
>>> 
>>> Password: 
>>> 
>>> Download directory: /
>>> 
>>>  
>>> 
>>> ...
>>> 
>>> Transfer completed.
>>> 
>>> admin:
>>> 
>>>  
>>> 
>>> 11.5su8
>>> 
>>> admin:file get activelog mohprep/*
>>> 
>>> Please wait while the system is gathering files info ...
>>> 
>>> Get file: active/mohprep/CiscoMOHSourceReport.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SampleAudioSource.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/SilenceAudioSource.xml
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.alaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.g729.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.ulaw.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.wb.wav
>>> 
>>>  
>>> 
>>> Get file: active/mohprep/ToneOnHold.xml
>>> 
>>> done.
>>> 
>>> Sub-directories were not traversed.
>>> 
>>> Number of files affected: 16
>>> 
>>> Total size in Bytes: 18537609
>>> 
>>> Total size in Kbytes: 18103.133
>>> 
>>> Would you like to proceed [y/n]? y
>>> 
>>> SFTP server IP: 23.240.48.250
>>> 
>>> SFTP server port [22]: 22
>>> 
>>> User ID: ccmadmin
>>> 
>>> Password: 
>>> 
>>> Download directory: /
>>> 
>>>  
>>> 
>>> Could not connect to host 23.240.48.250 on port 22. Please verify SFTP 
>>> settings.
>>> 
>>> admin:
>>> 
>>> 
>>> 
>>> This email and all contents are subject to the following disclaimer:
>>> "http://www.dimensiondata.com/emaildisclaimer; 
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Re: [cisco-voip] SFTP on CUCM 11.5.su8

2020-09-04 Thread Bill Talley
Is there a reason you connected to different destination ports?   The good 
attempt used port 21 and the bad attempt used port 22.

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

> On Sep 4, 2020, at 5:49 AM, Louis Koekemoer (MEA) 
>  wrote:
> 
>  
> 
> Was wondering if anyone else was experiencing issues with SFTP on 11.5su8. I 
> have 5 clusters that was recently upgraded to 12.5su8 and when I want to 
> collect any files from them via SFTP it fails. I have done this numerous 
> times in my life and also tested with a 12.5su3 instance I have and it works, 
> but none of the 11.5su8 servers allows me. I used various different 
> servers/PC with FreeFTPD, Solarwinds and Mini SFTP.
>  
> Example would be to collect MOH files.
> 12.5su3
> admin:file get activelog mohprep/*
> Please wait while the system is gathering files info ...
> Get file: active/mohprep/CiscoMOHSourceReport.xml
>  
> Get file: active/mohprep/SampleAudioSource.alaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.g729.wav
>  
> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.wb.wav
>  
> Get file: active/mohprep/SampleAudioSource.xml
>  
> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.g729.wav
>  
> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.wb.wav
>  
> Get file: active/mohprep/SilenceAudioSource.xml
>  
> Get file: active/mohprep/ToneOnHold.alaw.wav
>  
> Get file: active/mohprep/ToneOnHold.g729.wav
>  
> Get file: active/mohprep/ToneOnHold.ulaw.wav
>  
> Get file: active/mohprep/ToneOnHold.wb.wav
>  
> Get file: active/mohprep/ToneOnHold.xml
> done.
> Sub-directories were not traversed.
> Number of files affected: 16
> Total size in Bytes: 18537609
> Total size in Kbytes: 18103.133
> Would you like to proceed [y/n]? y
> SFTP server IP: 23.240.48.250
> SFTP server port [22]: 21
> User ID: ccmadmin
> Password: 
> Download directory: /
>  
> ...
> Transfer completed.
> admin:
>  
> 11.5su8
> admin:file get activelog mohprep/*
> Please wait while the system is gathering files info ...
> Get file: active/mohprep/CiscoMOHSourceReport.xml
>  
> Get file: active/mohprep/SampleAudioSource.alaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.g729.wav
>  
> Get file: active/mohprep/SampleAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SampleAudioSource.wb.wav
>  
> Get file: active/mohprep/SampleAudioSource.xml
>  
> Get file: active/mohprep/SilenceAudioSource.alaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.g729.wav
>  
> Get file: active/mohprep/SilenceAudioSource.ulaw.wav
>  
> Get file: active/mohprep/SilenceAudioSource.wb.wav
>  
> Get file: active/mohprep/SilenceAudioSource.xml
>  
> Get file: active/mohprep/ToneOnHold.alaw.wav
>  
> Get file: active/mohprep/ToneOnHold.g729.wav
>  
> Get file: active/mohprep/ToneOnHold.ulaw.wav
>  
> Get file: active/mohprep/ToneOnHold.wb.wav
>  
> Get file: active/mohprep/ToneOnHold.xml
> done.
> Sub-directories were not traversed.
> Number of files affected: 16
> Total size in Bytes: 18537609
> Total size in Kbytes: 18103.133
> Would you like to proceed [y/n]? y
> SFTP server IP: 23.240.48.250
> SFTP server port [22]: 22
> User ID: ccmadmin
> Password: 
> Download directory: /
>  
> Could not connect to host 23.240.48.250 on port 22. Please verify SFTP 
> settings.
> admin:
> 
> 
> This email and all contents are subject to the following disclaimer:
> "http://www.dimensiondata.com/emaildisclaimer; 
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Re: [cisco-voip] [External] That's some nice quality control...

2020-08-26 Thread Bill Talley
Don’t beat yourself up too bad.  At first I thought the Cisco label on the top 
of the phone said Cifco.



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

> On Aug 26, 2020, at 10:59 AM, Hunter Fuller  wrote:
> 
> It is absolutely embarrassing how long it took me to figure out what
> the problem was.
> 
> --
> Hunter Fuller (they)
> Router Jockey
> VBH Annex B-5
> +1 256 824 5331
> 
> Office of Information Technology
> The University of Alabama in Huntsville
> Network Engineering
> 
>> On Wed, Aug 26, 2020 at 10:07 AM Balk, David  wrote:
>> 
>> 
>> 
>> 
>> 
>> David Balk
>> IS Team Lead
>> Voice Services
>> 
>> Northwestern Memorial Health Care
>> 
>> 541 N. Fairbanks Court, Suite 2401
>> Chicago, Illinois 60611
>> 312.926.2642 office
>> 
>> db...@nm.org
>> northwesternmedicine.org
>> 
>> 
>> 
>> 
>> This message and any included attachments are intended only for the 
>> addressee. The information contained in this message is confidential and may 
>> constitute proprietary or non-public information under international, 
>> federal, or state laws. Unauthorized forwarding, printing, copying, 
>> distribution, or use of such information is strictly prohibited and may be 
>> unlawful. If you are not the addressee, please promptly delete this message 
>> and notify the sender of the delivery error by e-mail.
>> 
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Re: [cisco-voip] [External] IPCC best practice

2020-08-19 Thread Bill Talley
+1000



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

> On Aug 19, 2020, at 12:51 PM, Anthony Holloway 
>  wrote:
> 
> 
> GTFO
> 
>> On Wed, Aug 19, 2020 at 10:25 AM NateCCIE  wrote:
>> What about IP IVR?
>> 
>> Sent from my iPhone
>> 
 On Aug 19, 2020, at 9:16 AM, Lelio Fulgenzi  wrote:
 
>>> 
>>> +100 for Anthony! 
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> From: Anthony Holloway  
>>> Sent: Wednesday, August 19, 2020 10:48 AM
>>> To: Lelio Fulgenzi 
>>> Cc: Matthew Loraditch ; Charles 
>>> Goldsmith ; cisco-voip@puck.nether.net
>>> Subject: Re: [cisco-voip] [External] IPCC best practice
>>> 
>>>  
>>> 
>>> CAUTION: This email originated from outside of the University of Guelph. Do 
>>> not click links or open attachments unless you recognize the sender and 
>>> know the content is safe. If in doubt, forward suspicious emails to 
>>> ith...@uoguelph.ca
>>> 
>>>  
>>> 
>>> Wait Lelio, CRA is older terminology than CRS, so it should go:
>>> 
>>>  
>>> 
>>> +1 IPCC
>>> 
>>> +2 CRS
>>> 
>>> +3 CRA
>>> 
>>>  
>>> 
>>> Right?
>>> 
>>>  
>>> 
>>> On Wed, Aug 19, 2020 at 8:53 AM Lelio Fulgenzi  wrote:
>>> 
>>>  
>>> 
>>> Not much more to add here, except +1 for calling in IPCC. :) you’d have 
>>> gotten +2 if you called it CRA. ;)
>>> 
>>>  
>>> 
>>> But, seriously, you have to weigh the pros and cons of injecting a point of 
>>> failure vs ease of administration. 
>>> 
>>>  
>>> 
>>> My thought process is, can you build automatic recovery? Or easily 
>>> understood manual backup. 
>>> 
>>>  
>>> 
>>> And is the design something you can easily hand off to someone?
>>> 
>>>  
>>> 
>>> Lots of things to consider. 
>>> 
>>>  
>>> 
>>> Sent from my iPhone
>>> 
>>> 
>>> 
>>> 
>>> On Aug 19, 2020, at 9:00 AM, Matthew Loraditch 
>>>  wrote:
>>> 
>>> 
>>> 
>>> CAUTION: This email originated from outside of the University of Guelph. Do 
>>> not click links or open attachments unless you recognize the sender and 
>>> know the content is safe. If in doubt, forward suspicious emails to 
>>> ith...@uoguelph.ca
>>> 
>>>  
>>> 
>>> We still use Call Handlers. We have fewer resources who can handle script 
>>> editing and somewhat frequent requests to change hours and such that we 
>>> need the regular techs to be able to handle.
>>> 
>>>  
>>> 
>>> Definitely a preference thing.
>>> 
>>>  
>>> 
>>>  
>>> 
>>> Matthew Loraditch​
>>> 
>>> Sr. Network Engineer
>>> 
>>> p: 443.541.1518
>>> 
>>> w: www.heliontechnologies.com
>>> 
>>>  | 
>>> 
>>> e: mloradi...@heliontechnologies.com
>>> 
>>> 
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>> From: cisco-voip  On Behalf Of Charles 
>>> Goldsmith
>>> Sent: Wednesday, August 19, 2020 8:39 AM
>>> To: Johnson, Tim 
>>> Cc: cisco-voip@puck.nether.net
>>> Subject: Re: [cisco-voip] [External] IPCC best practice
>>> 
>>>  
>>> 
>>> [EXTERNAL]
>>> 
>>>  
>>> 
>>> Agreed with TIm, it's just simpler to involve less systems if you can.  
>>> With 12.0 UCCX and higher, the calendar function is a nice addition, no 
>>> more XML files for schedules.
>>> 
>>>  
>>> 
>>> On Wed, Aug 19, 2020 at 7:37 AM Johnson, Tim  wrote:
>>> 
>>> It seems to me that there's not a "best practice" label for most scenarios. 
>>> When I started with UCCX, we went to a call handler first to provide us 
>>> with an easy way to provide a schedule, and a familiar way for the customer 
>>> to record a greeting. Later, we ended up building the schedule into our 
>>> script and directing calls to the trigger. That's my preference, just to 
>>> involve less systems. 
>>> 
>>> Tim Johnson
>>> Voice & Video Engineer
>>> Central Michigan University
>>> Call me: +19897744406
>>> Video Call me: johns...@cmich.edu
>>> Fax me: +19897795900
>>> Meet me: http://cmich.webex.com/meet/johns10t
>>> 
>>> 
>>> -Original Message-
>>> From: cisco-voip  On Behalf Of 
>>> f...@browardcommunications.com
>>> Sent: Wednesday, August 19, 2020 8:19 AM
>>> To: cisco-voip@puck.nether.net
>>> Subject: [External] [cisco-voip] IPCC best practice
>>> 
>>> 
>>> Hello, I just have a quick question.
>>> When setting up a call center for a SMB, Is it best practice to have the 
>>> main number go to a unity call handler 1st, with caller input going to uccx 
>>> triggers, or is it considered best practice to have the main number go 
>>> right to CCX?  I have seen both ways.
>>> 
>>> Thank you.
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>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> ___
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> 
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>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> 
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>>> 

Re: [cisco-voip] [EXTERNAL] Re: UCCX 11.6 Real Time Port Usage

2020-07-10 Thread Bill Talley
Maybe I’m overlooking something.   Where in that data is the original challenge 
fulfilled, at least without performing multiple queries to determine the 
parameter from every active call?

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

> On Jul 10, 2020, at 5:59 PM, Tanner Ezell  wrote:
> 
> 
> The challenge was only to indicate which ports were in use, but I thought 
> it'd be more fun to also include details about the caller, or application 
> information. All data is pretty much available (we could list what queues 
> they're in if we wanted to), just a matter of what information is valuable.
> 
> Imagine a dashboard that fired off when a counter variable exceeded a certain 
> value within a running script; VIP caller is identified by an ECC variable 
> while in queue and is manually handled by an agent or supervisor (cherry 
> pick); Imagine troubleshooting a callers actual call flow, replaying their 
> experience step by step, seeing variable values change with each ste; Code 
> coverage testing, automated application testing.. lots of fun stuff we can do.
> 
>> On Fri, Jul 10, 2020 at 3:34 PM Bill Talley  wrote:
>> Getting data is easy, getting the right data, not so easy 
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Jul 10, 2020, at 5:21 PM, Tanner Ezell  wrote:
>>>> 
>>> 
>>> Getting data is easy, giving you presentation is a bit more challenging... 
>>> (sanitized)
>>> 
>>>  "1103010": {
>>> "state": "In Service",
>>> "ccgId": "6"
>>>   },
>>>   "1103011": {
>>> "state": "In Service",
>>> "contactDetails": {
>>>   "callingNumber": "removed",
>>>   "calledNumber": "removed",
>>>   "originalDialedNumber": "null",
>>>   "arrivalType": "2",
>>>   "CLID": "null",
>>>   "DNIS": "null",
>>>   "lastRedirectedNumber": "null",
>>>   "eccDataMap": {
>>> "SCRIPTCFG": "null",
>>> "ACCOUNT_NUMBER": "null",
>>> "CALLVAR9": "null",
>>> "ANI": "null",
>>> "CALLER_ENTERED_DIGITS": "null",
>>> "SCRIPTID": "null",
>>> "CALLVAR7": "null",
>>> "CALLVAR8": "null",
>>> "CALLVAR5": "null",
>>> "CALLVAR6": "null",
>>> "CALLVAR10": "null",
>>> "CALLVAR3": "null",
>>> "WRAPUP_DATA": "null",
>>> "CALLVAR4": "null",
>>> "CALLVAR1": "null",
>>> "CALLVAR2": "null"
>>>   },
>>>   "ANIIIDIGITS": "null",
>>>   "RDNIS": "null",
>>>   "ANI": "null",
>>>   "originalCalledNumber": "removed",
>>>   "origProtocolCallRef": "00355E9B065D6C2A",
>>>   "destProtocolCallRef": "null",
>>>   "CED": "null",
>>>   "dialedNumber": "removed"
>>> },
>>> "scriptVarsMap": {
>>>   "sAgents": {
>>> "name": "sAgents",
>>> "value": "\"2286854\"",
>>> "type": "java.lang.String"
>>>   },
>>>   "pTerminalMenu": {
>>> "name": "pTerminalMenu",
>>> "value": "P[6886/688601.wav]",
>>> "type": "com.cisco.prompt.Playable"
>>>   },
>>>   "svoicemail": {
>>> "name": "svoicemail",
>>> "value": "\"2286856\"",
>>> "type": "java.lang.String"
>>>   }
>>> },
>>> "ccgId": "6&quo

Re: [cisco-voip] [EXTERNAL] Re: UCCX 11.6 Real Time Port Usage

2020-07-10 Thread Bill Talley
Getting data is easy, getting the right data, not so easy 

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

> On Jul 10, 2020, at 5:21 PM, Tanner Ezell  wrote:
> 
> 
> Getting data is easy, giving you presentation is a bit more challenging... 
> (sanitized)
> 
>  "1103010": {
> "state": "In Service",
> "ccgId": "6"
>   },
>   "1103011": {
> "state": "In Service",
> "contactDetails": {
>   "callingNumber": "removed",
>   "calledNumber": "removed",
>   "originalDialedNumber": "null",
>   "arrivalType": "2",
>   "CLID": "null",
>   "DNIS": "null",
>   "lastRedirectedNumber": "null",
>   "eccDataMap": {
> "SCRIPTCFG": "null",
> "ACCOUNT_NUMBER": "null",
> "CALLVAR9": "null",
> "ANI": "null",
> "CALLER_ENTERED_DIGITS": "null",
> "SCRIPTID": "null",
> "CALLVAR7": "null",
> "CALLVAR8": "null",
> "CALLVAR5": "null",
> "CALLVAR6": "null",
> "CALLVAR10": "null",
> "CALLVAR3": "null",
> "WRAPUP_DATA": "null",
> "CALLVAR4": "null",
> "CALLVAR1": "null",
> "CALLVAR2": "null"
>   },
>   "ANIIIDIGITS": "null",
>   "RDNIS": "null",
>   "ANI": "null",
>   "originalCalledNumber": "removed",
>   "origProtocolCallRef": "00355E9B065D6C2A",
>   "destProtocolCallRef": "null",
>   "CED": "null",
>   "dialedNumber": "removed"
> },
> "scriptVarsMap": {
>   "sAgents": {
> "name": "sAgents",
> "value": "\"2286854\"",
> "type": "java.lang.String"
>   },
>   "pTerminalMenu": {
> "name": "pTerminalMenu",
> "value": "P[6886/688601.wav]",
> "type": "com.cisco.prompt.Playable"
>   },
>   "svoicemail": {
> "name": "svoicemail",
> "value": "\"2286856\"",
> "type": "java.lang.String"
>   }
> },
> "ccgId": "6"
>   },
> [CLIPPED]
> and if you're wondering, yes, those are real-time insights into the script 
> variables and caller ECC. I could tell you the step they're currently on.. ;)
> 
> It's too bad there isn't a market for these tools, lots of fun stuff we can 
> do.
> 
>> On Fri, Jul 10, 2020 at 1:42 PM Bill Talley  wrote:
>> I ran some tests, and as Anthony suggested, there is no data returned which 
>> indicates anything beyond the registration status of a device.   路‍♂️
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Jul 10, 2020, at 2:56 PM, JASON BURWELL via cisco-voip 
>>>>  wrote:
>>>> 
>>> 
>>> Thank you for all the responses! Been a busy day so late getting back.
>>> 
>>>  
>>> 
>>> I was able to see the data I needed in historical format by running the 
>>> licensing report shown in the thread Anthony posted. Very high level but 
>>> gives the overall numbers. I wish there was a way to monitor this real time 
>>> and in detail without having to do a lot of custom work which, unless I 
>>> missed something, sounds like what would need to happen.
>>> 
>>>  
>>> 
>>> RTMT does show CTI ports but only shows IN/OUT of service status, not what 
>>> the port is actually doing. I’ve long wondered when a refresh was coming to 
>>> RTMT with more functionality as it feels a bit outdated and seems like its 
>>> been essentially unchanged as far back as I can remember. Although maybe 
>>> the newer versions have improvements I am not aware of?
>>> 
>

Re: [cisco-voip] [EXTERNAL] Re: UCCX 11.6 Real Time Port Usage

2020-07-10 Thread Bill Talley
I ran some tests, and as Anthony suggested, there is no data returned which 
indicates anything beyond the registration status of a device.   路‍♂️

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

> On Jul 10, 2020, at 2:56 PM, JASON BURWELL via cisco-voip 
>  wrote:
> 
> 
> Thank you for all the responses! Been a busy day so late getting back.
>  
> I was able to see the data I needed in historical format by running the 
> licensing report shown in the thread Anthony posted. Very high level but 
> gives the overall numbers. I wish there was a way to monitor this real time 
> and in detail without having to do a lot of custom work which, unless I 
> missed something, sounds like what would need to happen.
>  
> RTMT does show CTI ports but only shows IN/OUT of service status, not what 
> the port is actually doing. I’ve long wondered when a refresh was coming to 
> RTMT with more functionality as it feels a bit outdated and seems like its 
> been essentially unchanged as far back as I can remember. Although maybe the 
> newer versions have improvements I am not aware of?
>  
> Jason
>  
>  
> From: cisco-voip  On Behalf Of Anthony 
> Holloway
> Sent: Friday, July 10, 2020 2:03 PM
> To: Tanner Ezell 
> Cc: Charles Goldsmith ; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] [EXTERNAL] Re: UCCX 11.6 Real Time Port Usage
>  
> Looks like this has been asked and answered in the past:
>  
> https://community.cisco.com/t5/contact-center/cucm-uccx-how-monitoring-cti-ports/td-p/2328292
>  
> The two people responding seem familiar to me, but I can't quite put my 
> finger on who they are.
>  
> On Fri, Jul 10, 2020 at 11:55 AM Tanner Ezell  wrote:
> 
> I'll see what I can do.
>  
> On Fri, Jul 10, 2020 at 9:45 AM UC Penguin  wrote:
> It’s been a long time since I’ve used uccx as uccx.  Is the option for real 
> time reporting present under the Tools menu? (It is when licensed as IP IVR)
>  
> It requires Java and is finicky, but does report.
>  
> In CCE instead I just look at the usage on the AW and dump that in AW Db and 
> graph it with Grafana. 
> 
> 
> On Jul 10, 2020, at 10:58, JASON BURWELL via cisco-voip 
>  wrote:
> 
> 
> Sorry, been tied us this morning. Just looking for real time usage data of 
> the 300 UCCX Ports we are licensed for. Thanks!
>  
>  
> From: Tanner Ezell  
> Sent: Friday, July 10, 2020 9:41 AM
> To: Charles Goldsmith 
> Cc: Anthony Holloway ; JASON BURWELL 
> ; cisco-voip@puck.nether.net
> Subject: [EXTERNAL] Re: [cisco-voip] UCCX 11.6 Real Time Port Usage
>  
> CAUTION: This email originated outside of Founders Federal Credit Union. Do 
> not click links or open attachments unless you recognize the sender and know 
> the content is safe.
> What information do you need?
>  
> On Thu, Jul 9, 2020 at 8:13 PM Charles Goldsmith  wrote:
> You can simply put Tanner in the To: field, old school I know, but it still 
> works :)
>  
> On Thu, Jul 9, 2020 at 4:46 PM Anthony Holloway 
>  wrote:
> That's nothing I've ever heard of.  I'd imagine you could use the CTI API, 
> but not the Admin API.
>  
> This isn't a REST based API though, and it is relatively harder to implement 
> and work with though.  My man Tanner at CTI Logic should be able to help.  Yo 
> Tanner! Where you at?  Ok, so one PRO for chat rooms are mentions.  Email 
> needs mentions.
> 
> The CTI Protocol:
>  
> Is a TCP/IP socket based message protocol
> Allows clients to send and receive information/events about:
> Current system configuration and future updates.
> Agents and their states
> Calls and their states
> Statistics for agents, calls, and queues on a real-time basis
> Third-party call control
> Device snapshots
> Provides support for two client modes for connecting with Unified CCX:
> Bridge mode clients receive all agent-state and call events for all logged in 
> agents in the system.
> Agent mode clients only receives messages related to the agent.
> Has version control
>  
> On Thu, Jul 9, 2020 at 4:07 PM JASON BURWELL via cisco-voip 
>  wrote:
> Is there any way to see real time CTI port usage with UCCX Admin API? I did a 
> quick search and it looks like it’s a supported function but having trouble 
> finding the correct name to use.
>  
> Thanks
> Jason
>  
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> ___
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> https://puck.nether.net/mailman/listinfo/cisco-voip
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> ___
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Re: [cisco-voip] UCCX 11.6 Real Time Port Usage

2020-07-10 Thread Bill Talley
I believe RIS should but I will test and reply back to the list. 

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

> On Jul 10, 2020, at 10:30 AM, Anthony Holloway 
>  wrote:
> 
> 
> Bill, just to be clear, do either of the CUCM APIs show the active call and 
> idle status of CTI ports?  Not just registration.
> 
>> On Fri, Jul 10, 2020 at 10:05 AM Bill Talley  wrote:
>> 
>> Do you need to gather that data directly from CCX?  It might be easier to 
>> pull it from CCM using the PerfMon API or RisPort70 API.
>> 
>> 
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Jul 9, 2020, at 4:07 PM, JASON BURWELL via cisco-voip 
>>>>  wrote:
>>>> 
>>> 
>>> Is there any way to see real time CTI port usage with UCCX Admin API? I did 
>>> a quick search and it looks like it’s a supported function but having 
>>> trouble finding the correct name to use.
>>> 
>>>  
>>> 
>>> Thanks
>>> 
>>> Jason
>>> 
>>>  
>>> 
>>> ___
>>> 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 11.6 Real Time Port Usage

2020-07-10 Thread Bill Talley

Do you need to gather that data directly from CCX?  It might be easier to pull 
it from CCM using the PerfMon API or RisPort70 API.



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

>> On Jul 9, 2020, at 4:07 PM, JASON BURWELL via cisco-voip 
>>  wrote:
> 
> Is there any way to see real time CTI port usage with UCCX Admin API? I did a 
> quick search and it looks like it’s a supported function but having trouble 
> finding the correct name to use.
>  
> Thanks
> Jason
>  
> ___
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] Renewing Expressway E Cert

2020-04-17 Thread Bill Talley
I agree with all of potential solutions.   I’ve just come across a few isolated 
scenarios were we had to have the EXPC cert signed by a public CA since there 
were no alternatives (right, wrong or indifferent) and Let’s Encrypt isn’t 
available for the EXPC nodes (at least not using automated renewal). 


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

> On Apr 17, 2020, at 4:27 PM, Anthony Holloway 
>  wrote:
> 
> 
> Well, that depends.  And let me just ask, why did they do it this way?  If it 
> was even a self-signed cert, we could atleast import it to E, but it's not 
> even that. It's some invalid bogus cert in there.  Why?
> 
> I have seen the following:
> 
> 1. publicly sign it (name cheap has dirt cheap certs) 
> 2. get a private ca installed because just like you need a network, a server, 
> licensing, phones, an internet connection, etc.  it's apart of the solution
> 3. sign it yourself with any ca you want to include the one running on your 
> home computer, and just don't tell anyone what you did because you setup it 
> for
>  34 years and it wont matter by then anyway (ok, just kidding here...or am I?)
> 
>> On Fri, Apr 17, 2020 at 3:55 PM Bill Talley  wrote:
>> Great info Anthony, thanks.
>> 
>> Question, what do you do for Expressway Core if you don’t have an internal 
>> CA to sign the EXPC (meaning no internal root cert to upload to EXPE to 
>> establish the traversal zone trust)?
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>>> On Apr 17, 2020, at 3:25 PM, Anthony Holloway 
>>>>  wrote:
>>>> 
>>> 
>>> This might be an unpopular opinion, but I think using the free certs 
>>> provided by let's encrypt, coupled with it being automatic from now on, 
>>> it's just an unbeatable combination.
>>> 
>>> Here are my cliff notes:
>>> 
>>> Reference Document:
>>> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-5/exwy_b_certificate-creation-use-deployment-guide/exwy_b_certificate-creation-use-deployment-guide_chapter_0100.html
>>>  
>>> High Level Steps:
>>> Expressway 12.5.7 to avoid ACMEv1 vs ACMEv2 registration issues 
>>> (https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvr82346)
>>> For your Unified CM registrations domains don’t use parent domain only 
>>> (E.g., company.com), switch to CollabEdgeDNS format instead (E.g., 
>>> collab-edge.company.com), because you’ll need that in the next step
>>> DNS A records for the Expressway-E FQDN and the CM registration domains
>>> Upload the root and intermediates for Let’s Encrypt (needed on both 
>>> Expressway-E and Expressway-C) (certs are linked in documentation)
>>> Enable the ACME client on Expressway-E and supply any email address you 
>>> want to link to this registration (This creates your account with Let’s 
>>> Encrypt)
>>> Generate a new CSR (Server Certificate Only, Domain Cert Was Not Needed)
>>> Click button to Submit CSR to ACME
>>> Click button to Deploy New Certificate on Expressway-E (documentation 
>>> states this is non-service impacting)
>>> Setup the automatic scheduler so you never have to deal with this again
>>> Sit back, relax and enjoy free shit
>>> 
>>> 
>>> 
>>>> On Fri, Apr 17, 2020 at 1:43 PM Riley, Sean  
>>>> wrote:
>>>> We had our Cisco partner setup our Expressways a couple of years ago.  It 
>>>> is a cluster with 2 E’s and 2 C’s currently at v 12.5.7 using for MRA.  I 
>>>> have been managing them, installing updates, troubleshooting etc.  The 
>>>> public Edge cert is up for renewal.  Can anyone provide advice on renewing 
>>>> this cert?  I am planning on just renewing with the same cert provider, 
>>>> but was interested in if there is anything to watch out for.  Example, 
>>>> will there be a service interruption when replacing the cert?  Or just 
>>>> install the new cert/pk and rest easy?
>>>> 
>>>>  
>>>> 
>>>> Thanks in advance.
>>>> 
>>>>  
>>>> 
>>>> Sean.
>>>> 
>>>> ___
>>>> cisco-voip mailing list
>>>> cisco-voip@puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] Renewing Expressway E Cert

2020-04-17 Thread Bill Talley
Great info Anthony, thanks.

Question, what do you do for Expressway Core if you don’t have an internal CA 
to sign the EXPC (meaning no internal root cert to upload to EXPE to establish 
the traversal zone trust)?

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

> On Apr 17, 2020, at 3:25 PM, Anthony Holloway 
>  wrote:
> 
> 
> This might be an unpopular opinion, but I think using the free certs provided 
> by let's encrypt, coupled with it being automatic from now on, it's just an 
> unbeatable combination.
> 
> Here are my cliff notes:
> 
> Reference Document:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-5/exwy_b_certificate-creation-use-deployment-guide/exwy_b_certificate-creation-use-deployment-guide_chapter_0100.html
>  
> High Level Steps:
> Expressway 12.5.7 to avoid ACMEv1 vs ACMEv2 registration issues 
> (https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvr82346)
> For your Unified CM registrations domains don’t use parent domain only (E.g., 
> company.com), switch to CollabEdgeDNS format instead (E.g., 
> collab-edge.company.com), because you’ll need that in the next step
> DNS A records for the Expressway-E FQDN and the CM registration domains
> Upload the root and intermediates for Let’s Encrypt (needed on both 
> Expressway-E and Expressway-C) (certs are linked in documentation)
> Enable the ACME client on Expressway-E and supply any email address you want 
> to link to this registration (This creates your account with Let’s Encrypt)
> Generate a new CSR (Server Certificate Only, Domain Cert Was Not Needed)
> Click button to Submit CSR to ACME
> Click button to Deploy New Certificate on Expressway-E (documentation states 
> this is non-service impacting)
> Setup the automatic scheduler so you never have to deal with this again
> Sit back, relax and enjoy free shit
> 
> 
> 
>> On Fri, Apr 17, 2020 at 1:43 PM Riley, Sean  
>> wrote:
>> We had our Cisco partner setup our Expressways a couple of years ago.  It is 
>> a cluster with 2 E’s and 2 C’s currently at v 12.5.7 using for MRA.  I have 
>> been managing them, installing updates, troubleshooting etc.  The public 
>> Edge cert is up for renewal.  Can anyone provide advice on renewing this 
>> cert?  I am planning on just renewing with the same cert provider, but was 
>> interested in if there is anything to watch out for.  Example, will there be 
>> a service interruption when replacing the cert?  Or just install the new 
>> cert/pk and rest easy?
>> 
>>  
>> 
>> Thanks in advance.
>> 
>>  
>> 
>> Sean.
>> 
>> ___
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>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] UCCX 11.6 Real Time Logged in Resources

2020-03-17 Thread Bill Talley
Several ways:

-  the Real Time Reporting Tool in appadmin
- CCx API website (https://ccxip/adminapi/agentstats
- CUIC Live reports
- Finesse Supervisor Desktop assuming supervisor is a supervisor of all queues.

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

> On Mar 17, 2020, at 11:19 AM, JASON BURWELL via cisco-voip 
>  wrote:
> 
> 
> On UCCX 11.6 is there a way to quickly see how many agents are logged in real 
> time? TAC was not helpful with providing a solution. Thanks
>  
> Jason
>  
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Re: [cisco-voip] gathering and offloading realtime stats (from RTMT)

2020-03-17 Thread Bill Talley
If you have access to the Cisco Live on demand repository, there’s a 
presentation and video by, I think, Paul Giralt, about using APIs to collect 
the RTMT data you’re asking about.  He mentions sharing his source files on 
GitHub.I don’t remember the session ID at the moment, sorry. 

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

> On Mar 17, 2020, at 10:10 AM, Lelio Fulgenzi  wrote:
> 
> 
> Thanks Kent.
> 
> While a script sounds lovely (and please do share), I'm currently focused on 
> any way to dump the data, even to a text file, from RTMT, using RTMT 
> configurations.
> 
> If the only way to get this is from a script, then, well, guess I'm 
> scripting. 
> 
> 
> -Original Message-
> From: Kent Roberts  
> Sent: Tuesday, March 17, 2020 11:00 AM
> To: Pawlowski, Adam 
> Cc: Lelio Fulgenzi ; voyp list, cisco-voip 
> (cisco-voip@puck.nether.net) 
> Subject: Re: [cisco-voip] gathering and offloading realtime stats (from RTMT)
> 
> Wow lots of interest ok. Let me collect all the stuff 
> 
> It’s just a script. I have been thinking of making it a windows service for 
> better stability if there is interest.  But that will take a few days 
> everything is a mess with the virus response requests
> 
> 
> Kent
> 
>> On Mar 17, 2020, at 06:25, Pawlowski, Adam  wrote:
>> 
>> This would be cool and for sure something to work on when/if I get stuck in 
>> the home office. For now I've just been hastily adding code to my other 
>> scripts and libraries to bulk provision everyone for Jabber while trying not 
>> to implode the system, but that's about done.
>> 
>> 
>> 
>>> -Original Message-
>>> From: cisco-voip  On Behalf Of 
>>> Lelio Fulgenzi
>>> Sent: Tuesday, March 17, 2020 7:27 AM
>>> To: Kent Roberts 
>>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) >> v...@puck.nether.net>
>>> Subject: Re: [cisco-voip] gathering and offloading realtime stats 
>>> (from RTMT)
>>> 
>>> 
>>> Yes please! Thanks.
>>> 
>>> We are still using mrtg, but it should be fine for now.
>>> 
>>> Sent from my iPhone
>>> 
>> On Mar 16, 2020, at 8:21 PM, Kent Roberts  wrote:
> 
> Yes there is   I can send you links and stuff I’ve done this for years 
> and use
>>> grafania to graph it
 
 
 Kent
 
> On Mar 16, 2020, at 16:24, Lelio Fulgenzi  wrote:
> 
> 
> Is there a way I can start dumping real-time stats from CUCM? We're
>>> moving heavily to remote working in light of what's going on and I'd 
>>> like to keep an eye on things outside of SNMP polling.
> 
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> ___
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>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] Uccx maximum ivr ports

2020-03-10 Thread Bill Talley
The platform capacity on UCCX is 400 ports using the largest OVA, no?   

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_12_5/design/guide/uccx_b_solution-design-guide-125/uccx_b_solution-design-guide-125_chapter_01001.html

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

> On Mar 10, 2020, at 6:52 PM, Myron Young  wrote:
> 
> 
> You should’ve gotten 600 IVR ports I believe but check with the licensing 
> team. May need to open a TAC case at this point for assistance 
> 
>>> On Mar 10, 2020, at 7:46 PM, Fares Alsaafani  wrote:
>>> 
>> 
>> Hello everyone,
>> I’m in the middle of Uccx deployment. We bought 300 premium agent licenses 
>> but only got 400 ivr port. Shouldn’t we get 600 ivr ports ?
>> -- 
>> Best Regards 
>> 
>> FARES ALSAAFANI
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] Jabber and locked screens - it works! but should it?

2020-01-23 Thread Bill Talley
Might be handy for calling emergency services or security, but other than that, 
it’s probably not something I would want the cleaning crew or mischievous 
co-workers being able to do. 

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

> On Jan 23, 2020, at 8:40 AM, Lelio Fulgenzi  wrote:
> 
> 
> OK. What do people think about Jabber working while a screen is locked? By 
> this, I mean that if your headset has a "pickup" button, you can answer the 
> call. Now, I know the first thing you will say is, "what's different from the 
> phone being picked up", and really, there isn't, but there is. I think people 
> will assume that if my computer is locked, it should not be allowed to work. 
> OR at a minimum, allow it to be a configurable option.
> 
> Then there's the possibility of using a handset that is Jabber compatible 
> which could be used to dial numbers while the computer is locked.
> 
> Thoughts?
> 
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Re: [cisco-voip] CCX Editor Step Properties

2019-09-24 Thread Bill Talley
Haha whew!  I was like, I know Tanner is a lot smarter than me (I’ve seen 
ContactView), but was confused about how I haven’t heard a solution to this 
overall problem. 

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

> On Sep 24, 2019, at 7:42 PM, Tanner Ezell  wrote:
> 
> I wasn't specifically responding to you Bill, just the thread in general :)
> 
> That being said, it won't be slow if there isn't much data to transfer. The 
> script editor is talking in java object terms, so small data can be quite 
> "big" and slow over the transport medium, it is also why DB resources and 
> such can take time on those steps. (also for the call sub flow, it tries to 
> grab all the scripts on the system, the more there are the slower it will be).
> 
> VPN only compounds the already inherently slow nature of the communication. 
> 
>> On Tue, Sep 24, 2019 at 5:33 PM Bill Talley  wrote:
>> I’m not disagreeing with you.  Im saying there are scenarios where it’s slow 
>> even if you’re on the local LAN.  You’re saying that’s NOT the case for you 
>> in any scenario?
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>> On Sep 24, 2019, at 7:11 PM, Tanner Ezell  wrote:
>>> 
>>> It will always be an issue. It uses RMI to pull data from the UCCX which is 
>>> especially slow over VPN. 
>>> 
>>>> On Tue, Sep 24, 2019 at 9:58 AM Bill Talley  wrote:
>>>> Good idea, but not always true.  I have several locations with scripts 
>>>> that contain REST and DB steps.  These take 7-10 second to open properties 
>>>> on any step regardless of whether I’m onsite or remote. 
>>>> 
>>>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>>>> Please excude my typtos.
>>>> 
>>>>> On Sep 24, 2019, at 11:53 AM, UC Penguin  wrote:
>>>>> 
>>>>> It’s much faster if the client is run at the same site as the uccx server 
>>>>> (RDP to a machine with the editor installed).
>>>>> 
>>>>> Not an option that’s possible for all, but there is a huge improvement. 
>>>>> Especially when using the debugger.
>>>>> 
>>>>> 
>>>>>> On Sep 24, 2019, at 10:52, Matthew Loraditch 
>>>>>>  wrote:
>>>>>> 
>>>>>> I’ve seen it editing scripts locally, in the repository, versions old 
>>>>>> and new.  
>>>>>> 
>>>>>>  
>>>>>> 
>>>>>>  
>>>>>> Matthew Loraditch​
>>>>>> Sr. Network Engineer
>>>>>> p: 443.541.1518
>>>>>> w: www.heliontechnologies.com |  e: 
>>>>>> mloradi...@heliontechnologies.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> From: cisco-voip  On Behalf Of 
>>>>>> Johnson, Tim
>>>>>> Sent: Tuesday, September 24, 2019 11:47 AM
>>>>>> To: Anthony Holloway 
>>>>>> Cc: cisco-voip@puck.nether.net
>>>>>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>>>>> 
>>>>>>  
>>>>>> 
>>>>>> Yeah, it’s still an issue in 12.0. Right now it’s opening properties for 
>>>>>> me in 1-2 seconds, other times it’s 10.
>>>>>> 
>>>>>>  
>>>>>> 
>>>>>> Would be nice if they did a full makeover of the tool, but I don’t 
>>>>>> expect it.
>>>>>> 
>>>>>>  
>>>>>> 
>>>>>> From: cisco-voip  On Behalf Of 
>>>>>> Anthony Holloway
>>>>>> Sent: Tuesday, September 24, 2019 11:11 AM
>>>>>> To: Matthew Loraditch 
>>>>>> Cc: cisco-voip@puck.nether.net
>>>>>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>>>>> 
>>>>>>  
>>>>>> 
>>>>>> I would like to know this too!  Seems to be different lengths of delay 
>>>>>> depending on a few factors, however, I have not nailed down what those 
>>>>>> factors are.  If it happened more often, I'd put some time into it, but 
>>>>>> since it's generally quick-ish, I ignore it.
>>>>>> 
>>>>>>  
>>>>&g

Re: [cisco-voip] CCX Editor Step Properties

2019-09-24 Thread Bill Talley
I’m not disagreeing with you.  Im saying there are scenarios where it’s slow 
even if you’re on the local LAN.  You’re saying that’s NOT the case for you in 
any scenario?

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

> On Sep 24, 2019, at 7:11 PM, Tanner Ezell  wrote:
> 
> It will always be an issue. It uses RMI to pull data from the UCCX which is 
> especially slow over VPN. 
> 
>> On Tue, Sep 24, 2019 at 9:58 AM Bill Talley  wrote:
>> Good idea, but not always true.  I have several locations with scripts that 
>> contain REST and DB steps.  These take 7-10 second to open properties on any 
>> step regardless of whether I’m onsite or remote. 
>> 
>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>> Please excude my typtos.
>> 
>>> On Sep 24, 2019, at 11:53 AM, UC Penguin  wrote:
>>> 
>>> It’s much faster if the client is run at the same site as the uccx server 
>>> (RDP to a machine with the editor installed).
>>> 
>>> Not an option that’s possible for all, but there is a huge improvement. 
>>> Especially when using the debugger.
>>> 
>>> 
>>>> On Sep 24, 2019, at 10:52, Matthew Loraditch 
>>>>  wrote:
>>>> 
>>>> I’ve seen it editing scripts locally, in the repository, versions old and 
>>>> new.  
>>>> 
>>>>  
>>>> 
>>>>
>>>> Matthew Loraditch​
>>>> Sr. Network Engineer
>>>> p: 443.541.1518
>>>> w: www.heliontechnologies.com   |  e: 
>>>> mloradi...@heliontechnologies.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> From: cisco-voip  On Behalf Of 
>>>> Johnson, Tim
>>>> Sent: Tuesday, September 24, 2019 11:47 AM
>>>> To: Anthony Holloway 
>>>> Cc: cisco-voip@puck.nether.net
>>>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>>> 
>>>>  
>>>> 
>>>> Yeah, it’s still an issue in 12.0. Right now it’s opening properties for 
>>>> me in 1-2 seconds, other times it’s 10.
>>>> 
>>>>  
>>>> 
>>>> Would be nice if they did a full makeover of the tool, but I don’t expect 
>>>> it.
>>>> 
>>>>  
>>>> 
>>>> From: cisco-voip  On Behalf Of Anthony 
>>>> Holloway
>>>> Sent: Tuesday, September 24, 2019 11:11 AM
>>>> To: Matthew Loraditch 
>>>> Cc: cisco-voip@puck.nether.net
>>>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>>> 
>>>>  
>>>> 
>>>> I would like to know this too!  Seems to be different lengths of delay 
>>>> depending on a few factors, however, I have not nailed down what those 
>>>> factors are.  If it happened more often, I'd put some time into it, but 
>>>> since it's generally quick-ish, I ignore it.
>>>> 
>>>>  
>>>> 
>>>> I have one system I access exclusively via AnyConnect, and it's running 
>>>> 11.6(2), and it's god awful slow.  Just clicking Add on the Set Enterprise 
>>>> Call Info step takes like 10-20 seconds for the dialog box to pop open.
>>>> 
>>>>  
>>>> 
>>>> On Tue, Sep 24, 2019 at 9:51 AM Matthew Loraditch 
>>>>  wrote:
>>>> 
>>>> Is there anyway on this earth to make this load faster???  Currently 
>>>> painfully going through a script and setting up new parameters and slowly 
>>>> dying of impatience as I click properties and wait a seeming eternity for 
>>>> the window to open.
>>>> 
>>>> This has been a pet peeve forever.
>>>> 
>>>>  
>>>> 
>>>> Matthew Loraditch​
>>>> 
>>>> Sr. Network Engineer
>>>> 
>>>> p: 443.541.1518
>>>> 
>>>> w: www.heliontechnologies.com
>>>> 
>>>>  |
>>>> 
>>>> e: mloradi...@heliontechnologies.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> ___
>>>> 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] CCX Editor Step Properties

2019-09-24 Thread Bill Talley
Good idea, but not always true.  I have several locations with scripts that 
contain REST and DB steps.  These take 7-10 second to open properties on any 
step regardless of whether I’m onsite or remote. 

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

> On Sep 24, 2019, at 11:53 AM, UC Penguin  wrote:
> 
> It’s much faster if the client is run at the same site as the uccx server 
> (RDP to a machine with the editor installed).
> 
> Not an option that’s possible for all, but there is a huge improvement. 
> Especially when using the debugger.
> 
> 
>> On Sep 24, 2019, at 10:52, Matthew Loraditch 
>>  wrote:
>> 
>> I’ve seen it editing scripts locally, in the repository, versions old and 
>> new.  
>>  
>>  
>> Matthew Loraditch​
>> Sr. Network Engineer
>> p: 443.541.1518
>> w: www.heliontechnologies.com |  e: 
>> mloradi...@heliontechnologies.com
>> 
>> 
>> 
>> 
>> From: cisco-voip  On Behalf Of Johnson, 
>> Tim
>> Sent: Tuesday, September 24, 2019 11:47 AM
>> To: Anthony Holloway 
>> Cc: cisco-voip@puck.nether.net
>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>  
>> Yeah, it’s still an issue in 12.0. Right now it’s opening properties for me 
>> in 1-2 seconds, other times it’s 10.
>>  
>> Would be nice if they did a full makeover of the tool, but I don’t expect it.
>>  
>> From: cisco-voip  On Behalf Of Anthony 
>> Holloway
>> Sent: Tuesday, September 24, 2019 11:11 AM
>> To: Matthew Loraditch 
>> Cc: cisco-voip@puck.nether.net
>> Subject: Re: [cisco-voip] CCX Editor Step Properties
>>  
>> I would like to know this too!  Seems to be different lengths of delay 
>> depending on a few factors, however, I have not nailed down what those 
>> factors are.  If it happened more often, I'd put some time into it, but 
>> since it's generally quick-ish, I ignore it.
>>  
>> I have one system I access exclusively via AnyConnect, and it's running 
>> 11.6(2), and it's god awful slow.  Just clicking Add on the Set Enterprise 
>> Call Info step takes like 10-20 seconds for the dialog box to pop open.
>>  
>> On Tue, Sep 24, 2019 at 9:51 AM Matthew Loraditch 
>>  wrote:
>> Is there anyway on this earth to make this load faster???  Currently 
>> painfully going through a script and setting up new parameters and slowly 
>> dying of impatience as I click properties and wait a seeming eternity for 
>> the window to open.
>> This has been a pet peeve forever.
>>  
>> Matthew Loraditch​
>> Sr. Network Engineer
>> p: 443.541.1518
>> w: www.heliontechnologies.com
>>  | 
>> e: mloradi...@heliontechnologies.com
>> 
>> 
>> 
>> 
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

2019-05-28 Thread Bill Talley
That’s the process I typically follow without issues.

Also, I can’t recall if this was posted here, but wanted to make sure you’ve 
seen the recent changes to resource requirements for IM   This may not apply 
to you if you have more than 5000 users.

IM VM resource requirements needs to be updated
CSCvk65006
Description
Symptom:
IM version 11.5.1.x or 12.0.1.x installed using one of the following 
configuration:
150 users (Full UC) 1vCPU 2 GB RAM
1,000 users (Full UC) 1vCPU 4 GB RAM
5,000 users (Full UC) 2vCPU 4 GB RAM

Customers with any of the above configuration might notice an increase use of 
CPU and Memory resources.

This can be fixed by manually increasing the resources according to the table 
below:
150 users (Full UC) 2vCPU 8 GB RAM
1,000 users (Full UC) 2vCPU 8 GB RAM
5,000 users (Full UC) 2vCPU 8 GB RAM

Conditions:
Performance Issues

Workaround:
Manually increase resources according to the table below:
150 users (Full UC) 2vCPU 8 GB RAM
1,000 users (Full UC) 2vCPU 8 GB RAM
5,000 users (Full UC) 2vCPU 8 GB RAM

If possible, it is recommended to have 4vCPU and 8 GB RAM as we are seeing more 
cases with high CPU due to resources related.



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

> On May 28, 2019, at 8:47 AM, Lelio Fulgenzi  wrote:
> 
> 
> I'm reading the upgrade guide, specifically, the time sequencing:
> 
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-guide-cucm-115_chapter_010010.html
> 
> and it mentions upgrading the IM publisher (to inactive partition) at the 
> same time as upgrading the subscribers. Then doing a switch version on the 
> IMP pub at the same time as the CUCM subs.
> 
> Anyone do this parallel type upgrade before?
> 
> Sure would save a lot of time.
> 
> 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]
> 
> 
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Re: [cisco-voip] looks like we're back to rebooting the cluster

2019-04-26 Thread Bill Talley
I knew I used it somewhere recently, the command is still available in CCX
10.6.

Try looking at 'file view install upgradeVMwareTools.log' to see if the
last vmtools upgrade/migration corresponds to your latest upgrade.  You'll
see the vmtools version there as well.

On Fri, Apr 26, 2019 at 3:02 PM Lelio Fulgenzi  wrote:

> [image: Image result for n elf]
>
>
>
> ---
>
> *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:* Bill Talley 
> *Sent:* Friday, April 26, 2019 4:01 PM
> *To:* Lelio Fulgenzi 
> *Cc:* Matthew Loraditch ; Brian Meade <
> bmead...@vt.edu>; cisco-voip@puck.nether.net; Pawlowski, Adam <
> aj...@buffalo.edu>
> *Subject:* Re: [cisco-voip] looks like we're back to rebooting the cluster
>
>
>
> Yeah, my bad... it’s an old command.
>
> Sent from an iOS device with very tiny touchscreen input keys.  Please
> excude my typtos.
>
>
> On Apr 26, 2019, at 2:58 PM, Lelio Fulgenzi  wrote:
>
>
>
> Is that a hidden command? ‘Cause this is all I get:
>
>
>
> admin:utils vmtools
>
>   utils vmtools refresh
>
>
>
> admin:utils vmtools
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> 
>
>
>
> *From:* Bill Talley 
> *Sent:* Friday, April 26, 2019 3:41 PM
> *To:* Lelio Fulgenzi 
> *Cc:* Matthew Loraditch ; Brian Meade <
> bmead...@vt.edu>; cisco-voip@puck.nether.net; Pawlowski, Adam <
> aj...@buffalo.edu>
> *Subject:* Re: [cisco-voip] looks like we're back to rebooting the cluster
>
>
>
> does ‘utils vmtools status’ tell you what you’re looking for (i.e. ‘out of
> date’ or current)? Or are you looking for more info than is generated by
> that command?
>
> Sent from an iOS device with very tiny touchscreen input keys.  Please
> excude my typtos.
>
>
> On Apr 26, 2019, at 2:32 PM, Lelio Fulgenzi  wrote:
>
>
>
> My colleague helped me confirm that the initramfs timestamp changes when
> VMtools is upgraded. I’m going to see if TAC can help by logging in as root
> and getting a timestamp on the file.
>
>
>
> We’ll still plan for a possible issue, but at least this will give us a
> sense of how possible the problem is.
>
>
>
> 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
>
>
>
> 
>
>
>
> *From:* cisco-voip  *On Behalf Of *Lelio
> Fulgenzi
> *Sent:* Friday, April 26, 2019 2:24 PM
> *To:* Matthew Loraditch ; Brian Meade <
> bmead...@vt.edu>
> *Cc:* cisco-voip@puck.nether.net; Pawlowski, Adam 
> *Subject:* Re: [cisco-voip] looks like we're back to rebooting the cluster
>
>
>
> Can I see when the last VM tools upgrade happened? We’re on an isolated VM
> infrastructure, but had partner work done and I wasn’t involved.
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> 
>
>
>
> *From:* Matthew Loraditch 
> *Sent:* Friday, April 26, 2019 2:14 PM
> *To:* Brian Meade 
> *Cc:* Lelio Fulgenzi ; Pawlowski, Adam <
> aj...@buffalo.edu>; cisco-voip@puck.nether.net
> *Subject:* RE: [cisco-voip] looks like we're back to rebooting the cluster
>
>
>
> Am I missing something? I mean the way I’m reading it plus the bugs is
> that if you are on that version and do a vmtools upgrade automatically it
> can happen?
>
>
>
> To me that means every system is vulnerable to the issue. But a second
> workaround is just

Re: [cisco-voip] looks like we're back to rebooting the cluster

2019-04-26 Thread Bill Talley
Yeah, my bad... it’s an old command.

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

> On Apr 26, 2019, at 2:58 PM, Lelio Fulgenzi  wrote:
> 
>  
> Is that a hidden command? ‘Cause this is all I get:
>  
> admin:utils vmtools
>   utils vmtools refresh
>  
> admin:utils vmtools
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Bill Talley  
> Sent: Friday, April 26, 2019 3:41 PM
> To: Lelio Fulgenzi 
> Cc: Matthew Loraditch ; Brian Meade 
> ; cisco-voip@puck.nether.net; Pawlowski, Adam 
> 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> does ‘utils vmtools status’ tell you what you’re looking for (i.e. ‘out of 
> date’ or current)? Or are you looking for more info than is generated by that 
> command?
> 
> Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
> my typtos.
> 
> On Apr 26, 2019, at 2:32 PM, Lelio Fulgenzi  wrote:
> 
>  
> My colleague helped me confirm that the initramfs timestamp changes when 
> VMtools is upgraded. I’m going to see if TAC can help by logging in as root 
> and getting a timestamp on the file.
>  
> We’ll still plan for a possible issue, but at least this will give us a sense 
> of how possible the problem is.
>  
> 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
>  
> 
>  
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, April 26, 2019 2:24 PM
> To: Matthew Loraditch ; Brian Meade 
> 
> Cc: cisco-voip@puck.nether.net; Pawlowski, Adam 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Can I see when the last VM tools upgrade happened? We’re on an isolated VM 
> infrastructure, but had partner work done and I wasn’t involved.
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Matthew Loraditch  
> Sent: Friday, April 26, 2019 2:14 PM
> To: Brian Meade 
> Cc: Lelio Fulgenzi ; Pawlowski, Adam ; 
> cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Am I missing something? I mean the way I’m reading it plus the bugs is that 
> if you are on that version and do a vmtools upgrade automatically it can 
> happen?
>  
> To me that means every system is vulnerable to the issue. But a second 
> workaround is just disable the auto vmtools upgrade option. As long as one 
> hasn’t happened you are fine.
>  
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  |
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> From: Brian Meade  
> Sent: Friday, April 26, 2019 2:04 PM
> To: Matthew Loraditch 
> Cc: Lelio Fulgenzi ; Pawlowski, Adam ; 
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Not every system on those affected versions is vulnerable.
>  
> On Fri, Apr 26, 2019 at 1:56 PM Matthew Loraditch 
>  wrote:
> What do you mean? It lists the affected releases and the ones with the fix?
>  
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  |
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, April 26, 2019 1:49 PM
> To: Pawlowski, Adam ; 'cisco-voip@puck.nether.net' 
> 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
>  
> It’s too bad they don’t have instructions on how to check if you are 
> vulnerable. Not that I could figure out anyways. At least there’s a work 
> around. Now to find that recovery disk image.
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal

Re: [cisco-voip] looks like we're back to rebooting the cluster

2019-04-26 Thread Bill Talley
does ‘utils vmtools status’ tell you what you’re looking for (i.e. ‘out of 
date’ or current)? Or are you looking for more info than is generated by that 
command?

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

> On Apr 26, 2019, at 2:32 PM, Lelio Fulgenzi  wrote:
> 
>  
> My colleague helped me confirm that the initramfs timestamp changes when 
> VMtools is upgraded. I’m going to see if TAC can help by logging in as root 
> and getting a timestamp on the file.
>  
> We’ll still plan for a possible issue, but at least this will give us a sense 
> of how possible the problem is.
>  
> 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
>  
> 
>  
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, April 26, 2019 2:24 PM
> To: Matthew Loraditch ; Brian Meade 
> 
> Cc: cisco-voip@puck.nether.net; Pawlowski, Adam 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Can I see when the last VM tools upgrade happened? We’re on an isolated VM 
> infrastructure, but had partner work done and I wasn’t involved.
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Matthew Loraditch  
> Sent: Friday, April 26, 2019 2:14 PM
> To: Brian Meade 
> Cc: Lelio Fulgenzi ; Pawlowski, Adam ; 
> cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Am I missing something? I mean the way I’m reading it plus the bugs is that 
> if you are on that version and do a vmtools upgrade automatically it can 
> happen?
>  
> To me that means every system is vulnerable to the issue. But a second 
> workaround is just disable the auto vmtools upgrade option. As long as one 
> hasn’t happened you are fine.
>  
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  |
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> From: Brian Meade  
> Sent: Friday, April 26, 2019 2:04 PM
> To: Matthew Loraditch 
> Cc: Lelio Fulgenzi ; Pawlowski, Adam ; 
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Not every system on those affected versions is vulnerable.
>  
> On Fri, Apr 26, 2019 at 1:56 PM Matthew Loraditch 
>  wrote:
> What do you mean? It lists the affected releases and the ones with the fix?
>  
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  |
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> From: cisco-voip  On Behalf Of Lelio 
> Fulgenzi
> Sent: Friday, April 26, 2019 1:49 PM
> To: Pawlowski, Adam ; 'cisco-voip@puck.nether.net' 
> 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
>  
> It’s too bad they don’t have instructions on how to check if you are 
> vulnerable. Not that I could figure out anyways. At least there’s a work 
> around. Now to find that recovery disk image.
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: cisco-voip  On Behalf Of Pawlowski, 
> Adam
> Sent: Friday, April 26, 2019 1:30 PM
> To: 'cisco-voip@puck.nether.net' 
> Subject: Re: [cisco-voip] looks like we're back to rebooting the cluster
>  
> Maybe you won’t be rebooting it if you don’t patch out for this one
>  
> https://www.cisco.com/c/en/us/support/docs/field-notices/703/fn70379.html
>  
>  
>  
> ___
> 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 Challenge - Reference the Running Script

2018-11-27 Thread Bill Talley
Maybe use a Get Trigger Info step to capture the application name, followed by 
Make REST Call to get the script name? 

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

> On Nov 27, 2018, at 2:29 PM, Anthony Holloway 
>  wrote:
> 
> I have worked out a method to retrieve the running script as a Script object, 
> but I'm not super happy with how I did it.  I.e., It's kind of ugly looking 
> and complex.
> 
> I'm wondering if anyone of you can retrieve a reference to the running 
> script, without hard coding the name in the script.  I.e., It should work on 
> any script without modification.
> 
> E.g., If the App which is executing is called Test and its script which is 
> executing is called test.aef, then your script should populate a Script 
> variable containing SCRIPT[test.aef].
> 
> I don't want to submit my version right away, just so I don't send you down 
> any one particular path, but I will share it once a few people have 
> responded, or if this gains zero traction after a bit.
> 
> Share your solutions!
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] connected party information while calling webex

2018-11-02 Thread Bill Talley
What is your cluster fully qualified domain name set to under Enterprise 
Parameters in CUCM?

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

> On Nov 2, 2018, at 2:15 PM, Lelio Fulgenzi  wrote:
> 
>  
> Yeah – that caught my eye as well while I was comparing. I think you are 
> referring to:
>  
> Calling and Connected Party Info Format
>  
> Options are:
>  
> Deliver DN only in connected party
> Deliver URI only in connected party, if available
> Deliver URI and DN in connected party, if available
>  
> Both are set to the last one, Deliver URI and DN in connected party, if 
> available.
>  
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: NateCCIE  
> Sent: Friday, November 2, 2018 2:52 PM
> To: Lelio Fulgenzi 
> Cc: Ryan Huff ; voyp list, cisco-voip 
> (cisco-voip@puck.nether.net) 
> Subject: Re: [cisco-voip] connected party information while calling webex
>  
> There is a setting on the sip trunks to deliver dn or uri or both. I bet 
> there is a difference between your two clusters on that setting. I may not 
> win the bet, but that is where I’d check first. 
> 
> Sent from my iPhone
> 
> On Nov 2, 2018, at 12:20 PM, Lelio Fulgenzi  wrote:
> 
>  
> Gotcha. All using FQDN as far as I can tell. Very odd. I don’t think I’ve 
> made changes to jabberconfig.xml either.
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Ryan Huff  
> Sent: Friday, November 2, 2018 2:18 PM
> To: Lelio Fulgenzi 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
> 
> Subject: Re: [cisco-voip] connected party information while calling webex
>  
> I meant UC, Siri assumed I meant UDS. How about the CM server references, all 
> FQDN as well?
> 
> Sent from my iPhone
> 
> On Nov 2, 2018, at 14:15, Lelio Fulgenzi  wrote:
> 
> What is this UDS thoust speak of?
>  
> But seriously, referenced where?
>  
> I checked my UC Services, all of those are FQDN. I checked my Service 
> Profile, and that just has a “Use UDS for Contact Resolution” option that 
> I’ve checked off in both systems.
>  
> What would UDS have to do with replacing @acme.webex.com with 
> @ though? Curious. I was leaning towards connected party 
> information setup.
>  
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Ryan Huff  
> Sent: Friday, November 2, 2018 1:58 PM
> To: Lelio Fulgenzi 
> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
> 
> Subject: Re: [cisco-voip] connected party information while calling webex
>  
> UDS services referenced by FQDN or IP?
> 
> Sent from my iPhone
> 
> On Nov 2, 2018, at 13:55, Lelio Fulgenzi  wrote:
> 
>  
> I’ve migrated my development cluster configuration to my production cluster 
> configuration and have compared them as best as possible, but I seem to be 
> missing connected party information when calling webex from Jabber.
>  
> On the development cluster, when I dial coy...@acme.webex.com from Jabber and 
> am connected, the connected party information at the top of Jabber remains 
> coy...@acme.webex.com, however, when I dial the same from production, it 
> changes to coyote@
>  
> I’ve reviewed the SIP trunk and dependencies as much as I could and they all 
> seem the same. I did some comparison of enterprise parameters and ccm service 
> parameters and they too look the same.
>  
> I can’t imagine anything expresway or within webex site config would cause 
> this.
>  
> Thoughts? Pointers?
>  
> ---
> 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
>  
> 
>  
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voipdata=02%7C01%7C%7C562fab46bf0d42e667d308d640ec54b0%7C84df9e7fe9f640afb435%7C1%7C0%7C636767781103601084sdata=YJE317AfF9SoJ6%2BkPwj2UzimeauGWR91KiD295rmcDg%3Dreserved=0
> 

Re: [cisco-voip] HELP

2018-11-01 Thread Bill Talley
Everybody needs somebody...


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

> On Nov 1, 2018, at 9:33 AM, Gary Parker  wrote:
> 
> I need somebody…
> 
> Gary
> 
>> On 1 Nov 2018, at 13:27, Fry, John  wrote:
>> 
>> help
>> 
>> 
>> 
>> State of Illinois - CONFIDENTIALITY NOTICE: The information contained in 
>> this communication is confidential, may be attorney-client privileged or 
>> attorney work product, may constitute inside information or internal 
>> deliberative staff communication, and is intended only for the use of the 
>> addressee. Unauthorized use, disclosure or copying of this communication or 
>> any part thereof is strictly prohibited and may be unlawful. If you have 
>> received this communication in error, please notify the sender immediately 
>> by return e-mail and destroy this communication and all copies thereof, 
>> including all attachments. Receipt by an unintended recipient does not waive 
>> attorney-client privilege, attorney work product privilege, or any other 
>> exemption from disclosure. 
>> 
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> ___
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Re: [cisco-voip] Unity Call Handler recording upload

2018-10-12 Thread Bill Talley
(replying all this time doh)

Client side, as in using the built-in afconvert command in Mac OSX?

If you have a MacBook, keep reading, otherwise this may not interest you.
In addition to creating temporary prompt files from the command line on
your Mac, you can also convert incompatible prompt files quickly and easily
from the command line with the following command.

afconvert -d ‘ulaw@8000’ -f ‘WAVE’ {source file name} {destination file
name}

For example, to convert HelpDesk.mp3 to the compatible uLaw format, the
following command would be executed from the directory containing
HelpDesk.mp3.
afconvert -d ‘ulaw@8000’ -f ‘WAVE’ HelpDesk.mp3 HelpDesk.wav

On Fri, Oct 12, 2018 at 8:54 AM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I agree.  How awesome would it be if it did the conversion purely client
> side?
>
> On Fri, Oct 12, 2018 at 8:51 AM Gary Parker 
> wrote:
>
>>
>>
>> > On 12 Oct 2018, at 14:43, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>> >
>> > In addition to Audacity, which I use myself, try this site out:
>> http://g711.org/
>>
>> Neat idea, and well implemented, but obviously please be wary of sending
>> your data to someone else’s site!
>>
>>
>> ---
>> /-Gary Parker--f--\
>> | Unified Communications Service Manager  |
>> n  Loughborough University, IT Services   |
>> | tel:+441509635635 sip:g...@lboro.ac.uk  o
>> |https://www.osx.ninja/pubkey.txt |
>> \r--d-/
>>
>> ___
> 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 Call Redirect and Called Address Reset

2018-09-24 Thread Bill Talley


Maybe I’m oversimplifying the work around (having not seen your environment), 
but could you not just set your redirect CSS to one that doesn’t have 
visibility of the 1XXX route pattern?

FWIW, I have agreed with your assessment 100%.  It makes no sense at all why 
CCM is performing DA/DM on dialed numbers while also ignoring routing info.  It 
would make a little more sense if the call was matching the route pattern due 
to use of the redirect css and routing the call out the route list/gateway, but 
with it ONLY modifying called party info, that’s crazy as hell. 

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

> On Sep 24, 2018, at 2:01 PM, Anthony Holloway 
>  wrote:
> 
> Damn! You played a Reverse card on me!  Well played.
> 
> Well, here's how it made me feel.  You were just testing the use of different 
> CSSs, which caused a difference in Digit Analysis.  You didn't provide any 
> logs to review, but I assume you have a different CSS on each of the: CTI 
> Port, VGW, and IP Phone; as most designs would call for.
> 
> I did some testing with the Redirect CSS setting on the CTI Ports as well, 
> and here are some findings.
> 
> I have three phone numbers in this test:
> +16515551901 is my IP Phone
> +16515551902 is my VM Pilot (Call Redirect Destination)
> +16515551903 is my Voice Mailbox (Call Redirect Reset Called Address to)
> I have two Partitions in this test:
> calling has my IP Phone and my VM Pilot
> port has my CTI Port
> 
> Note that I do not have a pattern even built to match the voice mailbox, so 
> it should go to CUC unmolested (which it does).
> I have two CSSs in this test:
> calling CSS which can only reach calling PT (on my calling device (my phone))
> port CSS which can only reach port PT (on my CTI Port)
> Example call with CTI Port Redirect CSS set to Calling Party (uses my phone 
> CSS to reach VM pilot and called address)
> 27900093.006 |11:10:33.852 |AppInfo  |Digit analysis: 
> match(pi="1",fqcn="+16515551901", cn="+16515551901", plv="5", pss="calling", 
> TodFilteredPss="calling", dd="+16515551903",dac="1")
> 27900093.007 |11:10:33.852 |AppInfo  |Digit analysis: analysis results  
> 
> 27900103.012 |11:10:33.853 |AppInfo  |Digit analysis: match(pi="1", 
> fqcn="+16515551901", cn="+16515551901",plv="5", pss="calling", 
> TodFilteredPss="calling", dd="+16515551902",dac="1")
> 27900103.013 |11:10:33.853 |AppInfo  |Digit analysis: analysis results
> 
> Example call with CTI Port Redirect CSS set to Redirect Party (uses CTI Port 
> CSS to try and reach, but fails, the VM pilot and called address)
> 28020535.006 |11:13:51.421 |AppInfo  |Digit analysis: 
> match(pi="1",fqcn="+16515551901", cn="+16515551901", plv="5", pss="port", 
> TodFilteredPss="port", dd="+16515551903",dac="1")
> 28020535.007 |11:13:51.421 |AppInfo  |Digit analysis: 
> potentialMatches=NoPotentialMatchesExist  
> 
> 28020548.006 |11:13:51.422 |AppInfo  |Digit analysis: 
> match(pi="1",fqcn="+16515551901", cn="+16515551901", plv="5", pss="port", 
> TodFilteredPss="port", dd="+16515551902",dac="1")
> 28020548.007 |11:13:51.422 |AppInfo  |Digit analysis: 
> potentialMatches=NoPotentialMatchesExist
> 
> Note that, I did not test the Redirect CSS setting of DN Calling Search 
> Space, because the documentation says it's deprecated.
> 
> Also note the order of DA: first its for the mailbox (Call Redirect Called 
> Address), then the DA for the VM pilot happens second.  
> 
> Also note that, I don't have to place an inbound call from outside, since the 
> calling CSS would just be used on my VGW, as opposed to my phone in these 
> tests.  It doesn't matter what the calling device is, just what its CSS is.
> 
> Further, the only time an OffNet or OnNet determination is needed, is when 
> forwarding on the CTI Route Point, I.e., Call Forward Busy Internal vs 
> External.  I'm not testing that, so, we can skip it for now.
> 
> Ok, so what did this simple test show me?  Well, it shows the actual impact 
> of changing the setting on the CTI Ports' Redirect CSS: Calling Party and 
> Redirect Party, and how that changes the Partition Search Space (PSS) used in 
> DA, which affects the Call Redirect Called Address, as that DA is performed 
> before the DA for the Call Redirect Destination.  I already know that it's 
> Partitions and Calling Search Spaces which dictate which and how patterns are 
> matched, that was never the issue.  The issue is with CUCM performing DA on, 
> *and* manipulating the Called Address, when I was trying to call the VM 
> pilot, not the Called Address.  And changing the CSS/PT is not an option for 
> me right now, since the pattern we're hitting for DA on the Called Address is 
> a pattern that I actually need UCCX to see.
> 
> From a CTI Manager perspective (different test call, so timestamps are 
> later), you can see how CTI manager receives the Call Redirect request from 
> UCCX
> 79624786.000 |13:30:03.172 |SdlSig   |CtiLineCallRedirectReq   

Re: [cisco-voip] UCCX Call Redirect and Called Address Reset

2018-09-21 Thread Bill Talley
“When RCSS is set to Redirecting Party, the call was routed normally with an 
untransformed called party.”

I dunno, how did that make you feel?

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

> On Sep 21, 2018, at 12:03 PM, Anthony Holloway 
>  wrote:
> 
> "... and was able to replicate your results."
> 
> And how did that make you feel?
> 
>> On Wed, Sep 19, 2018 at 5:21 PM Bill Talley  wrote:
>> Anthony,
>> 
>> So I just tested this scenario as you've laid out and was able to replicate 
>> your results.  That said, I encountered different results testing with 
>> internal calls and external inbound calls.
>> 
>> With internal calls like an internal user calling the help desk queue, the 
>> redirected call acted in the manner Anthony described when the Redirecting 
>> Calling Search Space on the CTI Ports was set to anything but Redirecting 
>> Party.  If the RCSS was set to Calling Party or DN Calling Search Space, I 
>> encountered the results you've described in which the called party is 
>> transformed via the transform masking on the route pattern, then delivered 
>> to CUC.  When RCSS is set to Redirecting Party, the call was routed normally 
>> with an untransformed called party.
>> 
>> With external inbound calls, the call was treated the way we would expect in 
>> that the called number is not transformed and was routed to CUC with the 
>> correct called party destination.
>> 
>> The results were the same regardless of whether the route pattern was 
>> visible by the CSS of the CTI port or not.
>> 
>> Hope this helps.
>> 
>>> On Tue, Sep 18, 2018 at 5:04 PM Bill Talley  wrote:
>>> Not yet, just finished a customer install early this morning.  Planning to 
>>> test it out tomorrow when I get back to my lab.
>>> 
>>> Sent from an iOS device with very tiny touchscreen input keys.  Please 
>>> excude my typtos.
>>> 
>>>> On Sep 18, 2018, at 4:52 PM, Anthony Holloway 
>>>>  wrote:
>>>> 
>>>> Update: I've been working a TAC case for a few days on this now, and I 
>>>> have not made any progress.  I also posted this scenario in the Advanced 
>>>> Dial Plan WebEx Teams space, but no replies so far.
>>>> 
>>>> Bill, have you tested this out?  Anyone else have some experience with 
>>>> this?
>>>> 
>>>>> On Tue, Sep 11, 2018 at 4:57 PM Anthony Holloway 
>>>>>  wrote:
>>>>> I didn't know this, and so I thought I'd share, but who knows, maybe it 
>>>>> was common knowledge.
>>>>> 
>>>>> If you use the Call Redirect step in UCCX to send a call directly to a 
>>>>> mailbox/call handler in CUC, and thus, your Destination is the VM Pilot, 
>>>>> while your target object in CUC is your Called Adddress, like so:
>>>>> 
>>>>> 
>>>>> 
>>>>> Then either one of two things will happen (only one of them I'm ok with):
>>>>> 
>>>>> 1) If there is a pattern in CUCM for which 1000 will match; say a Route 
>>>>> Pattern such as 1XXX which prefixes an 8 and route calls to a 3rd Party 
>>>>> PBX, then CUCM will use the Called Number Transformations on this Route 
>>>>> Pattern to prefix the 8 on your 1000, and then send the call to CUC with 
>>>>> 81000 as the Redir number, and you'll be all messed up.  Actually, you'll 
>>>>> just get the opening greeting, but still...g
>>>>> 
>>>>> 2) If there is no pattern in CUCM for which 1000 will match, then CUCM 
>>>>> sends the call to CUC, and the redir is 1000 and everything works fine.
>>>>> 
>>>>> I'll let you guess which one I'm ok with, and which one I'm not.
>>>>> 
>>>>> Why in the hell is CUCM performing number transformations on this call 
>>>>> flow like that?  It makes no sense.  What am I missing here?
>>>> ___
>>>> 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 Call Redirect and Called Address Reset

2018-09-18 Thread Bill Talley
Not yet, just finished a customer install early this morning.  Planning to test 
it out tomorrow when I get back to my lab.

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

> On Sep 18, 2018, at 4:52 PM, Anthony Holloway 
>  wrote:
> 
> Update: I've been working a TAC case for a few days on this now, and I have 
> not made any progress.  I also posted this scenario in the Advanced Dial Plan 
> WebEx Teams space, but no replies so far.
> 
> Bill, have you tested this out?  Anyone else have some experience with this?
> 
>> On Tue, Sep 11, 2018 at 4:57 PM Anthony Holloway 
>>  wrote:
>> I didn't know this, and so I thought I'd share, but who knows, maybe it was 
>> common knowledge.
>> 
>> If you use the Call Redirect step in UCCX to send a call directly to a 
>> mailbox/call handler in CUC, and thus, your Destination is the VM Pilot, 
>> while your target object in CUC is your Called Adddress, like so:
>> 
>> 
>> 
>> Then either one of two things will happen (only one of them I'm ok with):
>> 
>> 1) If there is a pattern in CUCM for which 1000 will match; say a Route 
>> Pattern such as 1XXX which prefixes an 8 and route calls to a 3rd Party PBX, 
>> then CUCM will use the Called Number Transformations on this Route Pattern 
>> to prefix the 8 on your 1000, and then send the call to CUC with 81000 as 
>> the Redir number, and you'll be all messed up.  Actually, you'll just get 
>> the opening greeting, but still...g
>> 
>> 2) If there is no pattern in CUCM for which 1000 will match, then CUCM sends 
>> the call to CUC, and the redir is 1000 and everything works fine.
>> 
>> I'll let you guess which one I'm ok with, and which one I'm not.
>> 
>> Why in the hell is CUCM performing number transformations on this call flow 
>> like that?  It makes no sense.  What am I missing here?
> ___
> 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 Call Redirect and Called Address Reset

2018-09-11 Thread Bill Talley
I’ve been doing this that way for years and haven’t come across this flaw.   Is 
it happening because it’s matching a “route pattern” or because it’s matching a 
called party transformation pattern?  I will have to test this out to see if 
I’ve just been lucky.  What version of CUCM are you running? 

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

> On Sep 11, 2018, at 4:57 PM, Anthony Holloway 
>  wrote:
> 
> I didn't know this, and so I thought I'd share, but who knows, maybe it was 
> common knowledge.
> 
> If you use the Call Redirect step in UCCX to send a call directly to a 
> mailbox/call handler in CUC, and thus, your Destination is the VM Pilot, 
> while your target object in CUC is your Called Adddress, like so:
> 
> 
> 
> Then either one of two things will happen (only one of them I'm ok with):
> 
> 1) If there is a pattern in CUCM for which 1000 will match; say a Route 
> Pattern such as 1XXX which prefixes an 8 and route calls to a 3rd Party PBX, 
> then CUCM will use the Called Number Transformations on this Route Pattern to 
> prefix the 8 on your 1000, and then send the call to CUC with 81000 as the 
> Redir number, and you'll be all messed up.  Actually, you'll just get the 
> opening greeting, but still...g
> 
> 2) If there is no pattern in CUCM for which 1000 will match, then CUCM sends 
> the call to CUC, and the redir is 1000 and everything works fine.
> 
> I'll let you guess which one I'm ok with, and which one I'm not.
> 
> Why in the hell is CUCM performing number transformations on this call flow 
> like that?  It makes no sense.  What am I missing here?
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] IM & Presence Upgrade image

2018-09-04 Thread Bill Talley
Sadly he’s not.  You can verify the name by mounting the ISO on your 
workstation and opening the isofilename file in the root of the disc image.
The ISO filename needs to match that or it won’t be recognized as a valid 
upgrade.

I had the same issue a couple of months ago.

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

> On Sep 4, 2018, at 8:46 AM, Dana Tong  wrote:
> 
> You are kidding me right. I downloaded this ISO from PUT.
>  
> My production ISO didn’t have the Bootable which explains a lot. It’s working.
>  
> Thank-you.
>  
> Cheers
> Dana
>  
>  
> From: Matthew Loraditch  
> Sent: Tuesday, 4 September 2018 11:42 PM
> To: Dana Tong ; cisco-voip voyp list 
> 
> Subject: RE: IM & Presence Upgrade image
>  
> Remove the “Bootable_” from the file name
>  
>  
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com
>  |
> e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> 
> From: cisco-voip  On Behalf Of Dana Tong
> Sent: Tuesday, September 4, 2018 9:41 AM
> To: cisco-voip voyp list 
> Subject: [cisco-voip] IM & Presence Upgrade image
>  
> Hi all,
>  
> I was trying to upgrade a couple of IM nodes but can’t recall what the 
> trick is.
>  
> I upgraded the CUCM and Unity Connection okay. Unity Connection requires a 
> cop file to recognise the upgrade image.
> I can’t find a cop file for the Presence server.
>  
> The OS Admin | Install reports “No Valid Upgrade options were found”.
>  
> Current version – 11.5.1.12900-25
> New file - Bootable_UCSInstall_CUP_12.0.1.1-12.sgn.iso
> New CUCM version - 12.0.1.21900-7
>  
> I have these versions running in my production system fine. I just can’t 
> recall what the problem is.
>  
> Cheers
> Dana
>  
>  
> ___
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

2018-08-11 Thread Bill Talley
Right, I may have misunderstood this was CUBE only. 

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

> On Aug 11, 2018, at 2:55 PM, NateCCIE  wrote:
> 
> NAT as in ip nat outside and ip nat inside, etc, would be used for other 
> traffic that is flowing through the router.  Address hiding for a SIP 
> connection is different.
>  
> From: Bill Talley  
> Sent: Saturday, August 11, 2018 1:08 PM
> To: NateCCIE 
> Cc: Benjamin Turner ; cisco-voip voyp list 
> 
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> Is that not essentially what address hiding and binding the telco facing 
> dial-peer to the outside interface accomplishes?
> 
> Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
> my typtos.
> 
> On Aug 11, 2018, at 1:24 PM, NateCCIE  wrote:
> 
> No, if you want to do NAT, it has to be on a different interface than what 
> CUBE is using for SIP.
>  
> From: Benjamin Turner  
> Sent: Saturday, August 11, 2018 12:03 PM
> To: NateCCIE 
> Cc: Ryan Huff ; cisco-voip voyp list 
> 
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> So should I implement a voice class sip profile to modify the sip headers?
> 
> 
> 
> 
> Get Outlook for Android
>  
> From: NateCCIE 
> Sent: Saturday, August 11, 2018 1:46:31 PM
> To: Benjamin Turner
> Cc: Ryan Huff; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> You can’t have ip NAT outside on a CUBE. It used to work before the 4K but it 
> was never supported according to TAC and the BU.
>  
> Inbound traffic goes to the NAT process and the SIP stack never sees it
> 
> Sent from my iPhone
> 
> On Aug 10, 2018, at 9:38 PM, Benjamin Turner  wrote:
> 
> My brain is FRIED
> 
> Get Outlook for Android
>  
> From: Benjamin Turner 
> Sent: Friday, August 10, 2018 11:35:15 PM
> To: Ryan Huff
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> I did. They see my outside IP (multiple) invites and they are sending 1xx 
> responses. And, I see them too on my cube with a monitor capture. I just do 
> not see them with a debug ccsip messages
> 
> Get Outlook for Android
>  
> From: Ryan Huff 
> Sent: Friday, August 10, 2018 11:32:08 PM
> To: Benjamin Turner
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> and you’re sure you’re signaling the correct peer address for signaling that 
> the carrier gave you when they provisioned the trunk?
> 
> You may need to do a live TSHOOT with the carrier so they can tell you if 
> they are (and what) they see in their SBC.
> 
> On Aug 10, 2018, at 23:29, Benjamin Turner  wrote:
> 
> Yeah tried both ways. Inside and outside IP on the cucm trunk config
> 
> Get Outlook for Android
>  
> From: Bill Talley 
> Sent: Friday, August 10, 2018 11:28:02 PM
> To: Benjamin Turner
> Cc: Ryan Huff; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> Just confirming when you changed the bindings in IOS, did you change the SIP 
> trunk destination IP address in CUCM to point to the inside address of CUBE 
> and reset the trunk?
> 
> Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
> my typtos.
> 
> On Aug 10, 2018, at 9:25 PM, Benjamin Turner  wrote:
> 
> 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-interfac

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

2018-08-11 Thread Bill Talley
Is that not essentially what address hiding and binding the telco facing 
dial-peer to the outside interface accomplishes?

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

> On Aug 11, 2018, at 1:24 PM, NateCCIE  wrote:
> 
> No, if you want to do NAT, it has to be on a different interface than what 
> CUBE is using for SIP.
>  
> From: Benjamin Turner  
> Sent: Saturday, August 11, 2018 12:03 PM
> To: NateCCIE 
> Cc: Ryan Huff ; cisco-voip voyp list 
> 
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> So should I implement a voice class sip profile to modify the sip headers?
> 
> 
> 
> Get Outlook for Android
>  
> From: NateCCIE 
> Sent: Saturday, August 11, 2018 1:46:31 PM
> To: Benjamin Turner
> Cc: Ryan Huff; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> You can’t have ip NAT outside on a CUBE. It used to work before the 4K but it 
> was never supported according to TAC and the BU.
>  
> Inbound traffic goes to the NAT process and the SIP stack never sees it
> 
> Sent from my iPhone
> 
> On Aug 10, 2018, at 9:38 PM, Benjamin Turner  wrote:
> 
> My brain is FRIED
> 
> Get Outlook for Android
>  
> From: Benjamin Turner 
> Sent: Friday, August 10, 2018 11:35:15 PM
> To: Ryan Huff
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> I did. They see my outside IP (multiple) invites and they are sending 1xx 
> responses. And, I see them too on my cube with a monitor capture. I just do 
> not see them with a debug ccsip messages
> 
> Get Outlook for Android
>  
> From: Ryan Huff 
> Sent: Friday, August 10, 2018 11:32:08 PM
> To: Benjamin Turner
> Cc: Bill Talley; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> and you’re sure you’re signaling the correct peer address for signaling that 
> the carrier gave you when they provisioned the trunk?
> 
> You may need to do a live TSHOOT with the carrier so they can tell you if 
> they are (and what) they see in their SBC.
> 
> On Aug 10, 2018, at 23:29, Benjamin Turner  wrote:
> 
> Yeah tried both ways. Inside and outside IP on the cucm trunk config
> 
> Get Outlook for Android
>  
> From: Bill Talley 
> Sent: Friday, August 10, 2018 11:28:02 PM
> To: Benjamin Turner
> Cc: Ryan Huff; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> Just confirming when you changed the bindings in IOS, did you change the SIP 
> trunk destination IP address in CUCM to point to the inside address of CUBE 
> and reset the trunk?
> 
> Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
> my typtos.
> 
> On Aug 10, 2018, at 9:25 PM, Benjamin Turner  wrote:
> 
> 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\

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

2018-08-10 Thread Bill Talley
Just confirming when you changed the bindings in IOS, did you change the SIP 
trunk destination IP address in CUCM to point to the inside address of CUBE and 
reset the trunk?

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

> On Aug 10, 2018, at 9:25 PM, Benjamin Turner  wrote:
> 
> 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 
>>  dtmf-relay rtp-nte
>> no vad
>> !
>> dial-peer voice 4001 voip
>> description LD Dialing (11-Digit) to Bandwidth SIP Trunk
>> translation-profile outgoing OutgoingToBandwidthSIP
>> destination-pattern 1[2-9]..[2-9]..$
>> session protocol sipv2
>> session target ipv4:67.231.8.75
>> voice-class codec 1 
>>  dtmf-relay rtp-nte
>> no vad
>> !
>> dial-peer voice 2001 voip
>> description Incoming SIP Calls to CUCM Sub
>> translation-profile incoming LOCALIZE
>> destination-pattern [2-9]..[2-9]..$
>> session protocol sipv2
>> session target ipv4:192.168.1.20
>> voice-class codec 1 
>>  no voice-class sip outbound-proxy  
>>  dtmf-relay rtp-nte sip-notify
>> no vad
>> !
>> !
>> sip-ua
>> !
>>  
>>  
>> Sent from Mail for Windows 10
>>  
>> From: Ryan Huff 
>> Sent: Friday, August 10, 2018 10:04:54 PM
>> To: Benjamin Turner
>> Cc: Loren Hillukka; cisco-voip voyp list
>> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>>  
>> Can I get a show tech or at least a show run?
>> 
>> Sent from my iPhone
>> 
>> On Aug 10, 2018, at 22:03, Benjamin Turner  wrote:
>> 
>>> Its an edge device no fw in place between cube and provider 
>>> 
>>> Get Outlook for Android
>>> 
>>> From: Ryan 

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

2018-08-10 Thread Bill Talley
In other words, your signaling is bound to the public address on CUBE.  Unless 
the inside interface on CUBE is the default gateway for call manager, or Call 
Managers default gateway has a host route pointing that traffic to the inside 
address of CUBE, the traffic may be going out the firewall and hitting the 
outside address of CUBE?  Again, my apologies if this is outlined in one of the 
other emails.

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

> On Aug 10, 2018, at 10:01 PM, Bill Talley  wrote:
> 
> This may have already been asked, sorry.  Since everything is bound to the 
> outside address on CUBE, is call manager sending traffic back to your 
> firewall instead of CUBE and getting dropped by the firewall or redirect by 
> the firewall to CUBE?  
> 
> Sent from an iOS device with very tiny touchscreen input keys.  Please excude 
> my typtos.
> 
>> On Aug 10, 2018, at 9:15 PM, 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 
>>  dtmf-relay rtp-nte
>> no vad
>> !
>> dial-peer voice 4001 voip
>> description LD Dialing (11-Digit) to Bandwidth SIP Trunk
>> translation-profile outgoing OutgoingToBandwidthSIP
>> destination-pattern 1[2-9]..[2-9]..$
>> session protocol sipv2
>> session target ipv4:67.231.8.75
>> voice-class codec 1 
>>  dtmf-relay rtp-nte
>> no vad
>> !
>> dial-peer voice 2001 voip
>> description Incoming SIP Calls to CUCM Sub
&g

Re: [cisco-voip] CUBE ignoring SDP responses from ITSP

2018-08-10 Thread Bill Talley
This may have already been asked, sorry.  Since everything is bound to the 
outside address on CUBE, is call manager sending traffic back to your firewall 
instead of CUBE and getting dropped by the firewall or redirect by the firewall 
to CUBE?  

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

> On Aug 10, 2018, at 9:15 PM, 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 
>  dtmf-relay rtp-nte
> no vad
> !
> dial-peer voice 4001 voip
> description LD Dialing (11-Digit) to Bandwidth SIP Trunk
> translation-profile outgoing OutgoingToBandwidthSIP
> destination-pattern 1[2-9]..[2-9]..$
> session protocol sipv2
> session target ipv4:67.231.8.75
> voice-class codec 1 
>  dtmf-relay rtp-nte
> no vad
> !
> dial-peer voice 2001 voip
> description Incoming SIP Calls to CUCM Sub
> translation-profile incoming LOCALIZE
> destination-pattern [2-9]..[2-9]..$
> session protocol sipv2
> session target ipv4:192.168.1.20
> voice-class codec 1 
>  no voice-class sip outbound-proxy  
>  dtmf-relay rtp-nte sip-notify
> no vad
> !
> !
> sip-ua
> !
>  
>  
> Sent from Mail for Windows 10
>  
> From: Ryan Huff 
> Sent: Friday, August 10, 2018 10:04:54 PM
> To: Benjamin Turner
> Cc: Loren Hillukka; cisco-voip voyp list
> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>  
> Can I get a show tech or at least a show run?
> 
> Sent from my iPhone
> 
> On Aug 10, 2018, at 22:03, Benjamin Turner  wrote:
> 
>> Its an edge device no fw in place between cube and provider 
>> 
>> Get Outlook for Android
>> 
>> From: Ryan Huff 
>> Sent: Friday, August 10, 2018 10:02:42 PM
>> To: Benjamin Turner
>> Cc: Loren Hillukka; cisco-voip voyp list
>> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>>  
>> Is it a PA, ASA? Do you have inspection and ALG disabled on the firewall?
>> 
>> Sent from my iPhone
>> 
>> On Aug 10, 2018, at 22:00, Benjamin Turner  wrote:
>> 
>>> Sit in front of the fw
>>> 
>>> Get Outlook for Android
>>> 
>>> From: Loren Hillukka 
>>> Sent: Friday, August 10, 2018 9:58:49 PM
>>> To: Benjamin Turner
>>> Cc: Ryan Huff; cisco-voip voyp list
>>> Subject: Re: [cisco-voip] CUBE ignoring SDP responses from ITSP
>>>  
>>> Any firewall in place there? I have seen firewalls modify/drop just enough 
>>> to make the gateway not recognize a response. 
>>> 
>>> Sent from my iPhone
>>> 
>>> On Aug 10, 2018, at 

Re: [cisco-voip] CM 11.5 hunt group longest idle

2018-08-06 Thread Bill Talley
Try this from the CUCM CLI:

run sql select lg.name as LineGroup,n.dnorpattern,dhd.hlog from linegroup
as lg inner join linegroupnumplanmap as lgmap on lgmap.fklinegroup=lg.pkid
inner join numplan as n on lgmap.fknumplan = n.pkid inner join
devicenumplanmap as dmap on dmap.fknumplan = n.pkid inner join device as d
on dmap.fkdevice=d.pkid inner join devicehlogdynamic as dhd on
dhd.fkdevice=d.pkid order by lg.name

That should show you the current login status of each phone to at least
tell you which are logged in to the line group.

On Mon, Aug 6, 2018 at 10:55 AM, Scott Voll  wrote:

> Anyone getting reports in CM 11.5 of Hunt groups not working correctly?
> We have the hunt group set to longest idle but one person seems to be
> getting all the calls.
>
> How do you trouble shoot this?  it's not like UCCx reports.
>
> is there a way to see who is logged into the hunt group?  and How long?
>
> TIA
>
> Scott
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] multi-SAN / server certificates vs individual certs (CUCM/IMP)

2018-06-28 Thread Bill Talley
Scrolling through my phone and inadvertently replied to Charles email when it 
popped up instead of Lelio’s.  Sorry for duplicating what Charles said 浪

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

> On Jun 28, 2018, at 10:24 AM, Charles Goldsmith  wrote:
> 
> Generate a CSR from each server type (CUCM, CUC, UCCX, and each expressway) 
> and load all hostnames into each server, including your cluster name of the 
> expressway and the domain name.  At Digicert, load your csr, make sure the 
> Common name matches the CSR that the server came from.  Once you have one 
> cluster done, go back into the order and request duplicate, load your 2nd 
> csr, check the common name and issue the duplicate.  Rinse and repeat for all 
> systems.
> 
> Expressway clusters do not support multi-san, so just duplicate for each node.
> 
>> On Thu, Jun 28, 2018 at 10:17 AM Lelio Fulgenzi  wrote:
>> Wait. What? I understand how the internals of CUCM and IMP can distribute 
>> one multi-san cert (built on the publisher’s CSR) to each CUCM and IMP node 
>> and uses private keys to ensure they load, but….
>> 
>>  
>> 
>> How the heck do you install a cert that was built on the pub’s CSR into CUC 
>> and UCCx? Or Expressway for that matter?
>> 
>>  
>> 
>> We are a digicert client, so if you have specific breadcrumbs / drop down 
>> options, feel free to share.
>> 
>>  
>> 
>> 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
>> 
>>  
>> 
>> 
>> 
>>  
>> 
>> From: Charles Goldsmith  
>> Sent: Thursday, June 28, 2018 10:40 AM
>> To: Lelio Fulgenzi 
>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>> 
>> Subject: Re: [cisco-voip] multi-SAN / server certificates vs individual 
>> certs (CUCM/IMP)
>> 
>>  
>> 
>> I've used multi-san certs on at least a dozen installs and have had no 
>> issues at all.  In fact, with a good SSL provider, you can use the same 
>> Multi-SAN on CUCM, CUC, UCCX, Expressways.  I like how Digicert does it, 
>> just duplicate the cert  and make sure all of the hostnames are listed in 
>> the SAN.
>> 
>>  
>> 
>>  
>> 
>> On Thu, Jun 28, 2018 at 9:37 AM Lelio Fulgenzi  wrote:
>> 
>> 
>> We're in the process of installing signed certs and we have the choice 
>> between multi-SAN cert with the publisher CSR and rely on the internals to 
>> have that cert distributed to the subs and the imp nodes -OR- go with 
>> individual certs.
>> 
>> It's a last minute thing, so I still need to do some research, but I'm 
>> wondering what people have been doing out there. We're less concerned with 
>> cost than we are future stability. I know that this multi-san support is 
>> recent with v10.x - have they ironed out the bugs? We're going with 11.5.
>> 
>> Thoughts?
>> 
>> 
>> ---
>> 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
>> 
> ___
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Re: [cisco-voip] multi-SAN / server certificates vs individual certs (CUCM/IMP)

2018-06-28 Thread Bill Talley
I second the DigiCert recommendation.  They are unique in that they let you 
generate multiple server certificates (using multiple private keys) under the 
same multi-SAN cert order.  

You just add all of the fqdns to the same certificate order (via the CSR) and 
when you generate the multiSAN CSR from each server (or cluster) you will add 
the fqdn of the other servers as alternative names.   So when you generate a 
Unity Connection CSR, you will add the cucm nodes, CCX nodes, etc as 
alternative names.   If you don’t do that, digicert will invalidate any 
previous signed certs under that order so make sure you include the same 
alternative names in every CSR.

Every other provider we’ve reviewed requires you to either share the private 
keys (which Cisco UC servers don’t allow) or they make you order separate 
multi-SAN certs per cluster.

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

> On Jun 28, 2018, at 10:24 AM, Charles Goldsmith  wrote:
> 
> Generate a CSR from each server type (CUCM, CUC, UCCX, and each expressway) 
> and load all hostnames into each server, including your cluster name of the 
> expressway and the domain name.  At Digicert, load your csr, make sure the 
> Common name matches the CSR that the server came from.  Once you have one 
> cluster done, go back into the order and request duplicate, load your 2nd 
> csr, check the common name and issue the duplicate.  Rinse and repeat for all 
> systems.
> 
> Expressway clusters do not support multi-san, so just duplicate for each node.
> 
>> On Thu, Jun 28, 2018 at 10:17 AM Lelio Fulgenzi  wrote:
>> Wait. What? I understand how the internals of CUCM and IMP can distribute 
>> one multi-san cert (built on the publisher’s CSR) to each CUCM and IMP node 
>> and uses private keys to ensure they load, but….
>> 
>>  
>> 
>> How the heck do you install a cert that was built on the pub’s CSR into CUC 
>> and UCCx? Or Expressway for that matter?
>> 
>>  
>> 
>> We are a digicert client, so if you have specific breadcrumbs / drop down 
>> options, feel free to share.
>> 
>>  
>> 
>> 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
>> 
>>  
>> 
>> 
>> 
>>  
>> 
>> From: Charles Goldsmith  
>> Sent: Thursday, June 28, 2018 10:40 AM
>> To: Lelio Fulgenzi 
>> Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 
>> 
>> Subject: Re: [cisco-voip] multi-SAN / server certificates vs individual 
>> certs (CUCM/IMP)
>> 
>>  
>> 
>> I've used multi-san certs on at least a dozen installs and have had no 
>> issues at all.  In fact, with a good SSL provider, you can use the same 
>> Multi-SAN on CUCM, CUC, UCCX, Expressways.  I like how Digicert does it, 
>> just duplicate the cert  and make sure all of the hostnames are listed in 
>> the SAN.
>> 
>>  
>> 
>>  
>> 
>> On Thu, Jun 28, 2018 at 9:37 AM Lelio Fulgenzi  wrote:
>> 
>> 
>> We're in the process of installing signed certs and we have the choice 
>> between multi-SAN cert with the publisher CSR and rely on the internals to 
>> have that cert distributed to the subs and the imp nodes -OR- go with 
>> individual certs.
>> 
>> It's a last minute thing, so I still need to do some research, but I'm 
>> wondering what people have been doing out there. We're less concerned with 
>> cost than we are future stability. I know that this multi-san support is 
>> recent with v10.x - have they ironed out the bugs? We're going with 11.5.
>> 
>> Thoughts?
>> 
>> 
>> ---
>> 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
>> 
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Re: [cisco-voip] Bye Bye Cisco Spark, Hello Webex Teams

2018-04-19 Thread Bill Talley
If the words “team” or “teams” were protectable by trademark, I’m sure the MLB 
would have a lot to say about Microsoft and everyone else using the term 
“without the express written consent of Major League Baseball”.  

Yes, I know, different fields, pardon the pun.

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

> On Apr 19, 2018, at 9:31 AM, Ryan Huff  wrote:
> 
> Common words and phrases can be trademarked if the person or company seeking 
> the trademark candemonstrate that the phrase has acquired a distinctive 
> secondary meaning apart from its original meaning. That secondary meaning 
> must be one that identifies the phrase with a particular good or service.
> 
> https://www.uspto.gov/sites/default/files/documents/BasicFacts.pdf
> 
> I don’t think “teams” would qualify; I couldn’t imagine saying “teams” to 
> anyone and them automatically thinking, “Oh, that Microsoft thing”.
> 
> If the users won’t come to Spark, I guess you bring Spark to the users and 
> put a little dressing on it (to distance the platform from those early on 
> scale based outages ;).
> 
> It’s a pretty creative way get valuation for the product if you think about. 
> Tap into a well known and loved branding message with a massive pre existing 
> user base with a high adoption to churn ratio. It’s all about fiduciary 
> responsibility ;).
> 
> - Ryan -
> 
> On Apr 19, 2018, at 10:19, Matthew Loraditch 
>  wrote:
> 
>> I can’t imagine they didn’t thoroughly vet it before doing this. Also teams 
>> has a very generic meaning. It’s Like saying Kleenex Tissues or Puffs 
>> Tissues. Both providing the same service. That’s my mostly uninformed two 
>> cents.
>>  
>>  
>> Matthew Loraditch​
>> Sr. Network Engineer
>> p: 443.541.1518
>> w: www.heliontechnologies.com |  e: 
>> mloradi...@heliontechnologies.com
>> 
>> 
>> 
>> 
>> 
>> From: cisco-voip  On Behalf Of Ben Amick
>> Sent: Thursday, April 19, 2018 10:09 AM
>> To: Lamont, Joshua ; cisco-voip voyp list 
>> 
>> Subject: Re: [cisco-voip] Bye Bye Cisco Spark, Hello Webex Teams
>>  
>> Smells like a lawsuit from Microsoft waiting to happen.
>> From: cisco-voip  on behalf of Lamont, 
>> Joshua 
>> Sent: Wednesday, April 18, 2018 9:54:43 AM
>> To: cisco-voip voyp list
>> Subject: [cisco-voip] Bye Bye Cisco Spark, Hello Webex Teams
>>  
>> Security Notice: This is an EXTERNAL email. Please exercise caution and DO 
>> NOT open attachments or click links from unknown or unexpected senders. 
>> Announcement today at noon to be livestreamed on Cisco.com 
>>  
>> Some pre announcement details here: 
>> https://www.nojitter.com/post/240173438/bye-bye-cisco-spark-hello-webex-teams
>> 
>> Joshua Lamont
>> Senior Telecommunications Engineer
>> Brown University
>> office (401) 863-1003
>> cell(401) 749-6913
>> Disclaimer: The information contained in this transmission and any 
>> attachments may contain privileged and confidential information including 
>> patient information protected by federal and state privacy laws. This 
>> transmission is sent for the sole use of the intended recipient. If you are 
>> not the intended recipient, please immediately (1) notify the sender via 
>> reply email; (2) do not open or read the message or any attachments; and (3) 
>> delete this message and any attachments. The review, dissemination, 
>> distribution, or duplication of this transmission by anyone other than the 
>> intended recipient is strictly prohibited. 
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Re: [cisco-voip] Robo Call DoS

2018-04-16 Thread Bill Talley
I’m sure you’ve already considered this, but I wonder, with everything being 
back-hauled over IP these days, if their telco would be able to identify the 
IXC who is handing off the call to them, based on the logs for the original 
calling party number they blocked, and blacklist that address.  Sure that 
sounds extreme (voice is more critical them SMTP), but you’re also talking 
about criminal activity.

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

> On Apr 16, 2018, at 10:23 AM, Anthony Holloway 
>  wrote:
> 
> Technically or legally?
> 
> How does one stop a DoS attack on a network?  Or on anything for that matter? 
>  Say you were attending a protest, and someone is blowing an air horn in your 
> ear?  What can you do?
> 
> Technically, you could front end the whole thing with a captcha style gate, 
> so you could ask to push a single button, button combination, or solve a 
> simple addition problem resulting in two digits.  granted, just like on the 
> web, a captcha is burdensome to the user, but generally, it's preferable over 
> the site being down, or disrupted.
> 
> CUC and UCCX both could handle this task, though it would be easier in UCCX.
> 
>> On Mon, Apr 16, 2018 at 9:49 AM Matthew Loraditch 
>>  wrote:
>> So this is a curiosity question, we had a prospective client call us who is 
>> essentially getting robocalled to oblivion. Some scammer has robo dialers 
>> setup and is flooding all of their trunks. He got a ransom, stopped and then 
>> started again. He was originally using one number and then when the telco 
>> blocked that switching to random sources.
>> 
>> Are there are any legitimate defenses to this sort of thing?
>> 
>>  
>> Matthew Loraditch​
>> Sr. Network Engineer
>> p: 443.541.1518
>> w: www.heliontechnologies.com |  e: 
>> mloradi...@heliontechnologies.com
>> 
>> 
>> 
>> 
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Re: [cisco-voip] CUCM Pre-Upgrade Checklist

2018-03-31 Thread Bill Talley
I've seen it and had the same reaction you're having.  I *believe*, but
could be wrong, those disclaimers are added in relation to Prime
Collaboration Deployment (PCD) upgrades.  I've performed several PCD
upgrades and experienced the issues described in your email (and many
others) when using PCD and have stopped using PCD for those reason.  Doing
direct upgrades, I've taken the precautions outlined in the docs you
referenced, but have not encountered those issues during direct upgrades.

On Sat, Mar 31, 2018 at 10:38 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Has anyone seen this lately?  This thing is nuts.  I've been doing
> upgrades for like 10 years now, and this seems a little over the top.
>
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/
> upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-
> guide-cucm-115_chapter_010001.html
>
> Some Highlights:
>
> "If you have custom ringtones or background images in the TFTP directory,
> you need to create a separate backup for these files. They are not included
> in the Disaster Recovery System (DRS) backup file."
>
> Good to know.  I must admit that I didn't know that.
>
> "Record the following login and password information: all application
> users credentials, such as DRS, AXL, and accounts for other third-party
> integration"
>
> What?  Why?  What do you plan on doing with my configuration?
>
> "Record the settings for Enterprise Parameters on both [CM] nodes and
> [IM] nodes. [T]he settings that are configured on Unified Communications
> Manager nodes overwrite the settings configured on IM and Presence Service
> nodes during the upgrade process."
>
> Well that's some lazy software engineering right there folks.
>
> "Export user records using the Bulk Administration Tool (BAT)."
>
> That's a nice list of users you got there.  It would be a shame if this
> upgrade deleted all of them.
>
> And the list just goes on, and on, and on.  The pre-upgrade is as long as
> the upgrade.  Who legitimately is already doing 100% of these things?
>
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Re: [cisco-voip] Windows AXL Query Tool

2018-03-28 Thread Bill Talley
True but importing the WSDL file or files (I was referring to that as a
component) from the toolkit ensures you have the current schema, which is
important, especially if you're working with multiple CUCM versions.

On Wed, Mar 28, 2018 at 2:38 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I don't think this is splitting hairs, but the toolkit is just code Cisco
> is handing out that makes working with their AXL API easier.
>
> At the most basic level I'd care to explore, you could simply hand craft
> your SOAP payload, and send the HTTPS request to CUCM and never once use
> the toolkit nor WSDL.
>
> If all you're doing is using the WSDL from the toolkit, then you're
> technically not using the toolkit, rather you're using resources (the WSDL)
> that the toolkit also happens to be using.
>
> It would be great if the WSDL was available right from CUCM, so you didn't
> have to download it.
>
>
> On Wed, Mar 28, 2018 at 11:20 AM Bill Talley <btal...@gmail.com> wrote:
>
>> Lelio,
>>
>> The AXL toolkit is what the python CLI scripts I posted earlier are based
>> on.  The script imports the wsdl file from the AXL toolkit and uses it to
>> query CUCM.  The links I posted explain the setup required.
>>
>> As far as I know, all AXL/SOAP scripts or webpages regardless of
>> programming language will use the same AXL toolkit or components of it to
>> access CUCM.  Someone correct me if I’m wrong please.
>>
>> I’m cleaning up the full python script as well as working on a GUI and
>> distributable package for the script, but hopefully they will give you an
>> idea of how to get started with Python.  It can seem overwhelming at first,
>> but it’s not that bad.  As many people can probably tell from the scripts,
>> I’m not a programmer but once you pick up the basics, it’s can be quit
>> powerful even with limited experience.
>>
>> Bill
>>
>> Sent from a mobile device with very tiny touchscreen input keys. Please
>> excude my typtos.
>>
>> On Mar 28, 2018, at 10:52 AM, Lelio Fulgenzi <le...@uoguelph.ca> wrote:
>>
>> Ok. This is a great help. I’ll have to see which is better for me (or
>> which is easier to use) – this toolkit or CLI.
>>
>>
>>
>>
>>
>>
>>
>> ---
>>
>> *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
>> <https://maps.google.com/?q=50+Stone+Rd+E+%7C+Guelph,+ON+%7C+N1G+2W1=gmail=g>
>>
>> 519-824-4120 Ext. 56354 <(519)%20824-4120> | le...@uoguelph.ca
>>
>>
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>
>>
>>
>> 
>>
>>
>>
>> *From:* Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com
>> <avholloway+cisco-v...@gmail.com>]
>>
>> *Sent:* Wednesday, March 28, 2018 11:46 AM
>> *To:* Lelio Fulgenzi <le...@uoguelph.ca>
>> *Cc:* Brian Meade <bmead...@vt.edu>; Pete Brown <j...@chykn.com>;
>> cisco-voip@puck.nether.net
>> *Subject:* Re: [cisco-voip] Windows AXL Query Tool
>>
>> No not really.
>>
>>
>>
>> The work flow is like this:
>>
>>
>>
>> 1. Download the ZIP file from CUCM plugins page and extract to your PC.
>> It's simply called Cisco AXL Toolkit, but the file is axlsqltoolkit.zip
>>
>> 2. Open a command prompt to the working folder you extracted the files
>> to, such that you're in the same folder as the README.txt
>>
>> 3. Open the README.txt and make a copy of the command example for your
>> platform: windows or linux, and for me it's windows, so line 22 I will copy
>> to the line below the example (CTRL+d in Notepad++)
>>
>> 4. Correct the username and password, host for your environment
>>
>> 5. Add the input/output file switches: -input=test.xml -output=test.txt
>> (we'll create the test.xml in the next step, and it will create the
>> test.txt file for you)
>>
>> 6. Make a copy of the sample.xml file and we'll name it test.xml for this
>> exercise
>>
>> 7. Edit the test.xml so it only contains 1 query; we'll use this one for
>> testing: 
>>
>> 8. Copy your command line from the README.txt and paste it into your
>> command prompt
>>
>> 9. Watch the activity scroll up the screen, and check your test.txt file
>> for the output
>>
>>
>>
>> Be prepared: it's SOAP and it's ugly.  So, don't be expecting 

Re: [cisco-voip] Windows AXL Query Tool

2018-03-28 Thread Bill Talley
Lelio,

The AXL toolkit is what the python CLI scripts I posted earlier are based on.  
The script imports the wsdl file from the AXL toolkit and uses it to query 
CUCM.  The links I posted explain the setup required.

As far as I know, all AXL/SOAP scripts or webpages regardless of programming 
language will use the same AXL toolkit or components of it to access CUCM.  
Someone correct me if I’m wrong please.

I’m cleaning up the full python script as well as working on a GUI and 
distributable package for the script, but hopefully they will give you an idea 
of how to get started with Python.  It can seem overwhelming at first, but it’s 
not that bad.  As many people can probably tell from the scripts, I’m not a 
programmer but once you pick up the basics, it’s can be quit powerful even with 
limited experience.

Bill

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

> On Mar 28, 2018, at 10:52 AM, Lelio Fulgenzi  wrote:
> 
> Ok. This is a great help. I’ll have to see which is better for me (or which 
> is easier to use) – this toolkit or CLI.
>  
>  
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] 
> Sent: Wednesday, March 28, 2018 11:46 AM
> To: Lelio Fulgenzi 
> Cc: Brian Meade ; Pete Brown ; 
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Windows AXL Query Tool
>  
> No not really.
>  
> The work flow is like this:
>  
> 1. Download the ZIP file from CUCM plugins page and extract to your PC.  It's 
> simply called Cisco AXL Toolkit, but the file is axlsqltoolkit.zip
> 2. Open a command prompt to the working folder you extracted the files to, 
> such that you're in the same folder as the README.txt
> 3. Open the README.txt and make a copy of the command example for your 
> platform: windows or linux, and for me it's windows, so line 22 I will copy 
> to the line below the example (CTRL+d in Notepad++)
> 4. Correct the username and password, host for your environment
> 5. Add the input/output file switches: -input=test.xml -output=test.txt 
> (we'll create the test.xml in the next step, and it will create the test.txt 
> file for you)
> 6. Make a copy of the sample.xml file and we'll name it test.xml for this 
> exercise
> 7. Edit the test.xml so it only contains 1 query; we'll use this one for 
> testing: 
> 8. Copy your command line from the README.txt and paste it into your command 
> prompt
> 9. Watch the activity scroll up the screen, and check your test.txt file for 
> the output
>  
> Be prepared: it's SOAP and it's ugly.  So, don't be expecting CSV style data 
> formatting here.  I typically only use this tool when I need to run UPDATE 
> commands, like lots of them, to make changes to the DB, which is few and far 
> between.
>  
> More commonly, I'm using UPDATE commands right on the CLIwith careful 
> WHERE clausesor even more commonly, I'm just looking at data with SELECT 
> commands.
>  
> YMMV
>  
> On Wed, Mar 28, 2018 at 9:22 AM Lelio Fulgenzi  wrote:
> Is it “wizardish” in any way? Like CUDDLI?
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
> 
>  
> From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] 
> Sent: Wednesday, March 28, 2018 10:19 AM
> To: Brian Meade 
> Cc: Lelio Fulgenzi ; Pete Brown ; 
> cisco-voip@puck.nether.net
> 
> Subject: Re: [cisco-voip] Windows AXL Query Tool
>  
> Yeah, it's really just for batch SQL commands, otherwise, just SSHing to the 
> Publisher is easiest.
>  
> On Wed, Mar 28, 2018 at 9:16 AM Brian Meade  wrote:
> Check under Application->Plugins
>  
> On Wed, Mar 28, 2018 at 10:03 AM, Lelio Fulgenzi  wrote:
> Wait. What? There’s an AXL SQL tool kit?
>  
> ---
> Lelio Fulgenzi, B.A. | Senior Analyst
> Computing and Communications Services | University of Guelph
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
> 2W1
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>  
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>  
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Anthony Holloway
> Sent: Tuesday, March 27, 2018 9:58 PM
> To: Pete Brown 
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Windows 

Re: [cisco-voip] Check on calls in queue for a team?

2018-03-22 Thread Bill Talley
I’m behind all of the emails but am curious about how the light gets triggered. 
 Is it a SIP device registered to CUCM or how does that work?What version 
of CCX are you running?

I’m wondering if you could do a REST call to the CCX database for contacts 
waiting and parse the XML?  Might be easier to capture the data that way?

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

> On Mar 22, 2018, at 11:20 AM, Matthew Loraditch 
>  wrote:
> 
> I'm setting up that light system I’ve talked about and the managers wants 
> their lights to come on if any queue assigned to a team has calls waiting. I 
> can't think of any way to do that but iterate through every queue in a very 
> long nested if of get reporting statistics steps and anytime we add a queue I 
> will have to modify the script...
>  
> One team has 22 queues assigned…
>  
> If anyone has a better idea, I'd love to hear it!
>   
> Matthew Loraditch​
> Sr. Network Engineer
> p: 443.541.1518
> w: www.heliontechnologies.com  |  e: mloradi...@heliontechnologies.com
> 
> 
> 
> 
> 
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Re: [cisco-voip] building an API box - what do I need?

2018-03-21 Thread Bill Talley
Hi Lelio,

I just went through the same ordeal a few weeks ago which you're embarking
on now.   I'm no expert in this by any means, but know information is also
hard to come by and will share whatever I can.

Checkout my notes here,
https://www.iptinfo.com/2018/02/22/querying-cucm-via-axl-using-python-scripts-installing-python/,
but more importantly check out the links at the bottom of the page.  Those
are blogs I used to get started on installing and using Python to utilize
AXL to query and modify CUCM.

Here's a link to my sample Python script which will prompt you for a
client/directory name, CUCM IP address, CUCM version, CUCM AXL username and
CUCM AXL password, then will query for and output a list of phone (basic
info) and create a csv file containing that list of phones.  I have a more
complex script that queries for cluster nodes, software versions, phone
devices, hunt devices, gateways, media devices, etc if you're interested in
seeing something like that.

https://www.iptinfo.com/2018/02/28/querying-cucm-via-axl-sample-python-script/

As far as username and passwords embedded in the script, if you control the
machine the script will be executed on, you could create system variables
containing that info on the machine, then call the system variables from
the script.

Lastly, here's a link to the CUCM AXL developer guides and database
dictionaries.  The AXL programming guides tell you the supported search
criteria and response data that will be returned.

https://developer.cisco.com/docs/axl/#previous-versions


Bill

On Wed, Mar 21, 2018 at 3:17 PM, Lelio Fulgenzi  wrote:

>
> I'm going to make a (sad) attempt to try my hand at API usage. You know,
> running those sweet commands against CUCM or WebEx, etc to make my life
> easier. I'm hoping that by starting with some simple examples, I can build
> what I need as time goes on.
>
> My hope is to build scripts that I can run via CLI, but also create simple
> web pages with buttons that execute those scripts, with or without
> arguments. The webpages themselves will be protected appropriately.
>
> But I'm starting from scratch. I do have a resource that I'm hoping will
> help out, but I'm wondering, if you were building an api box from scratch,
> what sort of things would you want (or need) on it to make things run
> smoothly?
>
> I'm guessing if I want to automate any of these scripts, I'll have no
> choice but to store the userID/password either in the script or read from a
> file. I've asked this before of a few people, but there is no real way to
> store a password securely and decrypt it at time of running the script.
>
> For interactive scripts, either CLI or web form, do you use the same
> userID/password, or are you asking for credentials and use those
> credentials?
>
> I know... a big ask.
>
>
> ---
> 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]
>
>
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Re: [cisco-voip] e911

2018-03-08 Thread Bill Talley
Great points Ryan.  

We have a standard 9-1-1 disclaimer in our scopes (as I’m sure many VARs do), 
though it primarily focuses on location and alerting versus access.  I would 
imagine it would now also need to include reference to this mandate prohibiting 
restrictions to direct access of 9-1-1 services... i.e. call it out BEFORE the 
sale to establish baseline notification to the customer.

We also use the Uplinx Reporting Tool (unpaid plug) to capture a snapshot of 
the CUCM configuration database during handoff to customers,  as well as 
pre-upgrade and post-upgrade snapshots during upgrades of existing systems.  
Something like Uplinx or AXL/SOAP exports, BAT exports, or anything with time 
stamps bound to the config data will be crucial for everyone going forward.

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

> On Mar 7, 2018, at 9:16 PM, Ryan Huff <ryanh...@outlook.com> wrote:
> 
> Legal language aside, I see this as a HUGE area for VARs to get into civil 
> torts with customers.
> 
> Ideally the end customer is the true owner and stakeholder of the MLTS 
> however; when levied with a government fine (presumably how it would be 
> handled), due to e911 malfeasance, who was the last one to touch it?
> 
> Document everything, get sign off on everything and proceed with caution :) 
> brothers and sisters.
> 
> Sent from my iPhone
> 
> On Mar 7, 2018, at 22:10, Anthony Holloway <avholloway+cisco-v...@gmail.com> 
> wrote:
> 
>> I'd be cautious with this one.
>> 
>> 1) You penalize actual emergency calls from connecting as quickly as 
>> possible.  Do you really want to be the person responsible for that?
>> 
>> 2) You penalize the entire cluster by changing a global parameter, for the 
>> occasional accidental 911 call.
>> 
>> I think a better solution is to solve the human problem.  Just like we 
>> wouldn't tolerate our children playing on land lines or cell phones calling 
>> 911 (even my son has done it), we shouldn't tolerate adults doing it either.
>> 
>> Failing that, switch your PSTN trunk access code to another digit.  8 seems 
>> to be a popular second choice.
>> 
>>> On Wed, Mar 7, 2018 at 3:41 PM NateCCIE <natec...@gmail.com> wrote:
>>> This might be a good time to talk about my favorite way to enable 911.
>>> 
>>>  
>>> 
>>> Set the interdigit timeout to a small value, like 3-5 seconds.  Then create 
>>> a 911 route pattern, and a 911! Pattern, that does not route to 911.  If 
>>> the user dials 911 and stops, the call connects.  If they keep dialing 
>>> which usually what happens on a miss-dial, they get whatever your 911! 
>>> Pattern is configured to do, usually I like block this pattern.
>>> 
>>>  
>>> 
>>> -Nate
>>> 
>>>  
>>> 
>>> From: Bill Talley <btal...@gmail.com> 
>>> Sent: Wednesday, March 7, 2018 2:22 PM
>>> To: Matthew Loraditch <mloradi...@heliontechnologies.com>
>>> Cc: NateCCIE <natec...@gmail.com>; Ryan Huff <ryanh...@outlook.com>; 
>>> cisco-voip@puck.nether.net
>>> Subject: Re: [cisco-voip] e911
>>> 
>>>  
>>> 
>>> Seems like there's two key aspects we need to be concerned with.  1) As I 
>>> think Matthew is pointing out, notifications are only required if 
>>> notifications are a native feature available "without improvement", i.e. 
>>> add-on components.  2)  We now MUST configure direct 911 access without 
>>> regard to customer complaints or PSAP complaints about accidental 911 calls.
>>> 
>>>  
>>> 
>>> To answer your question Matthew, I have only ever used CER and Singlewire 
>>> for notifications, sorry I can't provide more feedback.
>>> 
>>>  
>>> 
>>> On Wed, Mar 7, 2018 at 3:06 PM, Matthew Loraditch 
>>> <mloradi...@heliontechnologies.com> wrote:
>>> 
>>> As far as I know that feature doesn’t notify anyone internally.
>>> 
>>> The part of the law I’m referring to is this:
>>> 
>>>  
>>> 
>>> “A person engaged in the business of installing, managing, or operating 
>>> multi-line telephone systems shall, in installing, managing, or operating 
>>> such a system for use in the United States, configure the system to provide 
>>> a notification to a central location at the facility where the system is 
>>> installed or to another person or organization regardless of location, if 
>>> the system is able to be configured to provide the not

Re: [cisco-voip] e911

2018-03-07 Thread Bill Talley
Digging further, does this mean the law doesn't go into affect until Feb,
2020?

(b) Effective Date.—The amendment made by subsection (a) shall apply with
respect to a multi-line telephone system that is manufactured, imported,
offered for first sale or lease, first sold or leased, or installed after
the date that is 2 years after the date of the enactment of this Act.

On Wed, Mar 7, 2018 at 2:11 PM, Ryan Huff  wrote:

> I wonder how cloud-based phone system like Cisco spark will answer this?
>
>
> https://www.linkedin.com/pulse/karis-law-you-compliant-edgar-salazar
>
> Sent from my iPhone
>
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Re: [cisco-voip] e911

2018-03-07 Thread Bill Talley
Seems like there's two key aspects we need to be concerned with.  1) As I
think Matthew is pointing out, notifications are only required if
notifications are a native feature available "without improvement", i.e.
add-on components.  2)  We now MUST configure direct 911 access without
regard to customer complaints or PSAP complaints about accidental 911 calls.

To answer your question Matthew, I have only ever used CER and Singlewire
for notifications, sorry I can't provide more feedback.

On Wed, Mar 7, 2018 at 3:06 PM, Matthew Loraditch <
mloradi...@heliontechnologies.com> wrote:

> As far as I know that feature doesn’t notify anyone internally.
>
> The part of the law I’m referring to is this:
>
>
>
> *“A person engaged in the business of installing, managing, or operating
> multi-line telephone systems shall, in installing, managing, or operating
> such a system for use in the United States, configure the system to provide
> a notification to a central location at the facility where the system is
> installed or to another person or organization regardless of location, if
> the system is able to be configured to provide the notification without an
> improvement to the hardware or software of the system.”*
>
>
>
>
>
>
>
> Matthew Loraditch
> Sr. Network Engineer
> p: *443.541.1518* <443.541.1518>
> w: *www.heliontechnologies.com*   |
> e: *mloradi...@heliontechnologies.com* 
> [image: Facebook] 
> [image: Twitter] 
> [image: LinkedIn] 
>
>
> *From:* NateCCIE [mailto:natec...@gmail.com]
> *Sent:* Wednesday, March 7, 2018 3:58 PM
> *To:* Matthew Loraditch ; 'Ryan Huff' <
> ryanh...@outlook.com>; cisco-voip@puck.nether.net
> *Subject:* RE: [cisco-voip] e911
>
>
>
> Um, I thought it did.
>
>
>
> https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-
> communications-manager-callmanager/200452-Usage-of-
> Native-Emergency-Call-Routing-F.html
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Matthew
> Loraditch
> *Sent:* Wednesday, March 7, 2018 1:36 PM
> *To:* Ryan Huff ; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] e911
>
>
>
> To piggy back on this, while Cisco doesn’t have emergency notifications
> built in, as the law mentions, and thus they are not required, does anyone
> know of options beyond Singlewire that they are happy with? The installs
> would monitor up to 1000 or so handsets but the folks that would be
> notified would probably be fewer than 50.
>
>
>
>
>
>
>
> *Matthew Loraditch*
>
> *Sr. Network Engineer*
>
> p: *443.541.1518* <443.541.1518>
>
> w: *www.heliontechnologies.com* 
>
>  |
>
> e: *mloradi...@heliontechnologies.com* 
>
> [image: Facebook] 
>
> [image: Twitter] 
>
> [image: LinkedIn] 
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
> ] *On Behalf Of *Ryan Huff
> *Sent:* Wednesday, March 7, 2018 3:11 PM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] e911
>
>
>
> I wonder how cloud-based phone system like Cisco spark will answer this?
>
>
>
>
>
> https://www.linkedin.com/pulse/karis-law-you-compliant-edgar-salazar
>
> Sent from my iPhone
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] IP Phones + MGCP FXS + Shared Lines

2017-12-21 Thread Bill Talley
How about this?  It's the closest I've seen to what you're asking about.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmsys/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100_chapter_010001.html#CUCM_RF_S05975F9_00





On Mon, Dec 18, 2017 at 4:48 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> No unfortunately, just gotta play the cards dealt on this one.
>
> Actually, I'm not interested in changing anything, I'm looking for
> documentation that supports what these two Cisco employee have publicly
> stated.  I personally will only cite sources from official documentation,
> and not from web forums, regardless of who posted it.  Because, at the end
> of the day, that employee can be wrong, and Cisco does not have to take
> responsibility for the infraction.  However, in documentation they do.
> Either they fix the documentation, or they fix the technology.
>
> On Sat, Dec 16, 2017 at 4:00 PM Lelio Fulgenzi  wrote:
>
>>
>> Looks like the thread points you to sccp controlled fxs ports. I'm sure
>> you have your reasons for wanting MGCP. Curious what those are.
>>
>> Could you get away with ATAs?
>>
>> Sent from my iPhone
>>
>> On Dec 16, 2017, at 1:50 AM, Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>> Does anyone have current information on whether or not MGCP FXS ports can
>> be on shared lines with Cisco IP Phones?
>>
>> No anecdotal or empirical evidence please.  I'm looking for documented
>> facts, preferably the kind that doesn't require re-reading it like 10 times
>> to come up with your own interpretation.
>>
>> Check out what the two Cisco Employees are saying in this thread from
>> 2010 (Spoiler - It's not supported)
>>
>> https://supportforums.cisco.com/t5/ip-telephony/shared-
>> line-between-ip-phones-and-fxs-port-with-sccp-setup/td-p/1045431
>>
>> Are you a Cisco employee who likes to stick your neck out for internet
>> strangers, and also believe this to be unsupported?  Please contact me.  Or
>> not.  I'm not crying.
>>
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
> ___
> cisco-voip mailing list
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

2017-12-16 Thread Bill Talley
Found this bug/feature enhancement.   I'll be opening a case and
associating it to the feature request.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvg62384

Phone text illegible on different backgrounds.
CSCvg62384
Description
Changing font colors or toggle font boxes for DNs would allow users to
personalize the phone aloowing them to mix between backgrounds and font
colurs.

Symptom:
On 88xx phones with 12.0.1.SR1, the white DN box was removed to allow the
background to be visible.
Now, with some background images, the DN numbers illegible because the
colors are similar.

Conditions:
88xx phones on version 12.0.1.SR1

Workaround:
Change the background image to one that allows the DNs to be distinguished.

Further Problem Description:
Enhancement request to allow more personalized changes on the phones.
Option 1:
The color should be selectable in the settings menu
Font Color:
Auto- Default
Black
White
Other?
This would allow Users to select any of the existing or custom Images on
the phone.

Option 2:
Toggle Font Boxes
Make this a selectable option in the phone

Option 3:
The Font has opposing color outline
White Font has black outline
Black font has White Outline

On Fri, Dec 15, 2017 at 10:38 AM, Haas, Neal <nh...@co.fresno.ca.us> wrote:

> We are creating lighter backgrounds to resolve the not active line color.
> Also I just noticed no Text on the top of the screen!
>
>
>
> Thank You,
>
>
>
> Neal Haas
>
> NSE, Communications
>
> *Please report Troubles to the Help Desk. 559-600-5900 <(559)%20600-5900>*
>
> Telephone (559) 600-5890
>
>
>
> *From:* Bill Talley [mailto:btal...@gmail.com]
> *Sent:* Thursday, December 14, 2017 7:51 AM
>
> *To:* JASON BURWELL <jason.burw...@foundersfcu.com>
> *Cc:* Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> Thanks for your feedback Jason.  I'll try a more solid background and will
> report back to the group with results.
>
>
>
> On Thu, Dec 14, 2017 at 9:47 AM, JASON BURWELL <
> jason.burw...@foundersfcu.com> wrote:
>
> Yes, dark wallpapers bring out plain white text for us. It’s been like
> this since upgrading off of I think FW11.5x or whatever version created an
> entire white block for each label which pretty much killed the wallpapers.
> We are using wallpaper that is mostly solid but a watermark of the company
> logo does appear under the text and it looks fine. None of our wallpapers
> use gradients though. Maybe that’s the issue, it having a problem
> interpreting and deciding what color text to use? This is on 8851 and 8865.
>
>
>
> Jason
>
>
>
> *From:* Bill Talley [mailto:btal...@gmail.com]
> *Sent:* Thursday, December 14, 2017 10:22 AM
>
>
> *To:* JASON BURWELL <jason.burw...@foundersfcu.com>
> *Cc:* Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> So your dark backgrounds on 12.0.1SR1 have white text?   Are they solid
> backgrounds or gradients?  We had white text over the same background on
> firmware 11.7.1.17 but after upgrading to 12.0.1SR1 the text changed to
> gray.   Downgrading to 11.7.1.17 reverted text to white.  Using solely
> 8845s and CCM 11.5.1.SU3 on our end.
>
> Sent from a mobile device with very tiny touchscreen input keys. Please
> excude my typtos.
>
>
> On Dec 14, 2017, at 9:06 AM, JASON BURWELL <jason.burw...@foundersfcu.com>
> wrote:
>
> Not sure, I saw your picture and I have not had that issue anywhere on the
> 12x train. We have light and dark wallpapers. The Text for Speed
> Dials/Features changes automatically between Black and White color text
> depending on the wallpaper chosen so the text always clear and visible. I
> do recall an issue with text being seen back in 11.5 or 11.7 somewhere.
> I’ve been through a lot of ES releases prior to 12 so hard to remember.
>
>
>
>
>
> Jason
>
>
>
> *From:* Bill Talley [mailto:btal...@gmail.com <btal...@gmail.com>]
> *Sent:* Thursday, December 14, 2017 9:07 AM
> *To:* JASON BURWELL <jason.burw...@foundersfcu.com>
> *Cc:* Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> Issue with display text still exists on SR1.  That’s what we were using.
>
> Sent from a mobile device with very tiny touchscreen input keys. Please
> excude my typtos.
>
>
> On Dec 14, 2017, at 7:24 AM, JASON

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

2017-12-14 Thread Bill Talley
Thanks for your feedback Jason.  I'll try a more solid background and will
report back to the group with results.

On Thu, Dec 14, 2017 at 9:47 AM, JASON BURWELL <
jason.burw...@foundersfcu.com> wrote:

> Yes, dark wallpapers bring out plain white text for us. It’s been like
> this since upgrading off of I think FW11.5x or whatever version created an
> entire white block for each label which pretty much killed the wallpapers.
> We are using wallpaper that is mostly solid but a watermark of the company
> logo does appear under the text and it looks fine. None of our wallpapers
> use gradients though. Maybe that’s the issue, it having a problem
> interpreting and deciding what color text to use? This is on 8851 and 8865.
>
>
>
> Jason
>
>
>
> *From:* Bill Talley [mailto:btal...@gmail.com]
> *Sent:* Thursday, December 14, 2017 10:22 AM
>
> *To:* JASON BURWELL <jason.burw...@foundersfcu.com>
> *Cc:* Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> So your dark backgrounds on 12.0.1SR1 have white text?   Are they solid
> backgrounds or gradients?  We had white text over the same background on
> firmware 11.7.1.17 but after upgrading to 12.0.1SR1 the text changed to
> gray.   Downgrading to 11.7.1.17 reverted text to white.  Using solely
> 8845s and CCM 11.5.1.SU3 on our end.
>
> Sent from a mobile device with very tiny touchscreen input keys. Please
> excude my typtos.
>
>
> On Dec 14, 2017, at 9:06 AM, JASON BURWELL <jason.burw...@foundersfcu.com>
> wrote:
>
> Not sure, I saw your picture and I have not had that issue anywhere on the
> 12x train. We have light and dark wallpapers. The Text for Speed
> Dials/Features changes automatically between Black and White color text
> depending on the wallpaper chosen so the text always clear and visible. I
> do recall an issue with text being seen back in 11.5 or 11.7 somewhere.
> I’ve been through a lot of ES releases prior to 12 so hard to remember.
>
>
>
>
>
> Jason
>
>
>
> *From:* Bill Talley [mailto:btal...@gmail.com <btal...@gmail.com>]
> *Sent:* Thursday, December 14, 2017 9:07 AM
> *To:* JASON BURWELL <jason.burw...@foundersfcu.com>
> *Cc:* Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> Issue with display text still exists on SR1.  That’s what we were using.
>
> Sent from a mobile device with very tiny touchscreen input keys. Please
> excude my typtos.
>
>
> On Dec 14, 2017, at 7:24 AM, JASON BURWELL <jason.burw...@foundersfcu.com>
> wrote:
>
> I recommend loading the latest available firmware for the 8800 phones.
> Right now 12-0-1SR1-1 is available. In some cases I have had to request the
> latest ES form TAC. The 11.7x firmware has been some of the buggiest phone
> firmware I have seen for Cisco phones in the last 15 years. 12x seems to be
> much better than 11.7 but I am still getting issues with phones freezing up
> occasionally and other weird problems. Waiting to see if 12-0-1SR1-1
> finally completely resolves those problems.
>
>
>
> Jason
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
> <cisco-voip-boun...@puck.nether.net>] *On Behalf Of *Haas, Neal
> *Sent:* Thursday, December 14, 2017 2:01 AM
> *To:* Bill Talley <btal...@gmail.com>; NateCCIE <natec...@gmail.com>
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> Thats exactly what I am seeing. When you change the background to a
> lighter coloring, you can read everything. I will need to put my glasses on
> to see if it is blurry..
>
>
>
> Fine quality control again by Cisco. I have already started the complaint
> train with TAC, tomorrow I will start with the account rep.
>
>
>
> Neal
> --
>
> *From:* Bill Talley <btal...@gmail.com>
> *Sent:* Wednesday, December 13, 2017 9:58:01 PM
> *To:* NateCCIE
> *Cc:* Haas, Neal; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text
> color
>
>
>
> The firmware didn’t adjust the text color in our case... I blurred out
> some characters before posting. He barely visible characters say”DID - 316”
> as well as “Brian” next to the speed dial button.
>
> 
>
>
>
> Sent from a mobile device with very tiny touchscreen input keys. Please
> excude my typtos.
>
> >

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

2017-12-14 Thread Bill Talley
So your dark backgrounds on 12.0.1SR1 have white text?   Are they solid 
backgrounds or gradients?  We had white text over the same background on 
firmware 11.7.1.17 but after upgrading to 12.0.1SR1 the text changed to gray.   
Downgrading to 11.7.1.17 reverted text to white.  Using solely 8845s and CCM 
11.5.1.SU3 on our end.

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

> On Dec 14, 2017, at 9:06 AM, JASON BURWELL <jason.burw...@foundersfcu.com> 
> wrote:
> 
> Not sure, I saw your picture and I have not had that issue anywhere on the 
> 12x train. We have light and dark wallpapers. The Text for Speed 
> Dials/Features changes automatically between Black and White color text 
> depending on the wallpaper chosen so the text always clear and visible. I do 
> recall an issue with text being seen back in 11.5 or 11.7 somewhere. I’ve 
> been through a lot of ES releases prior to 12 so hard to remember.
>  
>  
> Jason
>  
> From: Bill Talley [mailto:btal...@gmail.com] 
> Sent: Thursday, December 14, 2017 9:07 AM
> To: JASON BURWELL <jason.burw...@foundersfcu.com>
> Cc: Haas, Neal <nh...@co.fresno.ca.us>; NateCCIE <natec...@gmail.com>; 
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
>  
> Issue with display text still exists on SR1.  That’s what we were using.
> 
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
> 
> On Dec 14, 2017, at 7:24 AM, JASON BURWELL <jason.burw...@foundersfcu.com> 
> wrote:
> 
> I recommend loading the latest available firmware for the 8800 phones. Right 
> now 12-0-1SR1-1 is available. In some cases I have had to request the latest 
> ES form TAC. The 11.7x firmware has been some of the buggiest phone firmware 
> I have seen for Cisco phones in the last 15 years. 12x seems to be much 
> better than 11.7 but I am still getting issues with phones freezing up 
> occasionally and other weird problems. Waiting to see if 12-0-1SR1-1 finally 
> completely resolves those problems.
>  
> Jason
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Haas, Neal
> Sent: Thursday, December 14, 2017 2:01 AM
> To: Bill Talley <btal...@gmail.com>; NateCCIE <natec...@gmail.com>
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
>  
> Thats exactly what I am seeing. When you change the background to a lighter 
> coloring, you can read everything. I will need to put my glasses on to see if 
> it is blurry..
>  
> Fine quality control again by Cisco. I have already started the complaint 
> train with TAC, tomorrow I will start with the account rep.
>  
> Neal 
> From: Bill Talley <btal...@gmail.com>
> Sent: Wednesday, December 13, 2017 9:58:01 PM
> To: NateCCIE
> Cc: Haas, Neal; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
>  
> The firmware didn’t adjust the text color in our case... I blurred out some 
> characters before posting. He barely visible characters say”DID - 316” as 
> well as “Brian” next to the speed dial button.
> 
> 
> 
> 
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
> 
> > On Dec 13, 2017, at 11:17 PM, NateCCIE <natec...@gmail.com> wrote:
> > 
> > The firmware has a feature to detect if the background is dark or light, 
> > and then adjust the labels as needed. But it has to be really dark to be 
> > dark. For the majority of picture. 
> > 
> > Sent from my iPhone
> > 
> >> On Dec 13, 2017, at 7:50 PM, Haas, Neal <nh...@co.fresno.ca.us> wrote:
> >> 
> >> I wanted to push the new 12.0(1) firmware to our 8851/8861 phones. I did a 
> >> test and you cannot even see the line text on dark backgrounds, which is 
> >> 90% of the images . Anyone know how to fix line text color? The only image 
> >> that really works is the one that looks like someone socked the phone.
> >> 
> >> Neal
> >> 
> >> 
> >> ___
> >> cisco-voip mailing list
> >> cisco-voip@puck.nether.net
> >> https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip=02%7C01%7Cnhaas%40co.fresno.ca.us%7C028644c42a504bc5494b08d542b7a54c%7C3ccce0182cd74123960d6cc1d47e3550%7C1%7C0%7C636488278898591039=B1ck73ntCTF4hRNDHkrPXYzNTx4jnNLGYRRLg3kxkrE%3D=0
> > ___
> > cisco-voip mailing list
> > cisco-voip@puc

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

2017-12-14 Thread Bill Talley
Issue with display text still exists on SR1.  That’s what we were using.

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

> On Dec 14, 2017, at 7:24 AM, JASON BURWELL <jason.burw...@foundersfcu.com> 
> wrote:
> 
> I recommend loading the latest available firmware for the 8800 phones. Right 
> now 12-0-1SR1-1 is available. In some cases I have had to request the latest 
> ES form  TAC. The 11.7x firmware has been some of the buggiest phone firmware 
> I have seen for Cisco phones in the last 15 years. 12x seems to be much 
> better than 11.7 but I am still getting issues with phones freezing up 
> occasionally and other weird problems. Waiting to see if 12-0-1SR1-1 finally 
> completely resolves those problems.
>  
> Jason
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Haas, Neal
> Sent: Thursday, December 14, 2017 2:01 AM
> To: Bill Talley <btal...@gmail.com>; NateCCIE <natec...@gmail.com>
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
>  
> Thats exactly what I am seeing. When you change the background to a lighter 
> coloring, you can read everything. I will need to put my glasses on to see if 
> it is blurry..
>  
> Fine quality control again by Cisco. I have already started the complaint 
> train with TAC, tomorrow I will start with the account rep.
>  
> Neal 
> From: Bill Talley <btal...@gmail.com>
> Sent: Wednesday, December 13, 2017 9:58:01 PM
> To: NateCCIE
> Cc: Haas, Neal; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
>  
> The firmware didn’t adjust the text color in our case... I blurred out some 
> characters before posting. He barely visible characters say”DID - 316” as 
> well as “Brian” next to the speed dial button.
> 
> 
> 
> 
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
> 
> > On Dec 13, 2017, at 11:17 PM, NateCCIE <natec...@gmail.com> wrote:
> > 
> > The firmware has a feature to detect if the background is dark or light, 
> > and then adjust the labels as needed. But it has to be really dark to be 
> > dark. For the majority of picture. 
> > 
> > Sent from my iPhone
> > 
> >> On Dec 13, 2017, at 7:50 PM, Haas, Neal <nh...@co.fresno.ca.us> wrote:
> >> 
> >> I wanted to push the new 12.0(1) firmware to our 8851/8861 phones. I did a 
> >> test and you cannot even see the line text on dark backgrounds, which is 
> >> 90% of the images . Anyone know how to fix line text color? The only image 
> >> that really works is the one that looks like someone socked the phone.
> >> 
> >> Neal
> >> 
> >> 
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Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

2017-12-13 Thread Bill Talley
Nothing aside from the changes in color to the text and to the toast pop up.  
Toast was orange, now it’s white.  Opinions vary on that one though.

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

> On Dec 13, 2017, at 9:23 PM, Haas, Neal <nh...@co.fresno.ca.us> wrote:
> 
> I want to do that but we are hitting the answer/hold bug.
> 
> 
> I think my solution is to create a background image that doesn't look as bad 
> as the default cisco one.
> 
> 
> any other issues that you have seen with 12.0(1)?
> 
> 
> Neal Haas
> ITSD Analyst - Communications
> 559-600-5890
> 
> From: Bill Talley <btal...@gmail.com>
> Sent: Wednesday, December 13, 2017 7:12:32 PM
> To: Haas, Neal
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color
> 
> We have the same issue and have been unable to find a fix short of reverting 
> to 11.7 firmware.
> 
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
> 
>> On Dec 13, 2017, at 8:50 PM, Haas, Neal <nh...@co.fresno.ca.us> wrote:
>> 
>> I wanted to push the new 12.0(1) firmware to our 8851/8861 phones. I did a 
>> test and you cannot even see the line text on dark backgrounds, which is 90% 
>> of the images . Anyone know how to fix line text color? The only image that 
>> really works is the one that looks like someone socked the phone.
>> 
>> Neal
>> 
>> 
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Re: [cisco-voip] 8800 Series firmware upgrade 12.0(1) text color

2017-12-13 Thread Bill Talley
We have the same issue and have been unable to find a fix short of reverting to 
11.7 firmware.  

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

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

2017-10-03 Thread Bill Talley
I’ve always used a CTI route point to forward those calls to CUC, then used a 
forward routing rule to the sign in conversation with the criteria being 
forwarding station is 7000 where 7000 is the 4-digit DN of the route point.

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> On Oct 3, 2017, at 10:52 AM, Matthew Huff  wrote:
> 
> If we have an externally available number to access Unity voicemail, is there 
> anyway of changing it so that it doesn't prompt asking for phone number to 
> dial? You can hit * to enter your extension and passcode, but only if you 
> remember to hit *.
> 
> 
> 
> 
> Matthew Huff | 1 Manhattanville Rd 
> Director of Operations   | Purchase, NY 10577
> OTA Management LLC   | Phone: 914-460-4039
> aim: matthewbhuff| Fax:   914-694-5669
> 
> 
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Re: [cisco-voip] e164 dialplan conversion

2017-09-23 Thread Bill Talley
Thanks for the suggestions Bernhard and Florian.

On Thu, Sep 21, 2017 at 9:03 AM, Bernhard Albler <bernhard.alb...@gmail.com>
wrote:

> ucm:
> What Florian says is the correct way to do it.
> You can also run a sql update against the numplan table which will do 2
> things:
> 1.)I you typoed you are in for a lot of pain
> 2.)Ucm will load up on change notification (assuming lots of DNs) and
> might come to halt before recovering
>
> ccx:
> Agent DNs in my experience will reflect automatically. Routepoints and CTI
> ports you will need to change manually.
>
>
> Personally i suggest the SQL route as the adrenaline rush is just
> incredible once you realize you made a mistake.
>
> On Thu, Sep 21, 2017 at 3:20 PM, Florian Kroessbacher <
> florian.kroessbac...@gmail.com> wrote:
>
>> Hy out there,
>>
>> throug updateLine this is working
>>
>> use old pattern & partition
>> and set newpattern
>>
>> we have done this for 12000 Lines
>>
>> Am 21. Sep. 2017, 15:13 +0200 schrieb Bill Talley <btal...@gmail.com>:
>>
>> I know there have been some conversations around this in the past, but
>> I’m hoping there are new methods for converting from a 4-digit to full e164
>> dialplan.
>>
>> Is there a way to change dialplan entries on CUCM and UCCX without having
>> to either individually touch each dialplan pattern or without deleting and
>> reimporting/recreating? Has anyone seen or used AXL/SOAP to automate
>> modification of dialplan patterns, or used PCP or some third party product
>> to accomplish this?
>>
>> Thanks for any feedback.
>>
>> Bill
>>
>> Sent from a mobile device with very tiny touchscreen input keys. Please
>> excude my typtos.
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>>
>
>
> --
> Bernhard Albler, +4369917207384 <+43%20699%2017207384>
> --
> "Was Nachwelt! Wie komm' ich dazu was für die Nachwelt zu tun? Was hat
> denn die Nachwelt für mich getan?"
> --Carl Friedrich Zelter
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[cisco-voip] e164 dialplan conversion

2017-09-21 Thread Bill Talley
I know there have been some conversations around this in the past, but I’m 
hoping there are new methods for converting from a 4-digit to full e164 
dialplan.

Is there a way to change dialplan entries on CUCM and UCCX without having to 
either individually touch each dialplan pattern or without deleting and 
reimporting/recreating?  Has anyone seen or used AXL/SOAP to automate 
modification of dialplan patterns, or used PCP or some third party product to 
accomplish this?

Thanks for any feedback.

Bill

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Re: [cisco-voip] CUBE Binding Control on Dial-Peer With Active Calls

2017-09-05 Thread Bill Talley
Only thing I can think of regarding #1 is if the new dial peer changes the 
binding of an existing call as the result of a subsequent sip update message?

No comment re: #2

Re: #3 I've done it the exact same way you mentioned.

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> On Sep 5, 2017, at 4:44 PM, Anthony Holloway 
>  wrote:
> 
> I have a bone to pick with the people who wrote the code for IOS and how it 
> doesn't let you bind control on a net new dial-peer, which has no active 
> calls going through it, although, there are active calls on the CUBE as a 
> whole.
> 
> So, here's the deal.  I was trying to setup a new dial-peer for a specific 
> outgoing phone number to test some config changes.  I copied the normal 
> outgoing PSTN dial-peer to the ITSP, changed the destination-pattern to the 
> number I was testing with, and when I pasted it in to the CUBE, the old "Bind 
> will not take affect, there are active calls." error message popped up.
> 
> A "show sip-ua calls summary" confirmed plenty of call legs active on the 
> CUBE.
> 
> So, my next thing was to kill all active calls on the CUBE, which I have done 
> in the past with this:
> 
> voice service voip
>  shutdown forced
> !
> 
> However, after initiating that command, there were still some, but less, 
> active legs showing on the CUBE.  I waited a good 2 minutes or so for call 
> legs to clear, but some small amount just stayed active
> 
> I then had to further shutdown SIP like this:
> 
> voice service voip
>  sip
>   call service stop
> !
> 
> When finally, the rest of the calls cleared and I was able to bind control on 
> my new dial-peer.
> 
> I then reverse those two commands to get back up and running.
> 
> voice service voip
>  no shutdown forced
>  sip
>   no call service stop
> !
> 
> This was on a 2900 ISR running 15.5(3)M5.
> 
> Three questions:
> 
> 1) What could possibly be the decision process to decide that we cannot bind 
> control on net new dial-peers on CUBE while there are active calls.  Doesn't 
> make any sense to me.  So, if you have a CUBE that processes phones calls 
> 24/7, you have to drop all active calls to add any new dial-peers (albeit 
> ones requiring control binding)?  That seems pretty weak.  I feel like I 
> should have seen this coming, considering how many CUBEs I've configured over 
> the years, but unfortunately, this one surprised me.
> 
> 2) Why didn't the shutdown forced command clear all of the sip legs, despite 
> me having done this, on this very gateway even, in the past?  Are some legs 
> not considered "calls" and therefore are not impacted by the gateway shutdown?
> 
> 3) Outside of waiting until all calls are inactive, or dropping all calls, 
> how would you handle this type of scenario?
> 
> Thanks.
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Re: [cisco-voip] Naming conventions for Expressway-C / E

2017-09-04 Thread Bill Talley
Similar to Ryan, I usually try to include "expe" or "expc" as well when the 
longer name isn't allowed, depending on customers naming standards.

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> On Sep 4, 2017, at 7:05 AM, Ryan Huff  wrote:
> 
> I usually keep the names, "edge" and "control" incorporated in them somehow. 
> If I'm trying to save on typing and it's a single pair I might do, 
> "expcontrol" and "expedge". With clusters, I'll add more to the name that 
> denote its purpose or location.
> 
> Thanks,
> 
> Ryan
> 
> On Sep 4, 2017, at 6:17 AM, Matthew Huff  wrote:
> 
>> I’m about to deploy our first expressway cluster, what names do people use 
>> form expressway DNS names?
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Re: [cisco-voip] UCM Upgrade Poll

2017-08-23 Thread Bill Talley
1.  That's kind of a loaded question.   90% bugs where  bugs also includes the 
system is "functioning as designed".  Most failures I've encountered, as a 
partner, occur as a result of the applications heavy reliance on DNS (if 
configured) and NTP.  Would be nice if there was an option to bypass the error 
and proceed based on host file entries or hardware clock as opposed to outright 
failing because let's say, there are two different PTR records for the same 
host or DNS/NTP are down for maintenance (different server group).  I get the 
security/reliability piece of it, but why can't the developers permit the 
upgrade to complete and present the same  "DNS is unreachable/invalid" or "NTP 
is unreachable" message after the upgrade completes like they any other time of 
operation?

2.  Same amount of planning as I do when upgrades fail, about 8-12 hours.

Thanks for asking for feedback.


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> On Aug 23, 2017, at 8:37 AM, Ryan Ratliff (rratliff)  
> wrote:
> 
> Quick 2 question poll, feel free to unicast or share your response with the 
> group.
> 
> 1. When you or your customers have a UCM or IMP upgrade fail, what percentage 
> of failures are due to a bug vs something in the environment (user error, db 
> updates, etc)?  
> % bug:
> % not a bug:
> Yes it’s a very subjective question but that’s ok, use your judgement.
> 
> 2. When an upgrade goes smoothly with no issues, how much time do you put 
> into the planning and preparation for the upgrade (not the execution)?
> 
> Thanks,
> 
> -Ryan
> 
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Re: [cisco-voip] offline network - using NAT/PAT to connect to the outside world

2017-07-19 Thread Bill Talley
I have, on occasion built a Windows AD server on my offline network to
impersonate one of the production servers for LDAP authentication, DNS,
DHCP, etc.   You don't really need all of the user objects, you really just
need user objects you need to test with, so I normally create a few user
objects like the ldap bind account, and a few others just to test admin
access, jabber login, uccx agent login via AXL.  As long as an LDAP sync
doesn't run, I've never run into issues doing this.

For route/switch, I normally add a new vSwitch to the ESXi host, add vNICs
for the required VLANs, deploy the CSR-1000v OVA with one interface
assigned to the isolated network and another assigned to an interface on
another vSwitch or vNIC for production connectivity.  NAT and NTP are
performed on the CSR-1000v.

I've also done everything completely on an isolated network very minimally
using Hanewin DNS server and NTP/DHCP/NAT on the CSR-1000v.

>From a routing standpoint, I usually use host routes, no default route, no
routing protocols and strict ACLs.

On Wed, Jul 19, 2017 at 1:28 PM, Lelio Fulgenzi  wrote:

>
> A while back we had a good discussion about offline network upgrade
> preparation. A few people mentioned using NAT/PAT in order to connect to
> servers that would be too difficult to replicate in the off-line network.
>
> I'm trying to get my head around this and thought I'd share this
> simplified plan that I want to start off with.
>
> In the offline network we have:
>
>
> * a layer 3 switch that does the routing for the offline network
>
> * a router which does some other tasks like NTP, DHCP, etc.
>
> * layer 3 switch is connected to router
>
> Here's what I envision:
>
>
> * connect the router to a public network, i.e. network 45
>
> * configure NAT/PAT on the router so that all the inside networks
> (on the switch) are patted using the public network address
>
> * configure static routes on the router for all inside networks
> (on the switch) pointing to the switch
>
> * configure static routes on the router for specific outside
> networks pointing network 45 gw
>
> * configure static routes on the switch for specific outside
> networks pointing to the router interface
>
> The biggest issue here is that we have replicated many of our public
> networks in this offline network and there can't be any chance of that
> leaking.
>
> But I'm pretty sure this would prevent that.
>
> Thoughts? Gotchas?
>
> Lelio
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst, Network Infrastructure
> Computing and Communications Services (CCS)
> University of Guelph
>
> 519-824-4120 Ext 56354
> le...@uoguelph.ca
> www.uoguelph.ca/ccs
> Room 037, Animal Science and Nutrition Building
> Guelph, Ontario, N1G 2W1
>
>
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Re: [cisco-voip] UCCX Custom Call Variables - Script to Script Transfers

2017-07-19 Thread Bill Talley
✋️

The critical piece to our clients would be the report format both in summary 
format and call-by-call detail. 

For example, a summary report would show how many callers pressed 1, 2, 3, etc 
during the selected reporting range, whereas call detail report would show menu 
selections at each menu during a specific call.  To me, that's a significant 
deficiency when using custom call variables.  

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

> On Jul 19, 2017, at 4:07 PM, Tanner Ezell  wrote:
> 
> Out of curiosity, who or their customers would be interested in a drop in 
> solution that would automatically perform menu option and flow reporting 
> without much if any modification to scripts?
> 
> Essentially you would drop in an application, turn it on, and it would begin 
> reporting on menu selections (valid and invalid) automatically without using 
> custom call variables? The only thing you'd probably want to change at the 
> script level would be to add some kind of identifier, i.e: "Customer Service 
> Main menu", for reporting purposes.
> 
>> On Wed, Jul 19, 2017 at 2:00 PM, Anthony Holloway 
>>  wrote:
>> There's advantages and disadvantages to both, for sure.  If I in fact did a 
>> call subflow, I feel confident I would not have this issue, but then I will 
>> lose reporting for calls which enter the second script.  I.e., Calls 
>> presented.  So, double edge sword here.  Thanks for the suggestion though.
>> 
>> 
>>> On Wed, Jul 19, 2017 at 9:45 AM Blakeman, Stephen  wrote:
>>> I remember having an issue with going between UCCX scripts and was advised 
>>> to not do the Call Redirect but instead use the Call Subflow step from TAC. 
>>> Not sure if that would help or not. Our issues were having to do with 
>>> trying to deliver calls to agents on the second(transferred) script. I 
>>> changed to the Call Subflow and the issues went away for me.
>>> 
>>>  
>>> 
>>> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
>>> Anthony Holloway
>>> Sent: Tuesday, July 18, 2017 5:13 PM
>>> To: Cisco VoIP Group 
>>> Subject: [cisco-voip] UCCX Custom Call Variables - Script to Script 
>>> Transfers
>>> 
>>>  
>>> 
>>> I'm working on a menu selection tracking solution, which survives across 
>>> transfers between department IVRs.  It's not going well.  In fact, I'm 99% 
>>> positive it's not possible, and I might have to look outside of UCCX to 
>>> solve this.  I.e., DB or WS integration
>>> 
>>>  
>>> 
>>> Scenario: Caller dials Main Line script, selects option 7 for Help Desk, 
>>> caller gets transferred via Call Redirect step to the Help Desk script, 
>>> selects option 2 for password reset, then speaks to an Agent.  Each script 
>>> is writing CCV1 with the menu selections made during the script execution.
>>> 
>>>  
>>> 
>>> I'm seeing in the callcontactdetail (CCD) table that the Session Seq Num 0 
>>> is written immediately after the transfer to the Help Desk, and includes 
>>> the CCV value of 7.
>>> 
>>>  
>>> 
>>> However, and here's the unexpected part, the Session Seq Num 1 is also 
>>> written to the CCD table at the same time, with the same CCV value: 7, 
>>> despite the call still being active on the second script.  Then when the 
>>> call is disconnected, the CCV does not change or get updated for the 
>>> Session Seq Num 1, which was already written to the CCD table.
>>> 
>>>  
>>> 
>>> Example CCD Record Set - This query was ran while the call was active on 
>>> the second leg (seq num 1) and before the call reached an Agent).
>>> 
>>>  
>>> 
>>> admin:run uccx sql db_cra select limit 2 startdatetime, sessionid, 
>>> sessionseqnum, applicationname, applicationtaskid, originatordn, 
>>> callednumber, customvariable6 from contactcalldetail where originatordn 
>>> like '%16125551212' order by startdatetime desc
>>> 
>>> STARTDATETIMESESSIONIDSESSIONSEQNUM  APPLICATIONNAME  
>>> APPLICATIONTASKID  ORIGINATORDN  CALLEDNUMBER  CUSTOMVARIABLE1
>>> 
>>> 
>>> 
>>> 2017-07-18 20:36:57.856  5326646  1  HelpDesk 
>>> 47000343967+16125551212  +16125551313  7,
>>> 
>>> 2017-07-18 20:36:36.872  5326646  0  MainLine 
>>> 47000343963+16125551212  +16125551313  7,
>>> 
>>>  
>>> 
>>> Recall that both records are written to the CCD table after the transfer 
>>> from the Main Line to the Help Desk, but while the call is still active on 
>>> the Help Desk script, and before the call is sent to the Agent.
>>> 
>>>  
>>> 
>>> Anyway, all of your thoughts, opinions, confirmations and disputes are all 
>>> welcomed.  Thanks.
>>> 
>> 
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Re: [cisco-voip] UCCX Custom Call Variables - Script to Script Transfers

2017-07-18 Thread Bill Talley
What if you write the menu options to a placeholder variable, passing the 
placeholder variable in session info and only set the CCV when the contact 
terminates?

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

> On Jul 18, 2017, at 4:13 PM, Anthony Holloway 
>  wrote:
> 
> I'm working on a menu selection tracking solution, which survives across 
> transfers between department IVRs.  It's not going well.  In fact, I'm 99% 
> positive it's not possible, and I might have to look outside of UCCX to solve 
> this.  I.e., DB or WS integration
> 
> Scenario: Caller dials Main Line script, selects option 7 for Help Desk, 
> caller gets transferred via Call Redirect step to the Help Desk script, 
> selects option 2 for password reset, then speaks to an Agent.  Each script is 
> writing CCV1 with the menu selections made during the script execution.
> 
> I'm seeing in the callcontactdetail (CCD) table that the Session Seq Num 0 is 
> written immediately after the transfer to the Help Desk, and includes the CCV 
> value of 7.
> 
> However, and here's the unexpected part, the Session Seq Num 1 is also 
> written to the CCD table at the same time, with the same CCV value: 7, 
> despite the call still being active on the second script.  Then when the call 
> is disconnected, the CCV does not change or get updated for the Session Seq 
> Num 1, which was already written to the CCD table.
> 
> Example CCD Record Set - This query was ran while the call was active on the 
> second leg (seq num 1) and before the call reached an Agent).
> 
> admin:run uccx sql db_cra select limit 2 startdatetime, sessionid, 
> sessionseqnum, applicationname, applicationtaskid, originatordn, 
> callednumber, customvariable6 from contactcalldetail where originatordn like 
> '%16125551212' order by startdatetime desc
> STARTDATETIMESESSIONIDSESSIONSEQNUM  APPLICATIONNAME  
> APPLICATIONTASKID  ORIGINATORDN  CALLEDNUMBER  CUSTOMVARIABLE1
> 
> 2017-07-18 20:36:57.856  5326646  1  HelpDesk 
> 47000343967+16125551212  +16125551313  7,
> 2017-07-18 20:36:36.872  5326646  0  MainLine 
> 47000343963+16125551212  +16125551313  7,
> 
> Recall that both records are written to the CCD table after the transfer from 
> the Main Line to the Help Desk, but while the call is still active on the 
> Help Desk script, and before the call is sent to the Agent.
> 
> Anyway, all of your thoughts, opinions, confirmations and disputes are all 
> welcomed.  Thanks.
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Re: [cisco-voip] UCCX script related questions

2017-06-19 Thread Bill Talley
No.

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

> On Jun 19, 2017, at 11:16 PM, naresh rathore <nare...@hotmail.com> wrote:
> 
> is it possible to use broadcast algorithm in uccx?
> 
> 
> 
> From: Bill Talley <btal...@gmail.com>
> Sent: Tuesday, June 20, 2017 6:05 AM
> To: naresh rathore
> Cc: Ben Amick; Terry Oakley; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] UCCX script related questions
>  
> One thing to keep in mind about forcing an agent back to ready, if there is 
> only one agent in a ready state and that agent misses a call, the system will 
> continuously send the call to that agent.  The caller will NOT necessarily 
> hear queue music/message oh hold and will never receiver any queue prompts 
> with options to escape the queue (like zero out, leave vm, etc).  If that 
> agent is away for 15 minutes, the caller will literally be ringing that one 
> agents phone for the entire 15 minutes (assuming no other agents become 
> available).
> 
>  Even if there are multiple agents in a ready state, if the call is routed to 
> one agent who's not available to answer, you're subjecting the caller to a 
> minimum 60s delay before the call will be routed to another available agent.
> 
> Supervisors have the ability to change agent states using their supervisor 
> login.
> 
> Sent from a mobile device with very tiny touchscreen input keys. Please 
> excude my typtos.
> 
> On Jun 19, 2017, at 7:45 PM, naresh rathore <nare...@hotmail.com> wrote:
> 
>> i tried to explain manager about benefit of agent being in not ready state 
>> after he doesnt pick the call. but he is keen on automatically changing the 
>> state from not ready to ready state. is there an option to revert back to 
>> ready state from not ready state.
>> 
>> 
>> do the supervisor have to login using his credentials on appadmin page to 
>> get rights to change skills?
>> 
>> From: Ben Amick <bam...@humanarc.com>
>> Sent: Monday, June 19, 2017 7:59 PM
>> To: Terry Oakley; 'naresh rathore'; cisco-voip@puck.nether.net
>> Subject: RE: UCCX script related questions
>>  
>> Regarding the last question: If a supervisor logs into the appadmin page, 
>> they will be given a dialogue that will enable them to change skills and 
>> skill levels.
>>  
>> As for skills vs skill based routing, etc. it’s a question of how you want 
>> to engineer it. Not ready on no-answer is only good if you have an 
>> environment where agents have to manage their status. Remember that when 
>> they are on a call they will be in the offhook or talking state, so you will 
>> not have to worry about them not answering calls, as they will not be 
>> presented.
>> As for the tiering with SBR, the question is how you want to prioritize 
>> calling. I’m assuming this is a helpdesk style structure, in which case I 
>> can understand segmenting apart calls.
>> However, I could see it done with multiple groups, with L3 engineers also 
>> being in the L1/2 queues but at a lower skill level and so on, but that’s 
>> just another option.
>> In general I would want L1 engineers to be given calls unless they’re 
>> completely unavailable and then given to L2 and then L3. You could 
>> accomplish this by checking qty of ready state in the queue as opposed to 
>> using timers.
>> This would also make it so that clients would not have to wait up to 180 sec 
>> before voicemail, and would be reactive to the number of ready agents, only 
>> escalating to l3 if all the L1/2 engineers are exhausted.
>> Also, managing ready state should be a function of HR/Supervisory, 
>> auto-setting back to ready would just cause it so that you’d have longer 
>> queue times.
>> You should be able to change the ring timer per user for timeout, but the 
>> longer you set it out the longer your queue times will be as it will be 
>> presented to less users. If you set it to 60sec, then in your design it will 
>> only ring 1 L1, 1 L2, and 1 L3 over a period of 180 sec.
>>  
>> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
>> Terry Oakley
>> Sent: Monday, June 19, 2017 10:40 AM
>> To: 'naresh rathore' <nare...@hotmail.com>; cisco-voip@puck.nether.net
>> Subject: Re: [cisco-voip] UCCX script related questions
>>  
>> I am not an expert in this at all but have some experience with a similar 
>> situation.I would suggest, and am glad to be corrected by those that 
>> have more expertise, that you use skills based routing.   The skills can be 
>> adjusted by the superviso

Re: [cisco-voip] UCCX script related questions

2017-06-19 Thread Bill Talley
One thing to keep in mind about forcing an agent back to ready, if there is 
only one agent in a ready state and that agent misses a call, the system will 
continuously send the call to that agent.  The caller will NOT necessarily hear 
queue music/message oh hold and will never receiver any queue prompts with 
options to escape the queue (like zero out, leave vm, etc).  If that agent is 
away for 15 minutes, the caller will literally be ringing that one agents phone 
for the entire 15 minutes (assuming no other agents become available).

 Even if there are multiple agents in a ready state, if the call is routed to 
one agent who's not available to answer, you're subjecting the caller to a 
minimum 60s delay before the call will be routed to another available agent.

Supervisors have the ability to change agent states using their supervisor 
login.

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

> On Jun 19, 2017, at 7:45 PM, naresh rathore  wrote:
> 
> i tried to explain manager about benefit of agent being in not ready state 
> after he doesnt pick the call. but he is keen on automatically changing the 
> state from not ready to ready state. is there an option to revert back to 
> ready state from not ready state.
> 
> 
> do the supervisor have to login using his credentials on appadmin page to get 
> rights to change skills?
> 
> From: Ben Amick 
> Sent: Monday, June 19, 2017 7:59 PM
> To: Terry Oakley; 'naresh rathore'; cisco-voip@puck.nether.net
> Subject: RE: UCCX script related questions
>  
> Regarding the last question: If a supervisor logs into the appadmin page, 
> they will be given a dialogue that will enable them to change skills and 
> skill levels.
>  
> As for skills vs skill based routing, etc. it’s a question of how you want to 
> engineer it. Not ready on no-answer is only good if you have an environment 
> where agents have to manage their status. Remember that when they are on a 
> call they will be in the offhook or talking state, so you will not have to 
> worry about them not answering calls, as they will not be presented.
> As for the tiering with SBR, the question is how you want to prioritize 
> calling. I’m assuming this is a helpdesk style structure, in which case I can 
> understand segmenting apart calls.
> However, I could see it done with multiple groups, with L3 engineers also 
> being in the L1/2 queues but at a lower skill level and so on, but that’s 
> just another option.
> In general I would want L1 engineers to be given calls unless they’re 
> completely unavailable and then given to L2 and then L3. You could accomplish 
> this by checking qty of ready state in the queue as opposed to using timers.
> This would also make it so that clients would not have to wait up to 180 sec 
> before voicemail, and would be reactive to the number of ready agents, only 
> escalating to l3 if all the L1/2 engineers are exhausted.
> Also, managing ready state should be a function of HR/Supervisory, 
> auto-setting back to ready would just cause it so that you’d have longer 
> queue times.
> You should be able to change the ring timer per user for timeout, but the 
> longer you set it out the longer your queue times will be as it will be 
> presented to less users. If you set it to 60sec, then in your design it will 
> only ring 1 L1, 1 L2, and 1 L3 over a period of 180 sec.
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> Terry Oakley
> Sent: Monday, June 19, 2017 10:40 AM
> To: 'naresh rathore' ; cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] UCCX script related questions
>  
> I am not an expert in this at all but have some experience with a similar 
> situation.I would suggest, and am glad to be corrected by those that have 
> more expertise, that you use skills based routing.   The skills can be 
> adjusted by the supervisor on the fly so that leaves it to the supervisor to 
> have the right staff (engineer) in place to respond to the call.As for 
> extending the timeout to being set to Not Ready I believe that can be 
> adjusted on the Application Management section under agentTimeout.   Again 
> very happy to be corrected as I am still working through scripting and how to 
> make it the most efficient.
>  
> Terry
>  
>  
> Terry Oakley
> Telecommunications Coordinator | Information Technology Services
> Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5
> work (403) 342-3521   |  FAX (403) 343-4034
>  
>  
>  
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
> naresh rathore
> Sent: June 19, 2017 1:48 AM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] UCCX script related questions
>  
> hi All,
>  
>  
> i have to configure CRS scripting on UCCX. following is the requirement.
> broadcast ring to level 1 engineer for 60 sec, then call goes to queue and 
> 

Re: [cisco-voip] best CUIC report

2017-04-29 Thread Bill Talley
P.S. Won't show queue stats, but will show trigger stats.  You likely won't get 
queue stats unless you're not checking agent availability or time of day until 
after the caller is queued 

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

> On Apr 29, 2017, at 8:20 PM, naresh rathore  wrote:
> 
> hi,
> 
> 
> 
> anyone?.  is there a way to get inbound call statistics when its out of 
> office hours and i want to search based on trigger and queue based out of 
> office hours ( from 6pm to 8 am next day).
> 
> 
> 
> Regards
> 
> 
> 
> Naresh Rathore
> 
> 
> 
> From: cisco-voip  on behalf of naresh 
> rathore 
> Sent: Friday, April 28, 2017 6:20 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] best CUIC report
>  
> hi all,
> 
> 
> 
> i have a uccx 11.x environment. one of our user/receptionist want a report in 
> which she can check the number of calls hit to IVR during out of office hours 
> (i.e from 5:00pm to next day 8:00 am, and when no agent is logged in). Could 
> you guys guide me which will give me this information.
> 
> 
> 
> Regards
> 
> 
> 
> Naray
> 
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Re: [cisco-voip] best CUIC report

2017-04-29 Thread Bill Talley
Application Summary Report under Inbound >> System, I believe.

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

> On Apr 29, 2017, at 8:20 PM, naresh rathore  wrote:
> 
> hi,
> 
> 
> 
> anyone?.  is there a way to get inbound call statistics when its out of 
> office hours and i want to search based on trigger and queue based out of 
> office hours ( from 6pm to 8 am next day).
> 
> 
> 
> Regards
> 
> 
> 
> Naresh Rathore
> 
> 
> 
> From: cisco-voip  on behalf of naresh 
> rathore 
> Sent: Friday, April 28, 2017 6:20 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] best CUIC report
>  
> hi all,
> 
> 
> 
> i have a uccx 11.x environment. one of our user/receptionist want a report in 
> which she can check the number of calls hit to IVR during out of office hours 
> (i.e from 5:00pm to next day 8:00 am, and when no agent is logged in). Could 
> you guys guide me which will give me this information.
> 
> 
> 
> Regards
> 
> 
> 
> Naray
> 
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Re: [cisco-voip] Jabber voicemail issue only on IPhone

2017-03-26 Thread Bill Talley
Any chance your iPhone is actually using wifi assist instead of wireless?

You're going through Expressway when using you're iPhone on mobile data?

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

> On Mar 23, 2017, at 1:39 AM, naresh rathore  wrote:
> 
> hi All,
> 
> 
> 
> when i login jabber from my desktop/laptop, i can access voicemail without 
> any issue. but if i login via IPhone the voicemail service keep reconnecting 
> and gives the following error.
> 
> 
> 
> "cannot connect to voicemail server. check your network settings. Reconnect"
> 
> 
> 
> I face this issue when i connect either internally (wifi) or via mobile data. 
> please suggest what could be the cause of this issue. 
> 
> 
> CUCM: 11.5.1.11900-26
> 
> 
> 
> 
> 
> Regards
> 
> 
> 
> 
> 
> Naray
> 
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Re: [cisco-voip] Uccx finesse 10.6 layout and custom variables for teams

2017-02-16 Thread Bill Talley
Are you certain about that?   I thought there was something under the Team
Resource tab in Finesse Administration where you could override the default
desktop layout by team?  Could have sworn I've done that once or twice, but
maybe I'm mistaken... I have been before, lol.

On Thu, Feb 16, 2017 at 8:42 AM, ccie collan 
wrote:

> Hello everyone ,
>
> It looks like Uccx 10.6 does not support different layout for each team .
> Does anyone have any idea how to manage all variables as not all of my
> teams have same variable information .
>
> Uccx 11 offers it but upgrading this year is not an option for us .
>
> Any input you all can provide is appreciated , thank you for the help .
>
>
>
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>
>
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Re: [cisco-voip] Voice Dialing Misdirects

2016-06-06 Thread Bill Talley
Isn't there a setting to play all matches? I don't recall for sure.  At least 
the caller could pick on or the other.  

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

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

2016-06-06 Thread Bill Talley
I believe this is probably a speech enabled directory handler.

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

> On Jun 6, 2016, at 1:13 PM, David Zhars  wrote:
> 
> We have a greeting when people call the main number, and if you press 9, you 
> can speak the name of the person you want to connect to. 
> 
> This oddity is happening:  
> 
> I speak "John Smith" who is extension 1305, and the system waits, then says 
> "transferring to Jane Doe" who is extension 1306.
> 
> I can't seem to figure out where this is off...
> Unity and UCM 8
> 
> Thanks
> 
> Dave
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Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting

2016-04-08 Thread Bill Talley
Closing the loop here, this was my issue. I was able to execute a stored
procedure which wasn't creating a temp table.  Thanks again for sharing Ed.

On Fri, Apr 8, 2016 at 7:27 AM, Ed Leatherman <ealeather...@gmail.com>
wrote:

> Bill,
>
> Does your SP use temporary tables? look at CSCul06940 - I dont know that
> its resolved in 10.X yet.
>
> Ed
>
>
>
> On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley <btal...@gmail.com> wrote:
>
>> Hi all,
>>
>> Has anyone imported custom stored procedures into the db_cra for use in
>> CUIC?  I'm having an issue getting a new stored proc loaded using either
>> RazorSQL or AGS Server Studio.  From what I've read, I need to execute the
>> stored proc; however, when I do that I get an error that uccxhruser doesn't
>> have permission to execute the stored proc.  It seems I can't apply
>> permissions to the uccxhruser account on the stored proc until the stored
>> proc is loaded.   Would anyone have any tips on how I can go about getting
>> the stored proc loaded to the database?
>>
>> Hopefully this makes sense to someone ;-)
>>
>> TIA,
>> Bill
>>
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
> Ed Leatherman
>
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Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting

2016-04-08 Thread Bill Talley
Cool, this includes everything I had been doing as far as I can tell.  Thanks a 
lot for sharing these details Ed!

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

> On Apr 8, 2016, at 7:29 AM, Ed Leatherman <ealeather...@gmail.com> wrote:
> 
> I haven't touched this in 2 years - here's notes I took when I was playing 
> around with this before in case it will help:
> http://www.evernote.com/l/AAVCJ8P2UbJG5o0yqjxIO6-9p7edydMHCfE/
> 
>> On Fri, Apr 8, 2016 at 8:27 AM, Ed Leatherman <ealeather...@gmail.com> wrote:
>> Bill,
>>  
>> Does your SP use temporary tables? look at CSCul06940 - I dont know that its 
>> resolved in 10.X yet.
>> 
>> Ed
>> 
>> 
>> 
>>> On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley <btal...@gmail.com> wrote:
>>> Hi all,
>>> 
>>> Has anyone imported custom stored procedures into the db_cra for use in 
>>> CUIC?  I'm having an issue getting a new stored proc loaded using either 
>>> RazorSQL or AGS Server Studio.  From what I've read, I need to execute the 
>>> stored proc; however, when I do that I get an error that uccxhruser doesn't 
>>> have permission to execute the stored proc.  It seems I can't apply 
>>> permissions to the uccxhruser account on the stored proc until the stored 
>>> proc is loaded.   Would anyone have any tips on how I can go about getting 
>>> the stored proc loaded to the database?
>>> 
>>> Hopefully this makes sense to someone ;-)
>>> 
>>> TIA,
>>> Bill
>>> 
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
>> 
>> 
>> -- 
>> Ed Leatherman
> 
> 
> 
> -- 
> Ed Leatherman
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Re: [cisco-voip] UCCx 10.5 custom stored procedure for reporting

2016-04-08 Thread Bill Talley
It does.  I'll take a look at it.  Thanks!

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

> On Apr 8, 2016, at 7:27 AM, Ed Leatherman <ealeather...@gmail.com> wrote:
> 
> Bill,
>  
> Does your SP use temporary tables? look at CSCul06940 - I dont know that its 
> resolved in 10.X yet.
> 
> Ed
> 
> 
> 
>> On Thu, Apr 7, 2016 at 6:06 PM, Bill Talley <btal...@gmail.com> wrote:
>> Hi all,
>> 
>> Has anyone imported custom stored procedures into the db_cra for use in 
>> CUIC?  I'm having an issue getting a new stored proc loaded using either 
>> RazorSQL or AGS Server Studio.  From what I've read, I need to execute the 
>> stored proc; however, when I do that I get an error that uccxhruser doesn't 
>> have permission to execute the stored proc.  It seems I can't apply 
>> permissions to the uccxhruser account on the stored proc until the stored 
>> proc is loaded.   Would anyone have any tips on how I can go about getting 
>> the stored proc loaded to the database?
>> 
>> Hopefully this makes sense to someone ;-)
>> 
>> TIA,
>> Bill
>> 
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> -- 
> Ed Leatherman
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[cisco-voip] UCCx 10.5 custom stored procedure for reporting

2016-04-07 Thread Bill Talley
Hi all,

Has anyone imported custom stored procedures into the db_cra for use in
CUIC?  I'm having an issue getting a new stored proc loaded using either
RazorSQL or AGS Server Studio.  From what I've read, I need to execute the
stored proc; however, when I do that I get an error that uccxhruser doesn't
have permission to execute the stored proc.  It seems I can't apply
permissions to the uccxhruser account on the stored proc until the stored
proc is loaded.   Would anyone have any tips on how I can go about getting
the stored proc loaded to the database?

Hopefully this makes sense to someone ;-)

TIA,
Bill
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Re: [cisco-voip] Need to modify P-Asserted Identity in SIP message for call to work...

2016-03-19 Thread Bill Talley
Outbound calls probably go out with your CUCM domain name (or IP address) and 
not VoIP.centurylink.com.  You would want to match that in your SIP profile.

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

> On Mar 16, 2016, at 12:12 AM, Jonathan Charles  wrote:
> 
> So, we have  a Century Link SIP trunk and we need to send a specific 
> P-Asserted Identity for the call to work.
> 
> We have: a 
> 
> voice class sip-profiles 101
> 
>  request INVITE sip-header P-Asserted-Identity modify 
> "P-Asserted-Identity:(.*)@voip.centurylink.com" "P-Asserted-Identity: 
> " 
> !
> 
> TO change the P-Asserted ID to what CL wants... however it is not making the 
> change...
> 
> 
> dial-peer voice 200 voip
>  voice-class sip asserted-id pai
>  voice-class sip profiles 101
> 
> What do we need to do to force a specific P-Asserted Identity...?
> 
> 
> TIA!
> 
> 
> 
> Jonathan
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Re: [cisco-voip] Need to modify P-Asserted Identity in SIP message for call to work...

2016-03-19 Thread Bill Talley
You're certain the PAI from CUCM to CUBE matches the pattern in your profile?

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

> On Mar 16, 2016, at 2:13 PM, Jonathan Charles  wrote:
> 
> Yeah, did that too.. and still seeing the unmodified P-Asserted Identity...
> 
>> On Wed, Mar 16, 2016 at 11:27 AM, Anthony Holloway 
>>  wrote:
>> If you apply it under voice service voip > sip, then it's applied to all 
>> outgoing messages, regardless of dial-peer.  Specifying it on the dial-peer 
>> level is redundant at that point, though it does take precedence over the 
>> global configuration.
>> 
>> You can debug ccsip mess and grab an outgoing INVITE example and paste it 
>> into the tool I linked before.
>> 
>>> On Wed, Mar 16, 2016 at 9:32 AM, Jonathan Charles  wrote:
>>> Thanks, I checked and the voice class sip profile is on every dial peer not 
>>> pointing at CCM, it is also applied under voice service voip - sip... 
>>> 
>>> I will run the debug after hours to see why it is not being applied...
>>> 
>>> 
>>> 
>>> Jonathan
>>> 
 On Wed, Mar 16, 2016 at 12:47 AM, Anthony Holloway 
  wrote:
 Hey Jonathan,
 
 Make sure that the profile is on the correct outgoing dial-peer.  SIP 
 profiles are only executed in the outgoing dial-peer, after the dial-peer 
 has been matched/selected.
 
 You can use debug voip ccapi inout or debug voip dialpeer to find out 
 which outgoing dial-peer is being matched.  Also, you can use debug ccsip 
 all to see if and what the SIP profile is doing.
 
 Assuming dial-peer 200 is your guys, then you can use this tool to help 
 you design your profile:
 
 http://www.cisco.com/web/tsweb/tools/sip-profile/index.html
 
 I pasted your profile into the tool and gave it a sample input (it's 
 brief, I know) and here is the result:
 
 
 
 You should note that the red indicates a match and a replace happened, but 
 also note that the replaced pattern contains an extra > at the end.  The 
 reason is because you told it the match ended with .com and not .com>.  
 The reason the first < was replace appropriately, is because your (.*) 
 accounts for it, as well as the space before it.
 
 I made a small adjustment to your profile, mainly to shorten it up, but to 
 also cleanup the extra > at the end
 
 
 
 I've never tried this next method before, but it seems a little cleaner 
 (less regex) than the above two modify profiles, albeit it takes two steps 
 to execute:
 
 
 
 That last option may not actually work in a real IOS device, as I'm not 
 sure if the message is re-read from the top after a rule matches.
 
 I hope that helps.
 
> On Wed, Mar 16, 2016 at 12:12 AM, Jonathan Charles  
> wrote:
> So, we have  a Century Link SIP trunk and we need to send a specific 
> P-Asserted Identity for the call to work.
> 
> We have: a 
> 
> voice class sip-profiles 101
> 
>  request INVITE sip-header P-Asserted-Identity modify 
> "P-Asserted-Identity:(.*)@voip.centurylink.com" "P-Asserted-Identity: 
> " 
> !
> 
> TO change the P-Asserted ID to what CL wants... however it is not making 
> the change...
> 
> 
> dial-peer voice 200 voip
>  voice-class sip asserted-id pai
>  voice-class sip profiles 101
> 
> What do we need to do to force a specific P-Asserted Identity...?
> 
> 
> TIA!
> 
> 
> 
> Jonathan
> 
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[cisco-voip] AT or Windstream issues mid-CONUS?

2016-02-18 Thread Bill Talley
Hi all,

We have a customer who thought they were suffering system issues, but we've 
since discovered there seem to be handoff issues specifically from AT 
(callers) to Windstream (customer's telco) as those calls never hit the 
customers PRIs.  Is anyone else experiencing or has anyone heard about similar 
issues elsewhere this morning?

TIA
Bill

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excude my typtos.
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Re: [cisco-voip] UCCX 9 prompts gibberish

2016-02-16 Thread Bill Talley
It would be even nicer if CCX developers added support for both, say like CCM 
or CUC.  If we need to switch codecs mid-call that's one thing but if one call 
comes across the WAN to an agent, and another call comes in locally to CCX, CCX 
is usually the only component in those media paths which doesn't support both 
g711 and g729 in those scenarios.  

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

> On Feb 16, 2016, at 11:18 AM, Anthony Holloway 
>  wrote:
> 
> I'm surprised no one has mentioned just using UCCX itself to record the audio.
> 
> See this link for the bare minimum steps required to record a prompt in UCCX, 
> no matter the audio codec.
> 
> https://supportforums.cisco.com/comment/7224091#comment-7224091
> 
> Also, and may I just state for the record that UCCX in G729 is more hassel 
> than it's worth.  Just account for G711 in your BW or use transcoders.
> 
>> On Tue, Feb 16, 2016 at 11:09 AM, Brian Meade  wrote:
>> According to the admin guide, Cisco has a white paper on recording prompts 
>> as G.729:
>> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_10_5/config/guides/UCCX_BK_U9F91527_00_unified-ccx-administration-guide-release-10-5/UCCX_BK_U9F91527_00_unified-ccx-administration-guide-release_chapter_01001.html#UCCX_TK_R3A08A4D_00
>> 
>> You just need to email to apps-supp...@cisco.com to obtain a copy.  It might 
>> have some other ideas not mentioned here.
>> 
>>> On Tue, Feb 16, 2016 at 11:51 AM, abbas Wali  wrote:
>>> have got that in window xp sound recorder. but again all this is giving 
>>> back G711.
>>> 
>>> I think the last resort is upload the file as MoH file and then dloading 
>>> again to get G729 :((
>>> pain..
>>> 
 On 16 February 2016 at 16:42, Andreas Sikkema  wrote:
 James,
 
 > No, just for G711. I don't have a solution for G729. Hopefully someone 
 > else
 > does.
 
 G.729 is patent encumbered, so no "free" (beer or otherwise) solutions 
 there.
 
 Also, CCIT U-Law is G.711 u-law is PCM u-law is PCMU. There must be
 one of these in Audacity.
 
 --
 Andreas Sikkema
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>>> 
>>> 
>>> -- 
>>> Abbas Wali
>>> 
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>> 
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[cisco-voip] Webex for Mac users - keystroke passthrough

2016-01-11 Thread Bill Talley
Hi all,

If you're not a Mac user, you can probably ignore this.

Do many of you use Webex Meeting Center on Mac for remote support?  We have
quite a few Mac users who are frustrated by the lack of support for passing
through the ctrl-alt keystroke needed to escape from the virtual machine
console in vSphere on the remote machine being controlled.  I believe
there's not currently a workaround for this, so I've opened a new feature
request on it with Webex.

The new feature request is to add either support of passing the ctrl-alt
keystroke, or adding keystroke mapping/macros to Meeting Center for Mac, as
this works on Windows without fail.  If you would like to see this feature
request implemented, would any of you mind reaching out to Webex to request
implementation of the feature referenced in Case #HD0008814712.

You can reach Webex Support at:

- @Webex on Twitter
-  Chat online with support via
https://cisco-support.webex.com/customers/apex/WebEx_CaseSystemDown
-  Call 866-229-3239.

Thanks
Bill
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Re: [cisco-voip] Agents Selectively Answer Queues

2015-09-03 Thread Bill Talley
Add the escalation queue and create a script parameter or XML file that is 
evaluated to determine if the escalation queue is active or not and escalate 
the call accordingly.  The manager can activate it or deactivate it through an 
IVR app or ip phone service?

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

> On Sep 3, 2015, at 12:25 PM, Matthew Loraditch 
>  wrote:
> 
> We only want them to get Tier 1 calls when the managers decide for them to.
>  
> Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
> Network Engineer
> Direct Voice: 443.541.1518
> 
> Facebook | Twitter | LinkedIn | G+
>  
> From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade
> Sent: Thursday, September 3, 2015 1:20 PM
> To: Matthew Loraditch 
> Cc: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Agents Selectively Answer Queues
>  
> Can you just do Most Skilled routing and give the Tier2 guys a lower Tier1 
> skill?  That will allow the calls to roll over automatically if everyone in 
> Tier1 is not ready.
>  
> On Thu, Sep 3, 2015 at 1:05 PM, Matthew Loraditch 
>  wrote:
> We have tier 2 agents. Generally we want them to only answer their calls, but 
> if Tier 1 becomes overwhelmed we need them to answer tier 1 calls. I’ve never 
> had this scenario before, but my only thought so far was secondary logins 
> that will have both Tier 1 and Tier 2 skills. We do not want to queue the 
> calls to another queue when thresholds are met nor do we want managers 
> hot-adding skills.
>  
> Am I missing any options here?
>  
> Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
> Network Engineer
> Direct Voice: 443.541.1518
> 
> Facebook | Twitter | LinkedIn | G+
>  
> 
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Re: [cisco-voip] MediaSense Configuration

2015-02-25 Thread Bill Talley
doh...

Make you set the source to phone preferred AND NOT gateway preferred
(assuming you are using the BIB).

On Wed, Feb 25, 2015 at 2:44 PM, Bill Talley btal...@gmail.com wrote:

 Correct on licensing.  Just went through this myself and it was really
 easy to setup to record forked audio.  The only issue we ran into relates
 to securing of the recording files.  With stand alone MediaSense, users who
 need access to the recordings are added as API users in MS and have access
 to ALL recordings, not just the recordings they are a party in.

 In CUCM you'll create a SIP trunk to MediaSense, create route pattern
 pointing to the trunk, create a recording profile with a DN to match the
 route pattern.  Then enable the BIB on the phone device and under the DN
 set the recording profile, recordig option and recording source on the
 phone.  Make you set the source to phone preferred gateway preferred
 (assuming you are using the BIB).

 On Wed, Feb 25, 2015 at 2:22 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  Trying to setup a PoC for media sense internally to validate I can
 deploy at a client for recording purposes… from what I can tell licensing
 is honor based?

 Secondly from what I am reading this section of the documentation is all
 I have to do:
 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/mediasense/105_SU1/User_Guide/CUMS_BK_M5B01864_00_ms-user-guide-105-SU1/CUMS_BK_M5B01864_00_ms-user-guide-105_chapter_010.html#CUMS_TP_M28EF01E_00

 Fork calls into media sense? The default rule in MS says record all audio



 Is there anything else I need to do? Trying now to figure out what logs
 to look at to see if it’s actually seeing anything or not, because nothing
 shows up on the system.





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

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Re: [cisco-voip] MediaSense Configuration

2015-02-25 Thread Bill Talley
Search and Play.  it's browser based and allows you to playback or
download the audio file as .wav or mp4.


​

On Wed, Feb 25, 2015 at 4:29 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 Does MediaSense have it's own playback UI now?  I thought I heard that it
 did.


 On Wed, Feb 25, 2015 at 2:50 PM Bill Talley btal...@gmail.com wrote:

 Correct on licensing.  Just went through this myself and it was really
 easy to setup to record forked audio.  The only issue we ran into relates
 to securing of the recording files.  With stand alone MediaSense, users who
 need access to the recordings are added as API users in MS and have access
 to ALL recordings, not just the recordings they are a party in.

 In CUCM you'll create a SIP trunk to MediaSense, create route pattern
 pointing to the trunk, create a recording profile with a DN to match the
 route pattern.  Then enable the BIB on the phone device and under the DN
 set the recording profile, recordig option and recording source on the
 phone.  Make you set the source to phone preferred gateway preferred
 (assuming you are using the BIB).

 On Wed, Feb 25, 2015 at 2:22 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  Trying to setup a PoC for media sense internally to validate I can
 deploy at a client for recording purposes… from what I can tell licensing
 is honor based?

 Secondly from what I am reading this section of the documentation is all
 I have to do:
 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/mediasense/105_SU1/User_Guide/CUMS_BK_M5B01864_00_ms-user-guide-105-SU1/CUMS_BK_M5B01864_00_ms-user-guide-105_chapter_010.html#CUMS_TP_M28EF01E_00

 Fork calls into media sense? The default rule in MS says record all audio



 Is there anything else I need to do? Trying now to figure out what logs
 to look at to see if it’s actually seeing anything or not, because nothing
 shows up on the system.





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

  Facebook https://www.facebook.com/heliontech?ref=hl | Twitter
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Re: [cisco-voip] MediaSense Configuration

2015-02-25 Thread Bill Talley
It should answer.

On Wed, Feb 25, 2015 at 5:15 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  It is forced on, on my test phone, albeit we have it on system wide as
 well. I've just been trying to eliminate things.

 Should the route pattern I have for the mediasense server ring busy if I
 dial it? I have another recording solution where the number answers in a
 similar setup.



 Matthew G. Loraditch - CCNP-V, CCNA-RS, CCDA
 Network Engineer


  We *understand *and *solve *your

 technology challenges so

 you can sell and service cars.


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 G+ https://plus.google.com/+Heliontechnologies/posts
--
 *From:* Chris Ward (chrward) [chrw...@cisco.com]
 *Sent:* Wednesday, February 25, 2015 5:48 PM
 *To:* Matthew Loraditch; Bill Talley
 *Cc:* cisco-voip@puck.nether.net
 *Subject:* RE: [cisco-voip] MediaSense Configuration

   Also, make sure the BiB on the phones are enabled. Its disabled by
 default. You need to do it per device or in the CCM service parameter.



 +Chris

 TME - MediaSense and Unity Connection



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Matthew Loraditch
 *Sent:* Wednesday, February 25, 2015 3:54 PM
 *To:* Bill Talley
 *Cc:* cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] MediaSense Configuration



 Ok that’s what I did and what I thought. Time to dig!



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

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



 *From:* Bill Talley [mailto:btal...@gmail.com btal...@gmail.com]
 *Sent:* Wednesday, February 25, 2015 3:46 PM
 *To:* Matthew Loraditch
 *Cc:* cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] MediaSense Configuration



 doh...

 Make you set the source to phone preferred AND NOT gateway preferred
 (assuming you are using the BIB).



 On Wed, Feb 25, 2015 at 2:44 PM, Bill Talley btal...@gmail.com wrote:

  Correct on licensing.  Just went through this myself and it was really
 easy to setup to record forked audio.  The only issue we ran into relates
 to securing of the recording files.  With stand alone MediaSense, users who
 need access to the recordings are added as API users in MS and have access
 to ALL recordings, not just the recordings they are a party in.

 In CUCM you'll create a SIP trunk to MediaSense, create route pattern
 pointing to the trunk, create a recording profile with a DN to match the
 route pattern.  Then enable the BIB on the phone device and under the DN
 set the recording profile, recordig option and recording source on the
 phone.  Make you set the source to phone preferred gateway preferred
 (assuming you are using the BIB).



 On Wed, Feb 25, 2015 at 2:22 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

   Trying to setup a PoC for media sense internally to validate I can
 deploy at a client for recording purposes… from what I can tell licensing
 is honor based?

 Secondly from what I am reading this section of the documentation is all I
 have to do:
 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/mediasense/105_SU1/User_Guide/CUMS_BK_M5B01864_00_ms-user-guide-105-SU1/CUMS_BK_M5B01864_00_ms-user-guide-105_chapter_010.html#CUMS_TP_M28EF01E_00

 Fork calls into media sense? The default rule in MS says record all audio



 Is there anything else I need to do? Trying now to figure out what logs to
 look at to see if it’s actually seeing anything or not, because nothing
 shows up on the system.





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

 Facebook https://www.facebook.com/heliontech?ref=hl | Twitter
 https://twitter.com/HelionTech | LinkedIn
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Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...

2015-01-29 Thread Bill Talley
Check the idle URL on the device, then check the auto registration template 
under User Management  User/Phone Add  universal device template.  The idle 
URL is populated under the Service Configuratuon Settings for the default 
template.

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

 On Jan 29, 2015, at 10:52 AM, Ryan Huff ryanh...@outlook.com wrote:
 
 CUCM 10.5 (New Build with DNS)
 
 Testing out phone registration with IP Communicator 8.6.4. Phone auto 
 registers just fine and is assigned all the auto registered defaults I have 
 specified (DN, partition, CSS ... etc) and the IP communicator is able to 
 resolve all the URLs by DNS. Everything appears to be fine.
 
 However, the Services menu keeps poping up. I haven't configured any services 
 beyond the shipped services so the service menu is blank. I hit the exit 
 button on the services menu and it goes away for a second or two, then it 
 comes right back.
 
 The phone can still process digits, I can dial, I can get dialtone ... etc 
  everything seems to work but the services menu. I feel like I have dealt 
 with this before but I can't recall what the solution was.
 
 Thanks,
 
 Ryan
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Re: [cisco-voip] Licenses upgrade issues

2015-01-08 Thread Bill Talley
You guys just don't know what you're doing.  Jeesh ;-)

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

 On Jan 8, 2015, at 6:08 PM, Mike mik...@msn.com wrote:
 
 Actually I beg to differ. It’s a system that needs to be improved upon.
  
 From: Josh Warcop [mailto:j...@warcop.com] 
 Sent: Thursday, January 08, 2015 7:01 PM
 To: Mike ; 'Dana Tong'; 'Erick Wellnitz'; 'Charles Goldsmith'
 Cc: cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] Licenses upgrade issues
  
 It's not a nightmare. It's a consequence of losing important information.
 From: Mike 
 Sent: ‎1/‎8/‎2015 6:52 PM
 To: 'Dana Tong'; 'Josh Warcop'; 'Erick Wellnitz'; 'Charles Goldsmith'
 Cc: cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] Licenses upgrade issues
 
 Dana,
  
 I’ve been down this road before. You need to do the following:
  
 1)  Have your Cisco account team open a PSTS case
 2)  Find out if the ESW contract changed and became unattached from UCSS
 3)  SO# this one is huge..sorry to say it’s your golden ticket for 
 entitlements you need to find it
 4)  Find all the emails from your previous upgrades you would have needed 
 to prove entitlements to get those so finding that will def help
 5)  Cross your fingers
  
 Welcome to the Cisco licensing nightmare J
  
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
 Dana Tong
 Sent: Thursday, January 08, 2015 6:10 PM
 To: Josh Warcop; Erick Wellnitz; Charles Goldsmith
 Cc: cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Licenses upgrade issues
  
 Hmmm. Kind of hard. They went from 4.1(3) on MCS hardware to 6.0 - 6.1(3)
  
 Then purchased CUWL and went to 7.x
  
 And migrated to 8.6(2) on UCS B series
  
 And finally upgraded to 10.x.
  
 Plus moved offices 4 times in the process. Will see if they have the original 
 PAKs.
  
 From: Josh Warcop [mailto:j...@warcop.com] 
 Sent: Friday, 9 January 2015 8:35 AM
 To: Dana Tong; Erick Wellnitz; Charles Goldsmith
 Cc: cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] Licenses upgrade issues
  
 You're not going to get far until you can prove what was paid for. They are 
 not being unhelpful intentionally as their systems can only go back so far. 
 Being that old those records are no longer online and have been archived.
 
 If the client has lost everything including PAK keys you're going to have to 
 get with the Cisco AM or that original partner to dig up entitlements. Hint: 
 PAK keys have the Cisco SO written on them typically in the bottom left 
 corner. 
 
 All of this should've been done prior to the upgrade using the LCU utility 
 and case opened with Cisco to check entitlement.
 From: Dana Tong
 Sent: ‎1/‎8/‎2015 5:19 PM
 To: Erick Wellnitz; Charles Goldsmith
 Cc: cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Licenses upgrade issues
 
 Hi Erick,
  
 They did all of the LCU Tool stuff prior to upgrade. All of the contracts 
 were active at time of upgrade.
  
 I still have the common issue of only obtaining licenses that the new PLM 
 reports and not what the customer has paid for.
  
  
 The original purchase (SO#) was some 8 years ago with another partner.
  
 Cheers
 Dana
  
  
 From: Erick Wellnitz [mailto:ewellnitzv...@gmail.com] 
 Sent: Friday, 9 January 2015 8:04 AM
 To: Charles Goldsmith
 Cc: Dana Tong; cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Licenses upgrade issues
  
 That's what we did to make sure everything was good with their contracts 
 contained the right licenses.  Then go through the pre-upgrade process to 
 verify then the post upgrade goes pretty fast.
  
 5 calendar days, three of which were weekend and holiday, for my last upgrade.
  
 On Thu, Jan 8, 2015 at 2:51 PM, Charles Goldsmith wo...@justfamily.org 
 wrote:
 Get the customer's Cisco AM involved is my advice.
  
 On Thu, Jan 8, 2015 at 2:47 PM, Dana Tong dana_t...@bridgepoint.com.au 
 wrote:
 Guys,
  
 A customer of mine did an upgrade last August and still has not been able to 
 obtain the licenses that they are entitled to. 
  
 The customer first bought CUWL licenses some 8 years ago and have had an 
 active and valid UCSS and ESW contract for this entire time. 
  
 Global Licensing provided the customer with a mix of UCL Enhanced and 
 Enhanced Plus licenses at the time of upgrade to get the system operational. 
 (They didn't have their Owner ID's set and it was an 8-10 upgrade.) They've 
 been battling with GLO ever since to get their CUWL licenses. 
  
 In the 5mths since the upgrade the customers UCSS went overdue but they have 
 since renewed with SWSS. GLO still won't provide the licenses. 
  
 Does anyone have any suggestions on how to get some traction here. 
  
  
 Regards,
  
 Dana Tong
  
 
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