Re: [cisco-voip] Question about detecting end-point analog telephony device present on ATA

[Brian Meade]

I'm not aware of anything like this for the ATAs.  We standardize a lot of
our large Enterprise customers on the VG Series for remote locations just
for things like this.  It's way more costly than ATAs though unfortunately.

On Thu, Jun 1, 2023, 11:26 AM Tim Reimers  wrote:

> Hi all -
> Thanks for the earlier assistance with obtaining details for ATAs.
> Turned out the only way to do it is a screenscrape and check each one for
> the DN.
>
> My next goal is:
> How to know if there's a device of any kind attached to the port of an
> ATA-186.
>
> I know for my VG-224 devices, I can do this:
> VG224#test voice port 2/0 line-test phone-detection
> measured result 0x20C48FB
> AC current = 0421 uA
> port 2/0 has phone connected
>
> Does anyone know if there's anything similar in an ATA, like somewhere in
> the web interface
> where a similar statistic is reported or a test available?
>
> We've got a lot of these that are remote to us, and I need to figure out
> which
> ones still have anything attached, and which ones are simply dead wire
> hanging off and the equipment
> may have been removed or powered off long ago.
> (yes, people don't tell us when they stop using something, or an outside
> vendor switches things around without IT knowing it.
> I bet we all have that T-shirt..)
>
> Thanks Tim
>
>
>
> --
>
> *Quis custodiet
> ipsos*
> * nexus*
>
> Tim Reimers
>
> Network Administrator
>
> I.T Services
>
> City of Asheville
>
> treim...@ashevillenc.gov
>
> (desk) 828-259-5512
>
> (cell)   828-552-1585
>
> "That’s no ordinary rabbit  packet! That’s the most foul, cruel, and bad
> tempered badly framed packet you ever set eyes on. Listen, that packet’s
> got a vicious streak a mile wide, he’s a killer.He’s got huge sharp
> MTU…eh, he can leap about and cross Vlans…. I warned you, I warned you
> but did you listen? No… ohhh no, it’s just a harmless little packet on
> the network, isn’t it now"
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Re: [cisco-voip] Question about RISDB queries and ATA devices?

[Brian Meade]

This is the way to do it. I add filters all the time to add additional
columns under the Device->Phone page.


On Thu, May 25, 2023, 2:37 PM  wrote:

> Just add the filter on the device/phone page to include directory number
> and you’ll be good to go.
>
>
>
> *From:* Tim Reimers 
> *Sent:* Thursday, May 25, 2023 12:27 PM
> *To:* natec...@gmail.com
> *Cc:* Ryan Ratliff (rratliff) ; Wes Sisk (wsisk) <
> ws...@cisco.com>; cisco-voip 
> *Subject:* Re: [cisco-voip] Question about RISDB queries and ATA devices?
>
>
>
> Sorry, sent before being ready...
>
>
>
> What I was hoping for was a list of all the registered active ATAs that
> would INCLUDE the DN on them.
>
> As it is -- if I simply use the admin website, as you said, I can get a
> list of all the devices that are registered.
>
>
>
> I've actually got the individual ATAs downloaded that way, and sorted by
> "Registered" in a Google Sheet.
>
>
>
> I cannot easily also match those up with the DNs.
>
> I suppose one could probably then write some sort of SQL command that
> would specify a long-ish list of ATA device names to query for DN.
>
>
>
> It may be time to just ask an Intern to click on each ATA that's
> registered and collect the DN and description.
>
>
>
> I thought about asking one of our DB people to do that --
>
> But they'd want a project set up, and require a complete list of the table
> schema, the IP/TCP port and data connection, etc
>
> to do things as they're used to, with a direct SQL connection to query
> against.
>
> They're not used to having to operate the way Cisco does with CLI only
> access to SQL in a limited fashion.
>
> And I get why Cisco does that... not arguing that point.
>
>
>
>
>
>
>
>
>
> On Thu, May 25, 2023 at 2:18 PM Tim Reimers 
> wrote:
>
> Nate, I may have to do that.
>
>
>
> I just thought I was crazy that ATAs don't show up in
>
> show risdb query phone
>
>
>
> T
>
>
>
> On Thu, May 25, 2023 at 2:08 PM  wrote:
>
> What is the goal?  I usually get what I want from device/phone, copy it
> all into the clipboard and paste without formatting into excel and continue
> on.  In older versions of CUCM you can change the rows per page, then edit
> the URL to make rows per page all of the devices on the system up to many
> thousands.
>
>
>
> *From:* cisco-voip  *On Behalf Of *Ryan
> Ratliff (rratliff)
> *Sent:* Thursday, May 25, 2023 10:46 AM
> *To:* Tim Reimers ; Wes Sisk (wsisk) <
> ws...@cisco.com>
> *Cc:* cisco-voip 
> *Subject:* Re: [cisco-voip] Question about RISDB queries and ATA devices?
>
>
>
> I’m not surprised the “phone” filter only shows you SEP devices. I was
> expecting RTMT to give you a friendlier way to browse around and find the
> ATAs.
>
>
>
> Do any of the other risdb CLI filters give you those devices?
>
> Interrogating the API directly is another option.
>
> https://developer.cisco.com/docs/sxml/#!risport70-api-reference/selectcmdevice
>
>
>
> -Ryan
>
>
>
> *From: *cisco-voip  on behalf of Tim
> Reimers 
> *Date: *Thursday, May 25, 2023 at 12:06 PM
> *To: *Wes Sisk (wsisk) 
> *Cc: *cisco-voip 
> *Subject: *Re: [cisco-voip] Question about RISDB queries and ATA devices?
>
> Hi Wes, Ryan, all
>
>
>
> I'm not seeing any of the registered ATAs showing up in RTMT under a
> Device Search either -- only SEP devices.
>
>
>
> I gather that you are all expecting that 'show risdb query phone' as well
> as RTMT should be showing the ATAs registered and counted alongside the SEP
> devices
>
> so long as the ATAs are running in SCCP mode and not something else...
>
>
>
> Thanks Tim
>
>
>
> On Thu, May 25, 2023 at 10:17 AM Wes Sisk (wsisk)  wrote:
>
> Yes, registration information is in RIS not in SQL(informix). I see some
> other mentions of this, but not clear resolution.
>
>
>
> Note that ATA may follow different CM server resolution and 'show risdb'
> is per-node. Aka, have you checked all nodes with CM service activated
> where ATAs might be registered?
>
>
>
> Oh, and ATAs could be h.323 for a while, so are they registering as SCCP?
>
>
>
> -w
>
>
>
> On May 25, 2023, at 9:50 AM, Tim Reimers  wrote:
>
>
>
> Hi all -
>
>
>
> I'm trying to find the ACTIVELY REGISTERED devices on my UCM 9.1 system.
>
>
>
> I need to find the list of actively registered ATA 186 devices and their
> DNs.
>
>
>
> * I'm using "show risdb query phones" command, as documented here among
> other sites
>
>
> https://getpractical.co.uk/2021/10/11/cisco-cucm-reports-from-sql-show-risdb/
>
>
>
> That seems to show only the SEPzzz devices, aka my 79XX SCCP
> phones.
>
>
>
> I don't see any ATA devices being returned.
>
> Are they not in the "phone" table of the RISDB?
>
>
>
> Thanks, Tim
>
>
>
> * my understanding is that any variation on the "run sql select" is
>
> simply querying the Oracle? database for _configured_ devices only, and
> isn't looking
>
> at the memory table of the Callmanager process to see the _currently
> registered_ devices.
>
> (I've seen a number of other forum posts where people suggested "run sql"

Re: [cisco-voip] CUCM and Webex contact center

[Brian Meade]

By hosted CUCM, do you mean Webex Calling Dedicated Instance or UCM Cloud
or some sort of HCS environment?

On Tue, Mar 14, 2023 at 4:29 PM SK  wrote:

> Hello ,
>
> Has anyone here implemented  Webex contact center with hosted CUCM ? Any
> pointers / documentation help is appreciated .
>
> Thank you .
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Re: [cisco-voip] Bug CSCwe21416 UCCX 12.5.1SU2ES04

[Brian Meade]

Some additional info:

*Workaround:* - If the "Agent Device Selection" System Parameter is not
needed we can set it to "disable" and restart the CCX Engine for the
changes to take effect. - If the "Agent Device Selection" System Parameter
is needed then Logout during extended Not-Ready periods like Lunch, breaks
etc.. to avoid receiving calls that are monitored by the system

On Fri, Feb 3, 2023, 2:45 PM JASON BURWELL 
wrote:

> FYI-For anyone planning to upgrade to UCCX 12.5.1SU2ES04 be aware of newly
> created Bug CSCwe21416. It causes agents who have selected a Not Ready
> state to be automatically put back in to a Ready state if they receive a
> direct call and answer it or not. The workaround is for Agents to Log out
> of Finesse instead of go Not Ready. Causing headaches with Agent Tracking,
> RNA, Incentives, etc. The fix is supposed to be in SU3 being released early
> March.
>
>
>
> Jason
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Re: [cisco-voip] Jabber for iPhone - Phone Services not working

[Brian Meade]

Make sure the signer of the Expressway-C cert is in the CallManager-Trust
on CUCM.

There's some caveats with the ECDSA certificate as well I believe you need
to potentially adjust from the release notes.

If using STUN keepalives, make sure the Expressways in CUCM under
Device->Expressway are using FQDN and make sure PTR records for
Expressway-C are correct in DNS.

Brian Meade

On Wed, Feb 8, 2023, 2:57 PM AbdusSaboor Khan  wrote:

> Hi Guys,
>
> Upgraded Expressway to 14, now Jabber on desktop working fine, but on
> Iphone giving No Phone services , showing other services fine.
>
> Do you giys know what config
> issue I am
> having
>
> Regards,
>
> Abdul
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Re: [cisco-voip] Problem with changing 7975g phone to SIP

[Brian Meade]

webAccess set to 1 is actually disabled.

Good reference here- https://usecallmanager.nz/sepmac-cnf-xml.html

Hopefully you can get in and view the console logs then via the webpage.
It's probably having issues parsing something.

Brian Meade

On Fri, Feb 3, 2023, 2:29 PM roger  wrote:

> Hello,
>
> I am trying to convert a 7975g phone to SIP and have it register to my PBX
> (Firebrick FB2700  latest firmware).
>
> I have done a full reset (3491672850*#) and have successfully updated the
> bootloader and firmware to SIP75.9-4-2-1S.
>
> However I am having trouble provisioning the phone and getting it to
> register with my PBX. I can get as far as the phone saying it is
> registering, but I do not see any SIP traffic from the phone. I am using a
> passive lan tap on the rj45 cable from the phone.
>
> I have tried a number of variations of the XMDefault.cnf.xml file. This is
> the current version I am trying.
>
> 
> 
> 
> 
> 
> 
> 2000
> 
> 10.151.0.1
> 
> 
> 
> 
> 
> 
>
> Similarly with the SEP.cnf.xml file.
>
> [xml]
> 
>
> SIP
>
> admin
> cisco
>
> 
> 
> D-M-Y
> GMT Standard/Daylight Time
> 
> 
> pool.ntp.org
> Unicast
> 
> 
> 
>
> 
> 
> 
> 
> 
> 2000
> 5060
> 5061
> 
> 10.151.0.1
> 
> 
> 
> 
> 
>
> 
> 
> true
> 2
> 
>
> 
> false
> false
> 0
> 1
> 0
> 0
> 0
> 0
>
> 1
> 1
> 1
> 
> 
>
> United_States
>
> 
> United_States
> 64
> 1.0.0.0-1
> 
>
> 1
>
> http://10.151.0.1/cisco/services/authentication.php
> http://10.151.0.1/xmlservices/PhoneDirectory.php
> http://10.151.0.1/xmlservices/index.php
> 
>
> 
> 
> http://10.151.0.1/xmlservices/index.php
> 96
> 0
> 96
>
> 4
>
> 0
> 
> 
> 3804
> 
> 
>
> 
> false
>
> 
> 
> 
> 
> 
> 
> 
> 
> true
> 
>
> 
> true
> x–serviceuri-cfwdall
> x-cisco-serviceuri-pickup
> x-cisco-serviceuri-opickup
> x-cisco-serviceuri-gpickup
> x-cisco-serviceuri-meetme
> x-cisco-serviceuri-abbrdial
> false
> 2
> true
> true
> 2
> 2
> 0
> true
> 
>
> 
> 6
> 10
> 180
> 3600
> 5
> 120
> 120
> 5
> 500
> 4000
> 70
> false
> None
> 
>
> 1
> false
> true
> false
> false
> none
> 101
> 3
> avt
> false
> false
> 3
>
> false
> 
>
> 0
>
> false
> 10
> false
>
> 16384
> 32766
>
> 5060
> 184
> 0
> dialplan.xml
>
> Roger
> 
> 
> 9
> SipUser
> SipUser
> SipUser
> SipUser
>
> 10.151.0.1
> 5060
> 
> 2
> 
> 3
>
> SipUser
> SipPass
>
> false
> 1
> *97
> 4
> 5
>
> 
> true
> false
> false
> true
> 
> 
> 
> 
> 
> [/xml]
>
> This combination gets the phone into registering state. But no sip traffic
> goes out on the LAN. In common with most attempts it also results in the
> loss of the web server access to the phone.
>
> $ nmap 10.151.0.129
> Starting Nmap 7.80 ( https://nmap.org ) at 2023-02-03 19:21 GMT
> Nmap scan report for 10.151.0.129
> Host is up (0.0011s latency).
> All 1000 scanned ports on 10.151.0.129 are closed
>
> Nmap done: 1 IP address (1 host up) scanned in 1.54 seconds
>
> So I have something wrong somewhere, but I cannot figure out what.
>
> Anyone got any ideas?
>
> Thanks.
>
> Roger
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Re: [cisco-voip] Alternative to Unity Connection

[Brian Meade]

You're really just doing a Forward No Answer/Forward Busy to another
number.  That would work for any voicemail platform that can look at the
Diversion header to map to a user.  I think it used to be fairly popular to
use the Microsoft voicemail platform I think through S4B but not sure it
works the same on teams.

On Fri, Nov 11, 2022 at 11:50 AM harbor235  wrote:

> Hi all,
>
> Does anyone know if Cisco Unified Communications support other voicemail
> applications other than Unity Connection?
>
> I have a ISR4331 w/VoiceBundle, latest version, testing Unity Connection
> 14.x but do not like the monthly licensing model, any options?
>
>
> thanx in advance,
>
> Mike
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Re: [cisco-voip] Call Flow data via transfers

[Brian Meade]

I think Variphy does a much better job of stitching these calls with
multiple legs together.

They'll help you setup a demo system if you want to compare.

On Tue, Sep 20, 2022, 10:28 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

[Brian Meade]

I believe 14.x is officially planned for UCCX but no commitment of a
release past that point.  UCCX 12.5 added remote agent support via reverse
proxy and agent device selection which were the big features people were
waiting for.

We do custom work to get Google Dialog Flow working with UCCX but it seems
like there's no plan to add CCAI natively to UCCX at any point and only
officially to support UCCE/WXCC.

We're moving several UCCX/UCCE customers to WXCC.  There's an ODBC
connector tool to allow accessing an on-prem DB via REST API.

On Tue, Sep 6, 2022 at 7:51 PM JASON BURWELL 
wrote:

> 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] Assistance if you are able.

[Brian Meade]

You'll want your re-routing CSS on the Remote Destination Profile to target
a partition/route pattern going to your SBC for Direct Routing.  If you're
using CUBE and not a dedicated CUBE for Direct Routing, probably worth
using some sort of prefix like  and match that to send calls to MS
Teams.  You can put that prefix right on the route pattern.

On Tue, Sep 6, 2022 at 6:39 PM Tim Smith  wrote:

> Hi Terry,
>
> I would probably try and avoid SNR.
> I think overall, it's going to give you problems.
> I think experience is going to be better if people are fully running on
> one or the other - and you just route between them.
> If you are moving away from Cisco.. what's the reasoning for still ringing
> a desk phone?
>
> What SBC are you using for Teams?
>
> Cheers,
>
> Tim
>
>
> On Wed, 7 Sept 2022 at 07:12, Terry Oakley 
> wrote:
>
>> We are adding MS Teams voice to our Cisco environment.What I am
>> hoping to do is use the feature of Single Number Reach (SNR) with Remote
>> destination so that callers into our PRI will continue to be handled by the
>> CUCM but also ring at the MS Teams voice device if setup.   So what I am
>> trying to do is take a call from the PSTN into the CUCM and using Single
>> Number Reach ring the Cisco phone (if present) and also send the call out
>> to the trunk and SBC to be sent to the MS Teams voice device. I have
>> been able to get the SNR to work to my cell phone but I have hit a mental
>> block, namely me, in now making this work to move the call to the SBC via
>> the configured trunk.
>>
>>
>>
>> I am thinking I need to setup a translation pattern to make this work but
>> I am only about 67% sure I am heading in the right direction.   The other
>> issue is that our Cisco support has been canceled in an effort to save
>> money so I am working on a production system with very little support.
>> Such fun.
>>
>>
>>
>> Any assistance would be wonderfully accepted.
>>
>>
>>
>> Terry
>>
>>
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Re: [cisco-voip] UCCX Script Browsing Issue

[Brian Meade]

Notice anything different about the scripts?  Maybe a filename or path size
limit?

On Wed, Jul 13, 2022 at 9:03 AM JASON BURWELL 
wrote:

> Has anyone had an issue where some scripts could not be seen by the Script
> Editor application browser or when searching the directory with the GUI
> however the script is there, works and can be opened up with the editor if
> you type the exact script name in the to the filename field manually? I saw
> this before once back in 11.6 but I’m seeing it again in 12.5, same server
> upgraded a few months back. Only seems to be affecting a limited number of
> scripts.
>
>
>
> Thanks
>
> Jason
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Re: [cisco-voip] Jabber 14.1 ?

[Brian Meade]

There's a big one-way audio bug for Jabber for Windows 14.0.3 and above.
Working with a customer to downgrade everything to 14.0.2.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCwb15837

It's still not fixed so I assume 14.1 is affected.

On Mon, May 16, 2022 at 10:15 PM Lelio Fulgenzi  wrote:

>
> Who’s all made the jump to 14.0/14.1?
>
> I’ve just been too busy to handle Jabber desktop updates, but might get a
> chance to take a look.
>
> Are people sticking with 12.9 or moving to 14.1? I think I saw at least
> one advisory that was, sorry, upgrade for v14
>
> Sent from my iPhone
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Re: [cisco-voip] On-premise UCCX with Cisco Collaboration Platform

[Brian Meade]

Might be worth looking at some alternatives like eGain or UpstreamWorks as
well.  The SocialMiner/CCP chat is pretty limited.

On Thu, Apr 14, 2022, 8:15 AM Adam Pawlowski  wrote:

> It is included with premium, but it is best if you are content with what
> it offers in your environment, as it’s relatively rudimentary.
>
>
>
>
> https://dcloud-cms.cisco.com/demo_news/cisco-unified-contact-center-express-uccx-enablement-lab-v2
> seems to cover CCX with SM.
>
>
>
>
> https://dcloud-cms.cisco.com/demo_news/cisco-unified-contact-center-express-uccx-12-5-su1-v1
>
> This appears to cover chat, but using 3rd party products? The AI/BOT demo
> is cool, but as far as I know the code was something internal that was
> whipped up for the demo and not shared publically.
>
>
>
> dCloud is also displaying a banner at the top that normal pedestrians
> cannot schedule sessions and need someone from Cisco to do it – hopefully
> your AM or partner contact would if you want to see it.
>
>
>
> Best,
>
>
>
> Adam
>
>
>
> *From:* cisco-voip  *On Behalf Of *Andy
> Carse
> *Sent:* Thursday, April 14, 2022 4:37 AM
> *To:* Cisco VoIP List 
> *Subject:* [cisco-voip] On-premise UCCX with Cisco Collaboration Platform
>
>
>
> Hi,
>
> We have an On-Premise UCCX v12.5 deployment and I've been asked to look at
> adding web chat to the system.
>
> I believe this was Social Miner now called Cisco Collaboration Platform
> does anyone have this in place or any helpful tips on the good or bad
> solutions as I'm struggling to find any good documentation on the Cisco
> Website for this
>
> Is there any issues around support from Cisco and is it an additional
> license cost.
>
> We have Premium licenses for the agents.
>
>
>
> TIA
>
> Rgds Andy
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Re: [cisco-voip] VMware ESXi v7.0u3

[Brian Meade]

I just did 7.0U3 for a customer and no issues reported after a couple of
weeks running so far,  They've got CUCM, Unity Connection, UCCX, and
Expressways.

On Fri, Mar 25, 2022 at 9:58 AM Thomas Maniccia 
wrote:

>
> Is anyone running ESXi v7.0u3? We're running our UC applications on spec
> based HP blades and we're looking to upgrade the ESXi but the Cisco matrix
> lists a smattering of support for 7.0 and 7.0u1.
>
> Thanks,
>  Tom
>
>
> --
> Tom Maniccia
> Network and Communication Services
> University at Buffalo
>
>
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Re: [cisco-voip] CUCM SQL queries of SQL

[Brian Meade]

It does support it.  Anything that Informix supports should work.

On Wed, Mar 16, 2022, 5:03 PM Ray Maslanka  wrote:

> Hello all,
>
> Does CUCM support SQL queries of SQL queries?
>
> For example (super simplified):
>
> run sql select dnorpattern from numplan x where intersting_things left
> join (select dnopattern from numplan where other_interesting_things) y
> where x.dnorpattern = y.dnopattern
>
>
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Re: [cisco-voip] [EXTERNAL] Re: Cost-Effective Public Certificate Authority for CUCM certificates

[Brian Meade]

We've been flipping a lot of customers over to NameCheap now.  $50/year for
multi-SAN DV certificates is pretty hard to beat.  For
CUCM/Unity/IM&P/UCCX/Expressway, ends up more like $250-$300/year.

They seem to issue certs pretty immediately since it's just Domain
Verification using email.

On Fri, Feb 18, 2022 at 1:17 PM Nick Russo via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> Unfortunately, Cisco doesn't allow for * certs with the UC platform.  If
> this is for Jabber MRA, they recently added support for ACME certificates,
> but I haven't used that.  The cheapest CA signed certs I've been able to
> find is ssls.com and the full set of certs for a typical cluster is going
> to set you back about $900 a year.  They have a couple of Collaboration
> packages that you can use for the multiple domains.  Also, they work well
> enough, but the support for ssls.com is pretty weak, so plan on at least
> a week to get your certs ordered, approved, and installed.
>
> On Friday, February 18, 2022, 09:39:50 AM PST, Lelio Fulgenzi <
> le...@uoguelph.ca> wrote:
>
>
> We use Entrust. But I think we had some sort of "Contract" that allowed
> for a specific number of certs to be issued, all on the credit system.
> Regardless of SANs.
>
> But, you're right. Cisco collab is an expensive solution to provide certs
> for.
>
> I'm really hoping that https://www.incommon.org/certificates/subscribe/ opens
> up to EDUs outside of the U.S. some time (soon).
>
> -Original Message-
> From: cisco-voip  On Behalf Of James
> Andrewartha
> Sent: Friday, February 18, 2022 4:28 AM
> To: cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] [EXTERNAL] Re: Cost-Effective Public Certificate
> Authority for CUCM certificates
>
> 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
>
>
> Digicert have killed the fact you could issue a cert for
> host.sub.example.com on your *.example.com wildcard, instead they want to
> charge you extra for those hosts so now I'm shopping around. The good news
> is there's now other places that will do wildcards with unlimited reissues
> (which most call "unlimited server licenses").
>
> I tried Comodo/Sectigo Positive Multi Domain Wildcard SSL which can even
> have multiple wildcards on the one certificate, but it only accepts CSRs
> for *.example.com, which UCM/UC/IM&P won't generate. But perhaps that's a
> limitation of the reseller I used. They also have the Comodo/Sectigo Multi
> Domain SSL Certificate (FLEX) which lets you have host SANs, but will
> charge you for each one.
>
> Anyone had success with any other CAs recently?
>
> --
> James Andrewartha
> Network & Projects Engineer
> Christ Church Grammar School
> Claremont, Western Australia
> Ph. (08) 9442 1757
> Mob. 0424 160 877
>
> On 31/3/20 04:49, Brian Meade wrote:
> > In this case, we're doing public certificates internally as well for
> > CUCM Tomcat, Unity Connection Tomcat, UCCX Tomcat, and IM&P CUP-XMPP.
> >
> > Adding the multiple presence domains is pretty easy on the IM&P side
> > and it will automatically add SAN's for those domains in the CSR.
> >
> > Expressway-E will also automatically add all domains to the CSR.
> >
> > On Mon, Mar 30, 2020 at 4:07 PM Jonatan Quezada
> > mailto:jonatan.quez...@chemeketa.edu>>
> > wrote:
> >
> >Brian, How challenging was it to do the jabber on all three domains?
> >
> >Where do you need the multiDomain cert, on the VCS-edge connector
> >right? Im looking to see what it would take to get this going for
> >our remote workers even though it seems
> >like there are few things to make sure are in place first.
> >
> >for so far its the :
> >
> >certs for dual domain- how
> >provision jabber users
> >
> >
> >On Mon, Mar 30, 2020 at 12:28 PM Brian Meade  ><mailto:bmead...@vt.edu>> wrote:
> >
> >I was originally going to go with that wildcard option but this
> >customer has 3 different presence domains to match their email
> >domains which makes the CUP-XMPP cert more complicated.
> >
> >This is my personal email so no access to InCommon certificates
> >unfortunately.
> >
> >On Mon, Mar 30, 2020 at 2:59 PM Matthew Ballard
> >mailto:mball...@otis.edu>> w

Re: [cisco-voip] SIP to iTSP best practices

[Brian Meade]

I'd do E.164 inbound/outbound on the CUBE side even if you don't do E.164
in CUCM.

On Thu, Feb 10, 2022 at 11:15 AM Matthew Huff  wrote:

> We are in the process of migrating for a legacy PTSN voice gateway (PRI)
> to a new CUBE based SIP connection to a iTSP connected via a private metro
> ethernet (not Internet based). Does anyone have a good source for recipes /
> dial-plans recommendations / best practices for this?
>
>
>
>
>
>
>
> *Matthew Huff* | Director of Technical Operations | OTA Management LLC
>
>
>
> *Office: 914-460-4039*
>
> *mh...@ox.com  | **www.ox.com *
>
>
> *...*
>
>
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[cisco-voip] Hiring- Cisco Voice Virginia Beach

[Brian Meade]

Hey Team,

We (ePlus) are looking for a Cisco Voice/UC Engineer willing to re-locate
to Virginia Beach, VA if you or anyone you know is interested-
https://recruiting.ultipro.com/EPL1000EPLUS/JobBoard/9f441248-81b7-4545-95b4-c3dfb8eadab1/OpportunityDetail?opportunityId=e74a2379-6ded-4139-a715-40259b31d8b0

Thanks,
Brian Meade
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Re: [cisco-voip] Copper plant

[Brian Meade]

I'm seeing a lot of providers trying to remove true analog copper and put
their customers on SIP services with devices on-site to convert back to
copper.  Probably worth just putting in some VG450's at that point and they
can run off 48V DC as well.

I know Phybridge also has some devices for doing PoE/Voice over a single
pair of CAT3 that I've used for campuses before if you want to go VoIP to
some of the call box devices.

On Fri, Feb 4, 2022 at 1:09 PM Lisa Notarianni 
wrote:

> Hello –
>
>
>
> We are planning future projects and would appreciate input on what others
> have done with analog lines.  We currently use Verizon for over 500 analog
> lines on campus.  They provide service to call boxes, alarm lines, elevator
> lines, house phones etc…  We also don’t have network cable runs in some
> areas so we just kept the analog service running.
>
>
>
> The idea behind all of this was to rely on Verizon Centrex service if our
> premise based VOIP phones or power went down and all phone service was lost
> on campus.  When we transitioned years ago to VoIP and moved the majority
> of lines away from Centrex, our General Counsel felt it would help with
> safety if we provided these phones in case of emergency.  I recently passed
> this by General Counsel and they still feel we need to continue to use this
> service for the same reason.  But I think the clock is ticking and from
> what I understand Verizon is abandoning copper.  They have suggested we
> transition to their VoIP service but it wouldn’t make sense to do that
> since they rely on our power. So, we would just switch to VoIP if we were
> to do that.
>
>
>
> I know there is also an LTE option but many callboxes are in fields or
> parking lots and the equipment is dated.  So, on top of needing to address
> this, we really don’t have funding to replace expensive callboxes to
> accommodate LTE service.  I know we really need to evaluate and rethink the
> need for this equipment.  We have considered transitioning the funding to a
> safety app that students, staff and faculty can use but again we would put
> the onus of safety on the user and their wireless phone – not preferred.
>
>
>
> This is complicated for Higher Ed.
>
>
>
> Any solutions or steps anyone has taken?   Is Verizon really abandoning
> all copper?
>
>
>
> Thanks,
>
>
> Lisa
>
>
>
> *Lisa Notarianni*
>
> University of Scranton
>
> Telecommunications Engineer
>
> Infrastructure Services
>
> 800 Linden St.
>
> Scranton PA 18510
>
> 570.941.4325
>
>
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Re: [cisco-voip] Hopefully this is an easy question.... user activity reporting..

[Brian Meade]

Tomcat/Tomcat Security Logs should have that info.

On Fri, Feb 4, 2022 at 1:28 PM Tim Reimers  wrote:

> Hopefully this is an easy question user activity reporting..
>
> I have an instance of UCM 9.0 still running.. sadly, totally end of
> support, etc..
>
> Does anyonw know whether it is possible to get a list of users who've
> recently used the
> CCMUser page?
>
> We're trying to figure out who still uses that to forward/unforward
> phones..
> Have bever been asked that before..
> There doesn't seem to be an "user audit log" in the canned Reports..
>
> Again, this is for the unsupported old UCM 9.0 platform...
>
> Thanks Tim
>
> --
>
> *Quis custodiet
> ipsos*
> * nexus*
>
> Tim Reimers
>
> Network Administrator
>
> I.T Services
>
> City of Asheville
>
> treim...@ashevillenc.gov
>
> (desk) 828-259-5512
>
> (cell)   828-552-1585
>
> Please do NOT use the contact information above for unsolicited sales
> contacts.
>
> All unsolicited sales calls > /dev/null
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Re: [cisco-voip] 8851 phone question

[Brian Meade]

You can have a different ring-tone per line.  You could use the route call
based on calling party number to send calls from a specific number to a
different line on the phone.

On Fri, Feb 4, 2022 at 11:31 AM James Dust 
wrote:

> Afternoon all,
>
>
>
> I have been asked if it is possible to have a different ring tone on a
> Cisco 8851 when a specific CLI calls it? (so not the selected ring tone)
>
>
>
> We are using CUCM 12.5
>
>
>
> I personally I have never heard of this, but I though I’d ask.
>
>
>
> Thanks allot
>
>
>
> James
>
> *Consider the environment - Think before you print*
>
> The contents of this email are confidential to the intended recipient and
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>
> You are advised that urgent, time-sensitive communications should not be
> sent by email. We hereby give you notice that a delivery receipt does not
> constitute acknowledgement or receipt by the intended recipient(s).
>
> Details of Charles Stanley group companies and their regulators (where
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Re: [cisco-voip] CER for Europe... variable length dial plan and PSAP callback

[Brian Meade]

For the callback, maybe just use translation patterns for each callback
number to go to a fixed-length number that matches the CTI RP for callback.

You could make dummy numbers like:
9131000
9131001

Then your ELIN would need to match in CER too to be 1000/1001 but you can
hard set the actual ELIN on your outbound route patterns.

It makes exporting PS-ALI info from CER not work but I don't see that used
much anyways.

On Wed, Feb 2, 2022 at 11:13 AM Jonathan Charles  wrote:

> We currently have offices throughout Western Europe and each site
> currently has a PRI/T1-CAS/Analog nightmare...
>
> We are implementing centralized SIP (ATT) out of London and we need to
> make sure 112/999 calls get directed to the proper nation's PSAP.
>
> We would also like to support callback.
>
> So, we spun up a CER cluster in Europe...
>
> First thing we found, CER only supports one entry point (so, we picked
> 112, and translate 999 in the UK to 112 and match an ERL that changes this
> back to 999 and tags a UK DID). This seems to work fine (albeit clunkily).
>
> The real problem is call back...
>
> CER also only supports one entry point for callback the problem is we
> have a variable length dial plan with + then 10, 11, 12 and 13 digits (each
> with 2 digit country codes).
>
>
> So, how do we do callback?
>
> First idea was to pay digits on the end in CER, strip them off on the
> outbound via SIP and then pad them back in on the inbound back to CER to
> route them to the station.
>
> Is this crazy?
>
>
> Thanks!
>
>
> Jonathan
>
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Re: [cisco-voip] CUCM Crashes on Back-up

[Brian Meade]

Take a look at the DRF traces to see what's happening.

On Fri, Jan 21, 2022, 9:48 AM Josh Nordquist 
wrote:

> This is a long shot but I have been having an issue where during scheduled
> and manual DRS back-ups our CUCM sub server is losing network connection to
> the CUCM pub. At no other time is this happening and things run fine during
> the day.
>
> It's not like a timing thing because I can try and run the back-up any
> other time and get the same disconnect.
>
> Connection to the SFTP is not a problem because the pub backs-up fine and
> another sub server backs up fine, it's just one sub that is the problem
> child.
>
> I've restarted DRS services on all nodes and have done full server resets
> as well.
>
> Has anyone seen this before? I'm tempted to just rebuild the sub with the
> issue.
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Re: [cisco-voip] Small business E911 solution

[Brian Meade]

9Line 911 is the cheapest option out there and does dynamic location
updates for users on Jabber or desk phones including MRA desk phones.

They do rely on CER though and the user is prompted to update their
location via the CER portal.  If you're on Flex licensing, CER is
free/included.

This may get confusing though on the Shared DID.  You can use cheap DIDs
just for E911 purposes I believe.

You're probably too small for UCM Cloud or Webex Calling Dedicated Instance
which include Redsky E911 for free now as well.  Webex Calling also
includes Redsky E911 now but you'd have to change your backend to no longer
be CUCM-based.

On Thu, Dec 9, 2021 at 1:41 PM Matthew Huff  wrote:

> We are in the process of moving from legacy ISDN PRI for inbound/outbound
> dialing to SIP, and E911 has hit us in the face. We have less than 50
> users, where > 90% currently are working from home. They have the same
> prime dn for both the office phone and their home phone. We have users
> that have phones in 3-4 locations including in multiple states. What is the
> simplest solution to setup and maintain that doesn’t require a user to have
> a separate DID in each location? Cisco Emergency Responder looks like major
> overkill.
>
>
>
> Our environment is:
>
> CUCM 14.x
>
> Cisco Expressway 14.x for MRA
>
> Cisco 8861 SIP phones (both at home and at work).
>
>
>
> *Matthew Huff* | Director of Technical Operations | OTA Management LLC
>
>
>
> *Office: 914-460-4039*
>
> *mh...@ox.com  | **www.ox.com *
>
>
> *...*
>
>
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Re: [cisco-voip] MRA Onboarding via activation code... phone trust list?

[Brian Meade]

The phone CA Trust List is part of the phone firmware.

I think this is still the latest-
https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cuipph/all_models/ca-list/CA-Trust-List.pdf

I don't see Let's Encrypt in there.

On Wed, Nov 17, 2021 at 9:53 AM Jonathan Charles  wrote:

> OK, TAC never responded to me, but I found the solution I did a packet
> capture from the phone and saw it come back with an invalid CA for the
> Let's Encrypt certs... I uploaded the cert chain for Let's Encrypt to
> Phone-Edge-Trust on the CCM Publisher and the phone registered.
>
> Phone-Edge-Trust uploads the certs to the Cisco Cloud, so when the phone
> gets the activation code it downloads those certs into its trust store.
>
> This cert store is designed for people using their own internal certs, but
> my phone was a CP-8845-K9=V03 I got in 2017 and probably predates the Lets
> Encrypt CA so, if you see TLS error or Invalid CA in the PCAP, it is
> worth a shot to upload the E's external cert chain to the Pub.
>
>
> Jonathan
>
> On Thu, Nov 11, 2021 at 4:57 PM Jonathan Charles 
> wrote:
>
>> Yes, they will, the Expressway E was designed around an ACME cert and
>> Let's Encrypt is super free.
>>
>> Anyway, I think the issue is between the Expressway and CUCM at this
>> point... escalating to TAc...
>>
>>
>> Jonathan
>>
>> On Thu, Nov 11, 2021 at 4:49 PM Brian V  wrote:
>>
>>> WIll the phones trust a LetsEncrypt cert ?
>>> Jabber works because the OS (Windows/MAC/iOS/Droid) gets updated root CA
>>> certs on a regular basis
>>> The trusted certs in the phone have to be placed there in the software
>>> by Cisco.
>>> This might be a situation where newer code on a phone is required if the
>>> trusted Root CA (or chain) for Lets Encrypt is missing on the phone.
>>>
>>> On Thu, Nov 11, 2021 at 11:27 AM Matthew Huff  wrote:
>>>
>>>> I wouldn’t put a lot of weight in the status on the phone with the TLS
>>>> error, I’ve seen that with working phones. Do you have the phone MRA domain
>>>> set? We have a separate device pool for MRA devices so it can set the time
>>>> from external ntp sources. If the time on the phone is off, the crypto
>>>> can fail as well.
>>>>
>>>>
>>>>
>>>> *Matthew Huff* | Director of Technical Operations | OTA Management LLC
>>>>
>>>>
>>>>
>>>> *Office: 914-460-4039*
>>>>
>>>> *mh...@ox.com  | **www.ox.com <http://www.ox.com>*
>>>>
>>>>
>>>> *...*
>>>>
>>>>
>>>>
>>>> *From:* Jonathan Charles 
>>>> *Sent:* Thursday, November 11, 2021 11:50 AM
>>>> *To:* Matthew Huff 
>>>> *Cc:* Brian Meade ; cisco-voip voyp list <
>>>> cisco-voip@puck.nether.net>
>>>> *Subject:* Re: [cisco-voip] MRA Onboarding via activation code...
>>>> phone trust list?
>>>>
>>>>
>>>>
>>>> It is running 12.8... it has been locally reg'd before...
>>>>
>>>>
>>>>
>>>> On Thu, Nov 11, 2021 at 10:44 AM Matthew Huff  wrote:
>>>>
>>>> In the lab, have you tried setting up the phone without MRA and get the
>>>> firmware uploaded first? Depending on how old the firmware is, you may have
>>>> issues with onboarding. Our 8861 wouldn’t onboard until at least 12.5.
>>>>
>>>>
>>>>
>>>> *Matthew Huff* | Director of Technical Operations | OTA Management LLC
>>>>
>>>>
>>>>
>>>> *Office: 914-460-4039*
>>>>
>>>> *mh...@ox.com  | **www.ox.com <http://www.ox.com>*
>>>>
>>>>
>>>> *...*
>>>>
>>>>
>>>>
>>>> *From:* cisco-voip  *On Behalf Of 
>>>> *Jonathan
>>>> Charles
>>>> *Sent:* Thursday, November 11, 2021 11:10 AM
>>>> *To:* Brian Meade 
>>>> *Cc:* cisco-voip voyp list 
>>>> *Subject:* Re: [cisco-voip] MRA Onboarding via activation code...
>>>> phone trust list?
>>>>
>>>>
>>>>
>>>> On the phone, we see TLS connection

Re: [cisco-voip] MRA calls dropping

[Brian Meade]

Grab the CallManager traces.  They should show why CUCM is sending a BYE.

On Wed, Nov 24, 2021 at 11:34 AM Jonathan Charles  wrote:

> Both internal and external, multiple users, multiple locations
>
> We are seeing call drops, and the only things the logs (on the Expressway
> E's) show is a Bye coming from CUCM.
>
> tvcs: Event="Request Received" Service="SIP" Src-ip="10.1.28.210"
> Src-port="25845" Dst-ip="10.2.9.165" Dst-port="7001"
> Call-serial-number="b5ef5cdf-c039-48ec-91af-8ef6f4f4204c"
> Tag="dfd1d116-6100-48f6-8dcf-d2efe0b6f552" Protocol="TLS" Method="BYE"
> Request-URI="sip:4ed5aa08-624f-82a4-8c7f-6863f6545aa9@192.168.10.25:53171;transport\=tls"
> To="sip:6656...@cucms01.banana.com" Level="2" UTCTime="2021-11-24
> 16:10:45,389"
>
> CUCM is 12.5
> Expressway is v14.0.3
>
> Just wondering what to check...
>
>
> Thanks!
>
>
> Jonathan
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Re: [cisco-voip] MRA Onboarding via activation code... phone trust list?

[Brian Meade]

What's the console logs show?

The Expressway needs to be signed by one of the trusted CAs listed that are
part of the phone firmware.

The Expressway cert authenticates the phone with the MIC.

Do you have activation code onboarding enabled under the MRA config on the
Expressway-C?

On Fri, Nov 5, 2021, 5:30 PM Jonathan Charles  wrote:

> So, I set up activation code MRA for an 8845 (lab first)...
>
> Cloud onboarding worked, got an activation code, tried it out...
>
> Phone kicks back 'check internet connectivtity' and on the status on the
> phone says:
>
> GDS Handshake Succeeded
> A TLS connection failed...
>
> GDS is Cisco's cloud onboarding thingy I am assuming it didn't like
> the TLS connection the expressway, but I don't see anything in the
> Expressway logs...
>
> There is a bug and it says we need to load a Hydrant cert back into the
> trust store...
> https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvt67257?rfs=iqvred
>
> But where do we need to load it? Tomcat Trust? On the Expressways? The bug
> doesn't say... it needs to be pushed to the phone's trust list, how do you
> do that?
>
>
> Thanks!
>
> Jonathan
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Re: [cisco-voip] remote training for uccx agents.

[Brian Meade]

We've made agents supervisors of some training teams in the past to allow
this out of the box.

Using a Quality Management add-on suite is probably going to be better for
both use cases though.

On Wed, Sep 15, 2021 at 5:46 PM Scott Voll  wrote:

> What are others doing with new agents in UCCx.  do you have a solution
> with new employees and training where the new user can listen in on more
> seasoned agents?  and then a way for trainers to be able to listen in on
> the new agent and provide feedback / info?  Silent monitoring / whisper?
>
> TIA
>
> Scott
>
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Re: [cisco-voip] Question about call forwarding single channel on ISDN PRI (H.323 to UCM)

[Brian Meade]

To do it on the router, you may have to switch your incoming dial-peer to
use answer-address rather than incoming called-number so you could match a
different incoming dial-peer based on calling number.

It's probably going to be easier to do it on the CUCM side using the same
setup as the CUCM call blocking where you choose next hop based on
calling-number.

On Thu, Sep 9, 2021 at 9:27 AM Tim Reimers  wrote:

> Hi all -
>
> Long time lurker here, haven't posted in several years.
>
> I am trying to figure out how to _selectively_  forward calls arriving on
> an ISDN PRI, based on incoming CallerID.
> (eg, test with just my cell phone, and not forward all calls)
>
> This is on an 2821 IOS voice gateway that runs H.323 to a UCM 9.1 server.
> No MGCP here.
>
> I'd prefer to handle this forwarding directly on the router, but can do so
> on the UCM
> if I need to.
>
> Normally the dial peers send all inbound calls on the IOS voice gateway to
> either
> the UCM or the RightFax server (session-target style)
>
> Can this be done based on the incoming CallerID?
> That way, I can forward only calls from certain test phones.
>
> Thanks Tim
>
>
> --
>
> *Quis custodiet
> ipsos*
> * nexus*
>
> Tim Reimers
>
> Network Administrator
>
> I.T Services
>
> City of Asheville
>
> treim...@ashevillenc.gov
>
> (desk) 828-259-5512
>
> (cell)   828-552-1585
>
> Please do NOT use the contact information above for unsolicited sales
> contacts.
>
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Re: [cisco-voip] odbc connection to uccx12

[Brian Meade]

Under Tools->Password Management under CCX Admin, you can set the password
and it will update CUIC as well.

The password has to be set separately on both nodes.

On Thu, Sep 2, 2021 at 11:57 PM naresh rathore  wrote:

> hi
>
>
> customer's database teams wants odbc connection with uccx version 12.
>
>
>
>
> https://ccxguru.wordpress.com/2018/01/31/creating-odbc-connection-with-uccx/
>
> it seems like uccxhruser user should be used with password  5:T{i,5e!KqD*8.
> but based on following link, the password is no longer valid. can I update
> password and will it have any impact on cuic?
>
>
> https://community.cisco.com/t5/contact-center/how-to-get-datasource-default-information-in-uccx-11-6/td-p/3230610
>
> Regards
>
>
> Naresh Rathore
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Re: [cisco-voip] SIP Log Export CDR to CSC

[Brian Meade]

Maybe better to just use the CUBE CDR features instead rather than having
to scrape the info out of the debugs?

On Wed, Jun 23, 2021 at 11:26 AM Lizzy Anderson 
wrote:

> Morning Folx,
>
> I have a mess of DIDs to test for a port and need a way to take SIP logs
> in txt format from ISR/CUBE debug ccsip messages and dump the high level
> cdr info from the calls into a CSV. I really only need info like calling
> number and called number. I looked at TranslatorX but it can only export
> the full CDR data one call at a time, ideally I would take that whole calls
> list and export that.
>
> Thanks for the help!
> --
> Erik Anderson
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Re: [cisco-voip] Suggestions on book on SIP protocol?

[Brian Meade]

This training is pretty great for that kind of information-
https://www.thesipschool.com/

On Tue, Jun 15, 2021 at 11:12 AM ROZA, Ariel via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> People,
>
>
>
> Can anyone give me a good reference on books that explain the SIP Protocol?
>
>
>
> I already have a significant working knowledge on SIP. I have set up some
> trunks On CUCM and CUBE and troubleshooted many more. But I am still
> lacking on the WHYs of many things. So I am looking, for example, for
> theoretical explanations of SIP call flows and the variations of headers
> and options, without being dry as hell as a RFC or Cisco technical
> documents. Some good book that can be used as reference material.
>
> One of the issues I have with Cisco tech notes is that when they do
> examples-by-debug often they lack explanations or discussion of
> possibilities.
>
>
>
> What have you people read and can recommend?
>
>
>
> Regards,
>
>
>
> 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
>
>
>
> [image: firmacalidad-1]
>
>
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Re: [cisco-voip] Old expressway

[Brian Meade]

Just running "xConfig" should give you the whole backup minus the
certificates.

On Tue, May 25, 2021, 11:59 AM Andy Carse  wrote:

> Gents,
> I know this is a long shot but I'll ask anyway.
> I have a customer who is running an old pair of Expressway C/E they are
> having an issue with the expressway-e you can't web browse to it and video
> calls are not working, I suspect it thinks it's in maintenance mode for
> some reason which no one is owning up too. As its old and the config hasn't
> changed they don't have or can't locate a backup..
> Is there someway to back it up via the cli, I'm guessing not and my only
> way out is to build it from scratch which is probably the quickest way out
> of this.
>
> Its version X8.7 and it hasn't been upgraded as "it just seemed to
> work"until it doesn't
>
> I'll take a stab at saying even if it is rebuilt the licensing will be the
> next issue and then they will be forced into upgrading it.
>
> Rgds Andy
>
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Re: [cisco-voip] Blf on cucm end user portal

[Brian Meade]

You can make your own webpage to allow them to do this.

On Fri, May 21, 2021, 2:05 PM Myron Young  wrote:

> Hello all,
>
> Just found out that BLF is not a configurable option for an end user in
> the CUCM my phone portal.
>
> Has anyone found a workaround for this scenario?
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Re: [cisco-voip] Pairing Webex Desktop (Windows app) with Desk Phone

[Brian Meade]

James,

Do you have the device as a Controlled Device under your end user with
appropriate CTI permissions?

Does your User Profile have appropriate CTI Servers configured in CUCM as
FQDN?

Are you testing internally or over MRA?  CTI Control doesn't work over MRA.

Thanks,
Brian Meade

On Sat, Mar 27, 2021 at 2:22 PM James Dust 
wrote:

> Hi all,
>
> I am having issues pairing my desk phones (Cisco 8851’s) with the Webex
> (formally teams) desktop app. Within the app itself no devices are
> discoverable.
>
> Are there any prerequisites I need to set on my CUCM environment? (version
> is 11.5)
>
> Any pointers much appreciated.
>
>
> Kind regards
>
> James
>
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Re: [cisco-voip] 8845 joining zoom meetings

[Brian Meade]

Any issues with regions/locations not having enough bandwidth?

Some Zoom orgs require encrypted audio/video so that could be another item
to check that a proper encrypted security profile is applied.

On Wed, Feb 10, 2021 at 11:03 AM Josh Nordquist 
wrote:

> All,
>
> Wondering if any others are having this issue. We have 8845 video phones
> and some of those folks want to join zoom meetings but when they dial the
> SIP URI (x...@zoomcrc.com) the 8845 just shows a black screen and
> it's just silence for audio. The black screen makes me think the 8845
> thinks it's in a video call but nothing gets displayed.
>
> These same 8845 phones can join external webex and skype meetings just
> fine and all our other video endpoints can join zoom meetings fine (jabber
> and webex room devices).
> Just wondering if anyone else has seen this with Zoom calls from a 8845.
> It can easily be tested by dialing 0...@zoomcrc.com from a 8845 and seeing
> it you reach a screen asking you to enter a meeting number.
>
> Thanks
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Re: [cisco-voip] Sip profiles

[Brian Meade]

You can make a new route pattern to Unity Connection and strip the Caller
ID there.  Or do you need it masked in the CDR?

You may be able to use a specific incoming called-number dial-peer with a
SIP Profile if you really need to keep the number out of CDR.

On Fri, Jan 29, 2021, 4:21 PM f...@browardcommunications.com <
f...@browardcommunications.com> wrote:

> Greetings all, looking for advice/ config example to make inbound calls to
> a voicemail box anonymous. Example number:
> 5556781234
>
> Need config for sip profile and dial peer. Thanks!
>
> Call flow would be:
> Pstn>5556781234>vcube>sip profile? Anonymous > cucm > unity mb.
>
> Thanks in advance!
>
>
>
> Sent from my iPhone
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Re: [cisco-voip] [External] RE: Third Party Softphone w/ TLS

[Brian Meade]

For SIP Phones, it shouldn't require mutual TLS if you have the Digest
Authentication set under the Security Profile.

Do the CallManager traces show the incoming registration attempt and have
anything interesting?

On Thu, Jan 21, 2021 at 7:41 PM Johnson, Tim  wrote:

> Thanks for the suggestions so far!
>
>
>
> I am using digest authentication. I have not tried restarting Tomcat, but
> since I did not upload anything to CallManager, I’m not sure it’ll be
> required. Either way, easy enough to try it!
>
>
>
> I know with a SIPS trunk, I was required to upload a client cert into
> CM-trust. I guess I was just hopeful that I wouldn’t have to do it with
> client devices because I can’t get my hands on the software to test myself,
> so I have to work through someone else. Hmm, maybe I’ll consider VPN if I
> can’t get it working otherwise.
>
>
>
> *From:* Adam Pawlowski 
> *Sent:* Thursday, January 21, 2021 7:25 PM
> *To:* Kent Roberts ; Johnson, Tim 
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* [External] RE: [cisco-voip] Third Party Softphone w/ TLS
>
>
>
> I looked at how to secure this briefly for a polycom endpoint and the
> explanation in that documentation was that you had to supply a certificate
> as the client.
>
> So, from that much your assessment that the softphone needs to be
> presenting some sort of client certificate sounds about right.
>
>
>
> I would be curious to hear what the outcome is, as we’re starting to let
> in some more 3rd party devices from Axis, ClearOne, Crestron. 9/10 times
> I ask about SRTP and SIPS support and the customer has no idea what I’m
> talking about, but some day someone is going to call my bluff.
>
>
>
> I’m not sure what your application is but a targeted VPN connection is
> probably going to be an easier lift, especially if you’re going to enable
> TLS 1.0.
>
>
>
> Adam
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Kent
> Roberts
> *Sent:* Thursday, January 21, 2021 6:35 PM
> *To:* Johnson, Tim 
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Third Party Softphone w/ TLS
>
>
>
> Did you restart tomcat after adding the trust?   Seems that is the thing
> with Cisco these days….. and I am told that in newer versions, restarting
> the server will be required, as restarting the service isn’t enough….
> Only thing I though of was ok windows….
>
>
>
> On Jan 21, 2021, at 9:55 AM, Johnson, Tim  wrote:
>
>
>
> Does anyone have a working configuration of using a third party SIP
> softphone with TLS? I have it working with Cisco phones and Jabber, but am
> trying to get a third party client working. I’m on CUCM 12.0.
>
>
>
> So far, I’m running into an issue with the TLS handshake. The client is
> using TLS 1.0, and I confirmed that my CUCM nodes do support 1.0. I’ve put
> the CallManager cert in the trusted root (local machine) on the Windows
> client. When attempting to register the client, CUCM gives an error “peer
> did not return a certificate.” That led me to think that I would need to
> get a signed cert uploaded as a CM-trust cert. I opened a ticket with TAC
> to ask if that’s the case (would rather not have to do a client cert if I
> don’t need to) and they suggested I may not need one. I haven’t been able
> to get more out of them on this yet (after a week), so I figured I’d ask
> here.
>
>
>
> Tim Johnson
>
> Voice & Video Engineer
>
> Central Michigan University
>
>
>
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Re: [cisco-voip] Third Party Softphone w/ TLS

[Brian Meade]

I think enabling Digest Authentication on the Phone Security Profile is the
workaround to not need mutual TLS.

Do you have Digest Authentication checked on the Phone Security Profile and
an end user set as the digest user on the phone config and a digest
password configured under the end user?

On Thu, Jan 21, 2021 at 11:56 AM Johnson, Tim  wrote:

> Does anyone have a working configuration of using a third party SIP
> softphone with TLS? I have it working with Cisco phones and Jabber, but am
> trying to get a third party client working. I’m on CUCM 12.0.
>
>
>
> So far, I’m running into an issue with the TLS handshake. The client is
> using TLS 1.0, and I confirmed that my CUCM nodes do support 1.0. I’ve put
> the CallManager cert in the trusted root (local machine) on the Windows
> client. When attempting to register the client, CUCM gives an error “peer
> did not return a certificate.” That led me to think that I would need to
> get a signed cert uploaded as a CM-trust cert. I opened a ticket with TAC
> to ask if that’s the case (would rather not have to do a client cert if I
> don’t need to) and they suggested I may not need one. I haven’t been able
> to get more out of them on this yet (after a week), so I figured I’d ask
> here.
>
>
>
> Tim Johnson
>
> Voice & Video Engineer
>
> Central Michigan University
>
>
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Re: [cisco-voip] Alternatives for MediaSense simple recording?

[Brian Meade]

I don't think Unity Live Record uses the BIB at all if I remember
correctly.  I think it just uses whatever CFB resources you have in your
MRGL.  I've never had an issue with 2 one-sided voicemails when I've used
it.

I've never tried the Record softkey though in CUCM.  How do you configure
the Live Record extension?

I think CME has a Live Record softkey and lets you set a Live Record
extension to point to CUE.

It sucks we can't have custom softkeys on CUCM but you can use a SURL
button to do some automation.  I think there's some companies out there
that have made stuff with the IP Phone Services SDK to automate some stuff
with a SURL button such as conferencing in another party.  Maybe 2Ring and
Imagicle?

On Mon, Jan 11, 2021 at 4:18 PM NateCCIE  wrote:

> There is a record softkey that does what you’re asking Leilo.  The problem
> with Unity Live record is the CUCM BIB calls the number twice, one for TX
> and one for RX, and you get two-one sided voicemails in the inbox.  If that
> gets fixed, it’s an amazing option.
>
>
>
> *From:* cisco-voip  *On Behalf Of *Lelio
> Fulgenzi
> *Sent:* Monday, January 11, 2021 2:00 PM
> *To:* Nick Barnett ; Brian Meade 
> *Cc:* cisco-voip 
> *Subject:* Re: [cisco-voip] Alternatives for MediaSense simple recording?
>
>
>
> It’s too bad that CUCM doesn’t have a LiveRecord softkey macro that does
> the conferencing and dialing of the live record extension.
>
>
>
> To ask people to press conference and then dial live record and the
> conference again, is just way to much to ask. I think.
>
>
>
> Has Live Record support from CUCM side improved at all?
>
>
>
>
>
> *From:* Nick Barnett 
> *Sent:* Monday, January 11, 2021 3:50 PM
> *To:* Brian Meade ; Lelio Fulgenzi 
> *Cc:* cisco-voip 
> *Subject:* Re: [cisco-voip] Alternatives for MediaSense simple recording?
>
>
>
> *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
>
>
>
> Unity live record, that's one I haven't thought of yet. Thanks!
>
>
>
> I'm pretty sure, mediasense is totally dead. We just upgraded to CUCM 12.5
> SU3 in October. MediaSense 11.5 su2 said it was compatible with CUCM 12.x,
> but in this case, it only meant 12.x THRU 12.5 SU2.  The BU's solution was
> to downgrade to SU2. We kind of pushed them and they came back with a fix.
> Apparently between CUCM 12.5 SU2 and 12.5 SU3, CUCM forced HTTPs for AXL
> connections.  to fix it, TAC had to root into my nodes and make a change to
> the haproxy.conf file to stop forcing HTTPS.
>
>
>
> This whole mess took me right up to the last day of support and I think
> everyone at cisco hated me, but they were clear there would be no more
> support for this monster. meh
>
>
>
> Thanks,
>
> Nick
>
>
>
> On Mon, Jan 11, 2021, at 2:24 PM, Brian Meade wrote:
>
> Unity Connection Live Record may be an option you could try and have it
> conference in that number.
>
>
>
> I think MediaSense is still around for video call handlers/video
> voicemail/video on hold if I remember correctly.  I think they only killed
> it for recording calls.
>
>
>
> On Mon, Jan 11, 2021 at 1:56 PM Lelio Fulgenzi  wrote:
>
> I sure was sad when they EOL’ed Media Sense. I really wanted to do video
> call handlers and video voicemail and greetings.
>
>
>
> Take a look at https://www.mns.vc/ they might have what you’re looking
> for.
>
>
>
> Lelio
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Nick
> Barnett
>
> *Sent:* Monday, January 11, 2021 12:54 PM
>
> *To:* cisco-voip 
>
> *Subject:* [cisco-voip] Alternatives for MediaSense simple recording?
>
>
>
> *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
>
>
>
> Hey folks. What are people using now that MediaSense is EOL? It was fine
> for what it was. It just recorded anything you threw into it. it weaseled
> it's way into some weird apps we have, and now I'm kinda stuck.  We have an
> iphone app that was developed to work in areas with poor data connectivity.
> It creates a conference call to a PSTN number that routes into our system
> and is a route pattern attached to a SIP trunk directly to MediaSense.
>
>
>
> From there, we use APIs to pull the file down and save it using meta data
> from the initial call.
>
>

Re: [cisco-voip] Alternatives for MediaSense simple recording?

[Brian Meade]

Unity Connection Live Record may be an option you could try and have it
conference in that number.

I think MediaSense is still around for video call handlers/video
voicemail/video on hold if I remember correctly.  I think they only killed
it for recording calls.

On Mon, Jan 11, 2021 at 1:56 PM Lelio Fulgenzi  wrote:

> I sure was sad when they EOL’ed Media Sense. I really wanted to do video
> call handlers and video voicemail and greetings.
>
>
>
> Take a look at https://www.mns.vc/ they might have what you’re looking
> for.
>
>
>
> Lelio
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Nick
> Barnett
> *Sent:* Monday, January 11, 2021 12:54 PM
> *To:* cisco-voip 
> *Subject:* [cisco-voip] Alternatives for MediaSense simple recording?
>
>
>
> *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
>
>
>
> Hey folks. What are people using now that MediaSense is EOL? It was fine
> for what it was. It just recorded anything you threw into it. it weaseled
> it's way into some weird apps we have, and now I'm kinda stuck.  We have an
> iphone app that was developed to work in areas with poor data connectivity.
> It creates a conference call to a PSTN number that routes into our system
> and is a route pattern attached to a SIP trunk directly to MediaSense.
>
>
>
> From there, we use APIs to pull the file down and save it using meta data
> from the initial call.
>
>
>
> We aren't using ANY of the recording profiles or advanced features of
> mediasense. Our new recording system is NICE Engage and they don't offer
> any way to record via route patterns.
>
>
>
> Are there any open source, or really ANYTHING else out there that can do
> this simple procedure? The most basic of requirements are 1) non
> proprietary audio format 2) retrievable with an API or script. My cisco
> account team can only recommend Webex for recording which doesn't look to
> allow recording with a route pattern. Our VAR sells NICE which requires an
> extra application to kick of a recording like this.
>
>
>
> What are you guys using? Any suggestions for me?
>
>
>
> Thanks,
>
> Nick
>
>
>
> P.S. just to be clear, MeidaSense is not our quality assurance platform.
> We use NICE Engage for that and it's fine for now... just looking for
> something to fill the gap left by a disappearing MediaSense and our route
> pattern recording method.
>
>
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Re: [cisco-voip] IPCC best practice

[Brian Meade]

Doing it on Unity Connection historically had some benefits with the
calendaring but now you can easily do that in UCCX.

If you're worried about running out of CTI ports, offloading non-queue
calls to Unity Connection can help a lot there.

There is some light reporting in Unity Connection around what options are
pressed in Call Handlers but you won't be able to pass that info to agents
or store in the UCCX reporting.  Sometimes just the reporting requirements
will mean you have to keep the whole call flow in UCCX.

On Wed, Aug 19, 2020 at 8:19 AM f...@browardcommunications.com <
f...@browardcommunications.com> wrote:

>
> 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] UCCX REST JSON Array Response to String

[Brian Meade]

This is probably something Anthony knows off the top of his head but at
least we'll get his answer archived.

I've got a UCCX Script doing a REST Call and the response JSON is actually
an array (indicated by the square brackets) even though it's only one
response.  The resulting String looks like this:
U"[{\"status\":\"Delivered\"}]"

Converting this into a Document then JSON Document seems to not work
properly because of this bracket issue.  The JSON Document ends up looking
like this:
TEXT[[{\"status\":\"Delivered\"}]]"

The GET JSON Document Data step then fails to find anything using the
JSONPath.

To work around it, we used substring to remove the set of square brackets
around the original response string before converting to a Doc which works
and resolves the issue but I'm thinking there's gotta be a better
solution.

I imagine many JSON responses are probably going to contain arrays.  In
this case, we're always getting a single result but I can imagine this
would be an issue where the REST step actually comes back with multiple
items in the array.

Any ideas?
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Re: [cisco-voip] Is CME media flow through by default?

[Brian Meade]

Gotta make sure it's extra hidden!

On Thu, Jul 30, 2020 at 2:07 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I agree with you Brian, so then, why do so many people type the
> command address-hiding in to their CUBE configuration?  It just baffles me.
>
> On Wed, Jul 29, 2020 at 1:36 PM Brian Meade  wrote:
>
>> Yea, CUBE is really just any config in which it's IP to IP on both sides
>> of the router.  CME with a SIP or SCCP phone and a VoIP dial-peer to a SIP
>> carrier would be flow-through/address-hiding by default just like any VoIP
>> dial-peer by default.
>>
>> On Wed, Jul 29, 2020 at 4:13 AM Gerence Guan 
>> wrote:
>>
>>> Hi all
>>>
>>> If using Cisco SIP IP phone on CME which terminates the SIP trunk from
>>> service provider SBC, will it be media flow through by default? Can CME do
>>> the IP address hiding between the voice vlan and service provider, like
>>> what CUBE does?
>>>
>>> Best Regards,
>>> Guan
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
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>>
>
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Re: [cisco-voip] Is CME media flow through by default?

[Brian Meade]

Yea, CUBE is really just any config in which it's IP to IP on both sides of
the router.  CME with a SIP or SCCP phone and a VoIP dial-peer to a SIP
carrier would be flow-through/address-hiding by default just like any VoIP
dial-peer by default.

On Wed, Jul 29, 2020 at 4:13 AM Gerence Guan  wrote:

> Hi all
>
> If using Cisco SIP IP phone on CME which terminates the SIP trunk from
> service provider SBC, will it be media flow through by default? Can CME do
> the IP address hiding between the voice vlan and service provider, like
> what CUBE does?
>
> Best Regards,
> Guan
> ___
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>
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Re: [cisco-voip] Automated SIP Testing

[Brian Meade]

I've used SIPP a lot in the past-  http://sipp.sourceforge.net/

On Wed, Jul 8, 2020 at 10:53 AM UC Penguin  wrote:

> I’m curious if anyone has any experience with SIP testing tools.
>
> I’m looking for a tool to be able to script testing valid configurations.
> Ex. Does MS Teams actually accept a call to this URI or does the group that
> manages MS Teams need to fix something on there side.
>
> TIA
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Re: [cisco-voip] Expressway Cluster failover for MRA

[Brian Meade]

Yea, I haven't ran into any issues with that.

On Tue, Jul 7, 2020 at 5:03 PM Mark H. Turpin  wrote:

> I haven't tried... Can you do two Unified Comm zones to a single CUCM?
>
> ------
> *From:* Brian Meade 
> *Sent:* Tuesday, July 7, 2020 3:46 PM
> *To:* Mark H. Turpin 
> *Cc:* Gerence Guan ; Anthony Holloway <
> avholloway+cisco-v...@gmail.com>; cisco-voip voyp list <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] Expressway Cluster failover for MRA
>
> *** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
>
> Those scenarios seem to refer to cross-connecting separate standalone C/E
> pairs.  In the case of 2 standalone C/E pairs, neither knows about the
> other so it shouldn't be an issue.
>
> On Tue, Jul 7, 2020 at 3:40 PM Mark H. Turpin  wrote:
>
> I don't believe running unclustered is supported though.
>
> The way I interpreted this section in Unsupported Deployments:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-6/exwy_b_mra-expressway-deployment-guide/exwy_b_mra-expressway-deployment-guide_chapter_011.html
> <https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2Fexpressway%2Fconfig_guide%2FX12-6%2Fexwy_b_mra-expressway-deployment-guide%2Fexwy_b_mra-expressway-deployment-guide_chapter_011.html&data=01%7C01%7Cmturpin%40covene.com%7C012388a587a9480f2b1f08d822b6e1ab%7C575b0cc755204e999cb37affbf511f45%7C1&sdata=6hBzSfQkY2VPrUz22DWnFp0BX29aLwKyBd0Cm2WFxfs%3D&reserved=0>
>  read
> to me like you needed to have your C's and E's clustered for your UC zones.
>
> That's just my interpretation, though, I might be wrong.
>
>
>
> --
> *From:* cisco-voip  on behalf of
> Gerence Guan 
> *Sent:* Monday, July 6, 2020 8:04 PM
> *To:* Anthony Holloway 
> *Cc:* cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] Expressway Cluster failover for MRA
>
> *** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
>
> @Brian
> Clustering is not that critical. As long as the Jabber can register back
> via the DR without any manual system level changes. It is acceptable even
> if users need to logout and login jaber again.
>
> @Anthony
> it would be good if someone has that table. It will help a lot.
>
>
>
>
>
>
>
>
>
>
> On Tue, Jul 7, 2020 at 1:57 AM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> Brian,
>
> This wouldn't support failover in all scenarios though, correct?  E.g.,
> CUCM sub to sub failover.
>
> Does anyone have a nice table of failover scenarios covered and not
> covered by expressway clustering versus not?
>
> On Mon, Jul 6, 2020 at 9:29 AM Brian Meade  wrote:
>
> I would not use Expressway clustering and just have 2 different C/E pairs
> with different SRV Weights/Priorities instead.
>
> On Mon, Jul 6, 2020 at 3:17 AM Gerence Guan  wrote:
>
> Hi Everyone.
>
> I was googling the answer for MRA failover and found this maillist.
> Got a similar setup as Jonathan's environment.
> Having a pair of expressway C&E in primary DC, and planning to setup
> another pair of expressway C&E in the DR site. All MRA should go via
> primary DC, only use DR site when primary is down.
>
> Can I achieve this with different priorities in SRV? Anyone tested  or
> make it working?
>
> Best Regards,
> Guan
>
> >>>* On Jan 28, 2020, at 8:49 PM, Jonathan Charles  >>><https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=01%7C01%7Cmturpin%40covene.com%7C012388a587a9480f2b1f08d822b6e1ab%7C575b0cc755204e999cb37affbf511f45%7C1&sdata=xdB6FXpGB0t1lft%2FiSCBgeuaClAQyfyqhJ9UUqbUG7U%3D&reserved=0>>
> >>> wrote:
> *>>>>>>* We have two pairs of Expressway clusters (C/E) at two different
> *>>>* locations (primary and DR)...
> *>>>>>>* The cluster is up, however, we want to make sure that we are in
> *>>>* Active/Standby.
> *>>>>>>* Currently, we have one of our SRV records for collab-edge set at 5 
> (the
> *>>>* backup is at 10) with the same weight.
> *>>>>>>* The clustering guide says we should set the priority and weight on 
> both
> *>>>* SRV records the same, which will cause half of the registrations to go 
> to
> *>>>* the DR site. It is far away and has less capability.
> *>>>>>>* How do we:
> *>>>>>>* 1 - Make sure the primary site handles all MRA registrations and the 
> DR
> *>>>* site is on

Re: [cisco-voip] Expressway Cluster failover for MRA

[Brian Meade]

Those scenarios seem to refer to cross-connecting separate standalone C/E
pairs.  In the case of 2 standalone C/E pairs, neither knows about the
other so it shouldn't be an issue.

On Tue, Jul 7, 2020 at 3:40 PM Mark H. Turpin  wrote:

> I don't believe running unclustered is supported though.
>
> The way I interpreted this section in Unsupported Deployments:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-6/exwy_b_mra-expressway-deployment-guide/exwy_b_mra-expressway-deployment-guide_chapter_011.html
>  read
> to me like you needed to have your C's and E's clustered for your UC zones.
>
> That's just my interpretation, though, I might be wrong.
>
>
>
> --
> *From:* cisco-voip  on behalf of
> Gerence Guan 
> *Sent:* Monday, July 6, 2020 8:04 PM
> *To:* Anthony Holloway 
> *Cc:* cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] Expressway Cluster failover for MRA
>
> *** EXTERNAL EMAIL - DO NOT CLICK LINKS ***
>
> @Brian
> Clustering is not that critical. As long as the Jabber can register back
> via the DR without any manual system level changes. It is acceptable even
> if users need to logout and login jaber again.
>
> @Anthony
> it would be good if someone has that table. It will help a lot.
>
>
>
>
>
>
>
>
>
>
> On Tue, Jul 7, 2020 at 1:57 AM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> Brian,
>
> This wouldn't support failover in all scenarios though, correct?  E.g.,
> CUCM sub to sub failover.
>
> Does anyone have a nice table of failover scenarios covered and not
> covered by expressway clustering versus not?
>
> On Mon, Jul 6, 2020 at 9:29 AM Brian Meade  wrote:
>
> I would not use Expressway clustering and just have 2 different C/E pairs
> with different SRV Weights/Priorities instead.
>
> On Mon, Jul 6, 2020 at 3:17 AM Gerence Guan  wrote:
>
> Hi Everyone.
>
> I was googling the answer for MRA failover and found this maillist.
> Got a similar setup as Jonathan's environment.
> Having a pair of expressway C&E in primary DC, and planning to setup
> another pair of expressway C&E in the DR site. All MRA should go via
> primary DC, only use DR site when primary is down.
>
> Can I achieve this with different priorities in SRV? Anyone tested  or
> make it working?
>
> Best Regards,
> Guan
>
> >>>* On Jan 28, 2020, at 8:49 PM, Jonathan Charles  >>><https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=01%7C01%7Cmturpin%40covene.com%7C0730371d5e454e63e75e08d822122c88%7C575b0cc755204e999cb37affbf511f45%7C1&sdata=sdGVX1ROeLsbvAuSCrywpKcWb%2B11J1gTeF2dY4oNmnk%3D&reserved=0>>
> >>> wrote:
> *>>>>>>* We have two pairs of Expressway clusters (C/E) at two different
> *>>>* locations (primary and DR)...
> *>>>>>>* The cluster is up, however, we want to make sure that we are in
> *>>>* Active/Standby.
> *>>>>>>* Currently, we have one of our SRV records for collab-edge set at 5 
> (the
> *>>>* backup is at 10) with the same weight.
> *>>>>>>* The clustering guide says we should set the priority and weight on 
> both
> *>>>* SRV records the same, which will cause half of the registrations to go 
> to
> *>>>* the DR site. It is far away and has less capability.
> *>>>>>>* How do we:
> *>>>>>>* 1 - Make sure the primary site handles all MRA registrations and the 
> DR
> *>>>* site is only used when the primary is down.
> *>>>* 2 = Make sure failover occurs automatically... currently Jabber users
> *>>>* have to log out and back in to connect to the DR site.
> *>>>
>
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Re: [cisco-voip] Expressway Cluster failover for MRA

[Brian Meade]

I would not use Expressway clustering and just have 2 different C/E pairs
with different SRV Weights/Priorities instead.

On Mon, Jul 6, 2020 at 3:17 AM Gerence Guan  wrote:

> Hi Everyone.
>
> I was googling the answer for MRA failover and found this maillist.
> Got a similar setup as Jonathan's environment.
> Having a pair of expressway C&E in primary DC, and planning to setup
> another pair of expressway C&E in the DR site. All MRA should go via
> primary DC, only use DR site when primary is down.
>
> Can I achieve this with different priorities in SRV? Anyone tested  or
> make it working?
>
> Best Regards,
> Guan
>
> >>>* On Jan 28, 2020, at 8:49 PM, Jonathan Charles  >>>> wrote:
> *>>* We have two pairs of Expressway clusters (C/E) at two different
> *>>>* locations (primary and DR)...
> *>>* The cluster is up, however, we want to make sure that we are in
> *>>>* Active/Standby.
> *>>* Currently, we have one of our SRV records for collab-edge set at 5 
> (the
> *>>>* backup is at 10) with the same weight.
> *>>* The clustering guide says we should set the priority and weight on 
> both
> *>>>* SRV records the same, which will cause half of the registrations to go 
> to
> *>>>* the DR site. It is far away and has less capability.
> *>>* How do we:
> *>>* 1 - Make sure the primary site handles all MRA registrations and the 
> DR
> *>>>* site is only used when the primary is down.
> *>>>* 2 = Make sure failover occurs automatically... currently Jabber users
> *>>>* have to log out and back in to connect to the DR site.
> *>>>
>
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Re: [cisco-voip] New to Social Miner 11.6

[Brian Meade]

For video, I've been pushing customers to utilize the Webex Browser SDK or
the Gadget SDK to just call a SIP URI that rings in to the contact center.
Once an agent accepts the call, video will be negotiated.  Agents can use
any video endpoint to answer the call such as Jabber.

https://developer.webex.com/docs/sdks/browser
https://developer.webex.com/docs/widgets



On Fri, Jul 3, 2020 at 3:35 AM 0703Manjunath  wrote:

>
> Hi All,
>
> Looking for inputs since im new to cisco social miner deployment.
>
> We are going to deploy social miner 11.6.1  to integrate with existing
> cisco PCCE (11.6.1) setup  video & chat features.
>
> In this new deployment , customer is involving vendor to enable video
> calling features. If someone has experience in similar such deployment with
> video , i'm looking for guidance in configuring social miner part of
> configuration.
>
> Any inputs or pointer would be really helpful & appreciated
>
>
> Thank you
>
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Re: [cisco-voip] Creating Jabber for non-existent phones

[Brian Meade]

Fiddler has been helping me out a lot lately for these!  I had to do the
same to figure out how the Webex Device Connector actually works under the
hood.

On Thu, Jul 2, 2020 at 12:58 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> You tease!
>
> Just kidding, thanks for sharing this.  Did you just get this from the
> logs then?  Or some other dark magic?
>
> On Wed, Jul 1, 2020 at 11:02 PM Brian Meade  wrote:
>
>> Lelio,
>>
>> I'm actually starting to work on creating some Python scripts to allow
>> bulk using the Quick User/Phone Add feature such as using it to add Jabber
>> devices for all users very quickly.  I'm also going to use this for bulk
>> deploying Remote Destination Profiles/Device Profiles.
>>
>> I'm not sure if I'll share the full code or keep that internal to ePlus
>> but I'll share some of my findings around how these internal/non-documented
>> API's seem to be working behind the scenes.
>>
>> Create a DN with a specific line template:
>>
>> POST https:///ucmadmin/directorynumber/addDn/ HTTP/1.1
>>
>> Body:
>>
>>
>> {"dnorpattern":"88","universallinetemplate":"3a63b9c6-e867-4a37-82ca-9f5ad55d515c"}
>>
>> 200 OK response will contain the PKID of the new Directory Number.
>>
>> Set Primary Extension.  Replace PKID after /enduser/ with PKID of the end
>> user, fknumplan/sortorder is the important part.
>> This one may be easier to set via AXL API instead.
>>
>> PUT https:///ucmadmin/enduser/98735c00-12d6-6e16-f985-7778d05806b1
>> HTTP/1.1
>>
>> Body:
>>
>>
>> {"externData":{"groupAssociations":[],"extensions":[{"pkid":"","fknumplan":"f8cfccf4-a37a-37f9-3b68-e3fa71eb5522","sortorder":1,"lineDirectoryURI":{"fknumplan":"f8cfccf4-a37a-37f9-3b68-e3fa71eb5522","directoryuri":"","fkroutepartition":"","isprimary":false}}]},"credentialInfo":{"useDefaultCredential":false,"password":null,"passwordConfirm":null,"pin":null,"pinConfirm":null},"pkid":"98735c00-12d6-6e16-f985-7778d05806b1","firstname":"Brian","middlename":"","displayname":"","lastname":"Meade","userid":"bmeade-test","fkdirectorypluginconfig":null,"fkfeaturegrouptemplate":"41128559-bd7b-93cc-1166-01acf5b5bd4d","fkucuserprofile":"b2d3d9d0-a6bd-0136-d54f-bfee73f3ed74","userRank":"1","directoryuri":"","telephonenumber":"","mailid":"","manager":"","department":""}
>>
>>
>>
>> Find a phone to move to a user.  Replace PKID in POST URI with PKID of
>> end user.  Device list shows PKID of device you want to move:
>>
>> POST 
>> https:///ucmadmin/enduser/movePhones/98735c00-12d6-6e16-f985-7778d05806b1
>> HTTP/1.1
>>
>> Body:
>>
>> {"devicelist":[{"pkid":"3784d3e8-62c9-4b26-bfaa-1e7c741511bd"}]}
>>
>>
>>
>> Add a new phone for a user.  Replace PKID in POST with PKID of end user.
>> Replace device template PKID with the device template you want to use.
>> tkproduct is the model number from the typeproduct table.  isProfile is
>> used to say it's a device profile for extension mobility:
>>
>> POST 
>> https:///ucmadmin/enduser/addPhone/98735c00-12d6-6e16-f985-7778d05806b1
>> HTTP/1.1
>>
>> Body:
>>
>>
>> {"tkproduct":"30041","tkdeviceprotocol":"0","name":"SEPBMEADE","fkcommondevicetemplate":"580497c1-15e1-4e27-91ef-6f1e00f2e417","isprofile":"f","moduleCount":0}
>>
>>
>>
>> On Wed, Jul 1, 2020 at 2:06 PM Lelio Fulgenzi  wrote:
>>
>>>
>>> Hello all. Looking for feedback and opinions and caveats.
>>>
>>> Right now, we’re deploying Jabber only to those with phones/DNs. But, we
>>> need to start deploying Jabber for those individuals without phones/DNs.
>>>
>>> Our SOPs include using Quick Add feature. (Thanks a million time Brian
>>> Meade for the pointer).
>>>
>>> My choices so far, to address Jabber for new those without phones:
>>>
>>> (a) Create a fake hardware phone first. This has many benefits, namely,
>>> all SOPs remain the same. Hardware phone would be deleted afterwards.
>>>
>>> (b) Use Directory Number admin page to create/update a DN first, then
>>> use Quick Add page to assign DN to user accordingly and then click manage
>>> devices and follow remaining SOP steps.
>>>
>>> (c) create line templates and use those when creating new extensions
>>> under quick add. The issue with this is we have so many combinations, I’d
>>> need a lot of templates.
>>>
>>> I’m leaning towards (b), since it gives me the best of both worlds.
>>>
>>> Thoughts?
>>>
>>> Lelio
>>>
>>> Sent from my iPhone
>>> ___
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>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
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Re: [cisco-voip] Creating Jabber for non-existent phones

[Brian Meade]

Lelio,

I'm actually starting to work on creating some Python scripts to allow bulk
using the Quick User/Phone Add feature such as using it to add Jabber
devices for all users very quickly.  I'm also going to use this for bulk
deploying Remote Destination Profiles/Device Profiles.

I'm not sure if I'll share the full code or keep that internal to ePlus but
I'll share some of my findings around how these internal/non-documented
API's seem to be working behind the scenes.

Create a DN with a specific line template:

POST https:///ucmadmin/directorynumber/addDn/ HTTP/1.1

Body:

{"dnorpattern":"88","universallinetemplate":"3a63b9c6-e867-4a37-82ca-9f5ad55d515c"}

200 OK response will contain the PKID of the new Directory Number.

Set Primary Extension.  Replace PKID after /enduser/ with PKID of the end
user, fknumplan/sortorder is the important part.
This one may be easier to set via AXL API instead.

PUT https:///ucmadmin/enduser/98735c00-12d6-6e16-f985-7778d05806b1
HTTP/1.1

Body:

{"externData":{"groupAssociations":[],"extensions":[{"pkid":"","fknumplan":"f8cfccf4-a37a-37f9-3b68-e3fa71eb5522","sortorder":1,"lineDirectoryURI":{"fknumplan":"f8cfccf4-a37a-37f9-3b68-e3fa71eb5522","directoryuri":"","fkroutepartition":"","isprimary":false}}]},"credentialInfo":{"useDefaultCredential":false,"password":null,"passwordConfirm":null,"pin":null,"pinConfirm":null},"pkid":"98735c00-12d6-6e16-f985-7778d05806b1","firstname":"Brian","middlename":"","displayname":"","lastname":"Meade","userid":"bmeade-test","fkdirectorypluginconfig":null,"fkfeaturegrouptemplate":"41128559-bd7b-93cc-1166-01acf5b5bd4d","fkucuserprofile":"b2d3d9d0-a6bd-0136-d54f-bfee73f3ed74","userRank":"1","directoryuri":"","telephonenumber":"","mailid":"","manager":"","department":""}



Find a phone to move to a user.  Replace PKID in POST URI with PKID of end
user.  Device list shows PKID of device you want to move:

POST 
https:///ucmadmin/enduser/movePhones/98735c00-12d6-6e16-f985-7778d05806b1
HTTP/1.1

Body:

{"devicelist":[{"pkid":"3784d3e8-62c9-4b26-bfaa-1e7c741511bd"}]}



Add a new phone for a user.  Replace PKID in POST with PKID of end user.
Replace device template PKID with the device template you want to use.
tkproduct is the model number from the typeproduct table.  isProfile is
used to say it's a device profile for extension mobility:

POST 
https:///ucmadmin/enduser/addPhone/98735c00-12d6-6e16-f985-7778d05806b1
HTTP/1.1

Body:

{"tkproduct":"30041","tkdeviceprotocol":"0","name":"SEPBMEADE","fkcommondevicetemplate":"580497c1-15e1-4e27-91ef-6f1e00f2e417","isprofile":"f","moduleCount":0}



On Wed, Jul 1, 2020 at 2:06 PM Lelio Fulgenzi  wrote:

>
> Hello all. Looking for feedback and opinions and caveats.
>
> Right now, we’re deploying Jabber only to those with phones/DNs. But, we
> need to start deploying Jabber for those individuals without phones/DNs.
>
> Our SOPs include using Quick Add feature. (Thanks a million time Brian
> Meade for the pointer).
>
> My choices so far, to address Jabber for new those without phones:
>
> (a) Create a fake hardware phone first. This has many benefits, namely,
> all SOPs remain the same. Hardware phone would be deleted afterwards.
>
> (b) Use Directory Number admin page to create/update a DN first, then use
> Quick Add page to assign DN to user accordingly and then click manage
> devices and follow remaining SOP steps.
>
> (c) create line templates and use those when creating new extensions under
> quick add. The issue with this is we have so many combinations, I’d need a
> lot of templates.
>
> I’m leaning towards (b), since it gives me the best of both worlds.
>
> Thoughts?
>
> Lelio
>
> 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] CUBE Config Dial Peers

[Brian Meade]

100
>  destination dpg 1200
>  dtmf-relay sip-kpml rtp-nte ; we support both in- and out-of-band
> internally and cube interworks dtmf
>  codec g711ulaw
>  ip qos dscp cs3 signaling
>  no vad
> !
> dial-peer voice 2200 voip
>  description Outgoing CUCM Call Leg
>  session protocol sipv2
>  session server-group 2200
>  destination-pattern ABC123
>  voice-class sip options-keepalive profile 2200
>  dtmf-relay sip-kpml rtp-nte
>  codec g711ulaw
>  ip qos dscp cs3 signaling
>  no vad
> !
> ! a little something extra here at the end
> alias exec attra show call active voice | in
> PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
> alias exec attrh show call history voice | in
> PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
>
> On Fri, Jun 12, 2020 at 6:36 PM Brian Meade  wrote:
>
>> Anthony,
>>
>> Thanks for the feedback.  I'll definitely take a look at yours as well.
>>
>> Here's some answers on mine:
>> 1. While I like that you can give a uri profile a name like ISP, I much
>> prefer to stick with numbers, since for most things, you must use numbers
>> when naming, so this keeps it consistent.
>> So I usually replace this with the name of the actual SIP carrier.  This
>> seems to make it easier for customers to understand but I understand so
>> many other things are number tags only.
>> 2. I don't specify the port in my server groups, since 5060 is default,
>> but I can see how it might help be more explicit for some people
>> Yea, I've never tried it without specifying the port.  I've got a lot of
>> SIP carriers with weird SIP ports so making it stand out in the template
>> helps to know where to change this.
>> 3. Speaking of being explicit though, if that is your intention, I would
>> recommend pref 1 and pref 2, instead of implied pref 0 and pref 1
>> That's a good idea.  I actually exported this from a customer not
>> realizing what it looks like when I use pref 0 and 1.  Making it 1 and 2
>> would make that look cleaner.
>> 4. Why didn't you should your translation profiles and rules too?
>> These seem to be so customer/SIP carrier specific that I didn't think it
>> was worth it.  My most recent one had 80 rules in it because the carrier
>> really cares about 10-digit/11-digit calling for the local area code.  So
>> we actually had to split it up for different NPA-NXX whether or not we
>> added a 1.
>> 5. I don't specify UDP as the transport, since it's the default, but
>> again, being explicit doesn't hurt anything
>> I also make UDP my default but it is nice to have it called out in a
>> template so people know where to change it if needed.
>> 6. I like the extra dtmf on there.  too many configs are limited to
>> rtp-nte only and mtps are being invoked for every call to UCCX as one
>> example
>> Yea, I always add both to make sure we never have to pull in an MTP.  I'm
>> not aware of a way to do this globally but would be nice.
>> 7. Why do you drop your fax rate down from 14400 to 9600 as a standard?
>> I might learn something here, as faxing is not my strongest area.
>> I'm always debugging faxing it seems like.  Disabling ECM and reducing
>> speed to 9600 has seemed to help a lot over the years.  It's slower but
>> seems to work more reliably with every source/destination fax device.  And
>> people don't expect their fax to send quickly anyways.
>> 8. Since you're doing DPGs, you don't need the destination-pattern .T
>> command on the outbound DPs.
>> It seems like IOS-XE will show a dial-peer as down and skip it if there
>> is no destination-pattern configured.  This looks to be called out as
>> explicitely required here even though it isn't used-
>> https://cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/multiple-outbound-dial-peer.html
>>
>> Using something like ABC123 for the destination-pattern may make more
>> sense to not confuse anyone.  Good call.
>> 9a. Why are you not doing sip options ping?  I would, and in which case
>> you need a voice class sip options-keepalive profile
>> <https://community.cisco.com/t5/telepresence-and-video/sip-options-ping-and-session-server-group-on-dial-peer/td-p/2994584>
>>  since
>> you're using server groups.
>> I've never been a fan of SIP Options ping.  I've always used SIP timers
>> for failover instead.  I guess I've had a few outages where waiting for
>> Options Ping to come back up after we fixed the underlying issue added
>> additional delay.  For monitoring purposes though, it probably is 

Re: [cisco-voip] CUBE Config Dial Peers

[Brian Meade]

; 6. I like the extra dtmf on there.  too many configs are limited to
> rtp-nte only and mtps are being invoked for every call to UCCX as one
> example
> 7. Why do you drop your fax rate down from 14400 to 9600 as a standard?  I
> might learn something here, as faxing is not my strongest area.
> 8. Since you're doing DPGs, you don't need the destination-pattern .T
> command on the outbound DPs.
> 9a. Why are you not doing sip options ping?  I would, and in which case
> you need a voice class sip options-keepalive profile
> <https://community.cisco.com/t5/telepresence-and-video/sip-options-ping-and-session-server-group-on-dial-peer/td-p/2994584>
> since you're using server groups.
> 9b. Also, if you do end up turning on options, you do in fact need a
> destination-pattern command, and in which case, since it's not being used
> for call routing, I just use like ABC123 as the pattern to ensure it never
> can be used, but also, mildly clear it's not supposed to be used
>
> I'll post a config as well, in a bit, and please feel free to
> comment/question mine.
>
>
>
>
> On Fri, Jun 12, 2020 at 3:20 PM Brian Meade  wrote:
>
>> I've been trying to make a standardized CUBE configuration using a lot of
>> the newer features like dial-peer groups.
>>
>> This is what I have now.  It's an inbound dial-peer for CUCM matching the
>> CUCM IP's via Via header.  Then an inbound dial-peer for the ISP.  Then an
>> outbound dial-peer to CUCM and an outbound dial-peer to the ISP.  If you
>> have more IP's for the ISP or CUCM, you can easily add them.
>> destination-pattern .T is not used at all due to using dial-peer group
>> matching.  This doesn't account for bindings that must be done per
>> dial-peer.  It also doesn't show translation-profiles/rules.
>>
>> This gives you 4 total dial-peers to match any number.
>>
>> If it comes in from CUCM, it will route to the SIP carrier.  If it comes
>> in from the SIP carrier, it will route to CUCM.
>>
>> voice class uri ISP sip
>>  host ipv4:8.8.8.8
>>
>> voice class uri CUCM sip
>>  host ipv4:192.168.100.100
>>  host ipv4:192.168.100.200
>>
>> voice class server-group 100
>>  ipv4 8.8.8.8 port 5060
>>
>> voice class server-group 200
>>  ipv4 192.168.100.100 port 5060
>>  ipv4 192.168.100.200 port 5060 preference 1
>>
>> voice class dpg 100
>>
>> voice class dpg 200
>>
>> dial-peer voice 100 voip
>>  description Incoming Dial-peer from ISP
>>  translation-profile incoming ISPInbound
>>  session protocol sipv2
>>  session transport udp
>>  destination dpg 200
>>  incoming uri via ISP
>>  voice-class codec 1
>>  dtmf-relay rtp-nte sip-kpml
>>  fax-relay ecm disable
>>  fax rate 9600
>>
>> dial-peer voice 200 voip
>>  description Incoming Dial-peer from CUCM
>>  session protocol sipv2
>>  destination dpg 100
>>  incoming uri via CUCM
>>  voice-class codec 1
>>  dtmf-relay rtp-nte sip-kpml
>>  fax-relay ecm disable
>>  fax rate 9600
>>
>> dial-peer voice 300 voip
>>  description Outbound to ISP
>>  translation-profile outgoing ISPOutbound
>>  destination-pattern .T
>>  session protocol sipv2
>>  session transport udp
>>  session server-group 100
>>  voice-class codec 1
>>  dtmf-relay rtp-nte sip-kpml
>>  fax-relay ecm disable
>>  fax rate 9600
>>
>> dial-peer voice 400 voip
>>  description Outbound to CUCM
>>  destination-pattern .T
>>  session protocol sipv2
>>  session server-group 200
>>  voice-class codec 1
>>  dtmf-relay rtp-nte sip-kpml
>>  fax-relay ecm disable
>>  fax rate 9600
>>
>> voice class dpg 100
>>  dial-peer 300
>>
>> voice class dpg 200
>>  dial-peer 400
>>
>> On Fri, Jun 12, 2020 at 3:12 PM JASON BURWELL via cisco-voip <
>> cisco-voip@puck.nether.net> wrote:
>>
>>> Does anyone have a good, straightforward reference doc to configuring
>>> CUBE dial peers? I have what I would have thought should be a fairly basic
>>> config but I’m having trouble getting everything to work properly. I’ve had
>>> some assistance but it seems like a whole lot of configuration to do what
>>> little I really need to do. Basically, I just need to send whatever comes
>>> from CUCM- either 10, 11 or 3 digits in the SIP invite for outbound and for
>>> inbound calls the provider sends me 10 digits in the invite, I just need to
>>> strip off the first 6 and send the last 4 to CUCM to route. I have a lot of
>>> adding and stripping digits going on between CUCM and CUBE to make this
>>> work. Just trying to find reference docs to see if any of this can be
>>> cleaned up. Thanks
>>> ___
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>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
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>>
>
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Re: [cisco-voip] CUBE Config Dial Peers

[Brian Meade]

I've been trying to make a standardized CUBE configuration using a lot of
the newer features like dial-peer groups.

This is what I have now.  It's an inbound dial-peer for CUCM matching the
CUCM IP's via Via header.  Then an inbound dial-peer for the ISP.  Then an
outbound dial-peer to CUCM and an outbound dial-peer to the ISP.  If you
have more IP's for the ISP or CUCM, you can easily add them.
destination-pattern .T is not used at all due to using dial-peer group
matching.  This doesn't account for bindings that must be done per
dial-peer.  It also doesn't show translation-profiles/rules.

This gives you 4 total dial-peers to match any number.

If it comes in from CUCM, it will route to the SIP carrier.  If it comes in
from the SIP carrier, it will route to CUCM.

voice class uri ISP sip
 host ipv4:8.8.8.8

voice class uri CUCM sip
 host ipv4:192.168.100.100
 host ipv4:192.168.100.200

voice class server-group 100
 ipv4 8.8.8.8 port 5060

voice class server-group 200
 ipv4 192.168.100.100 port 5060
 ipv4 192.168.100.200 port 5060 preference 1

voice class dpg 100

voice class dpg 200

dial-peer voice 100 voip
 description Incoming Dial-peer from ISP
 translation-profile incoming ISPInbound
 session protocol sipv2
 session transport udp
 destination dpg 200
 incoming uri via ISP
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax rate 9600

dial-peer voice 200 voip
 description Incoming Dial-peer from CUCM
 session protocol sipv2
 destination dpg 100
 incoming uri via CUCM
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax rate 9600

dial-peer voice 300 voip
 description Outbound to ISP
 translation-profile outgoing ISPOutbound
 destination-pattern .T
 session protocol sipv2
 session transport udp
 session server-group 100
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax rate 9600

dial-peer voice 400 voip
 description Outbound to CUCM
 destination-pattern .T
 session protocol sipv2
 session server-group 200
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax rate 9600

voice class dpg 100
 dial-peer 300

voice class dpg 200
 dial-peer 400

On Fri, Jun 12, 2020 at 3:12 PM JASON BURWELL via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> Does anyone have a good, straightforward reference doc to configuring CUBE
> dial peers? I have what I would have thought should be a fairly basic
> config but I’m having trouble getting everything to work properly. I’ve had
> some assistance but it seems like a whole lot of configuration to do what
> little I really need to do. Basically, I just need to send whatever comes
> from CUCM- either 10, 11 or 3 digits in the SIP invite for outbound and for
> inbound calls the provider sends me 10 digits in the invite, I just need to
> strip off the first 6 and send the last 4 to CUCM to route. I have a lot of
> adding and stripping digits going on between CUCM and CUBE to make this
> work. Just trying to find reference docs to see if any of this can be
> cleaned up. Thanks
> ___
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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 Flex Licensing

[Brian Meade]

rent from non flex licensing.
>
>
>
> Standard is inbound agent licensing essentially
>
>
>
> Premium is supervisor licensing,   email/chat agents, outbound campaign
> licensing.
>
>
>
> 2 CTI ports per agent/license.
>
>
>
> Admin still works no specific license needed as long as admin isn’t also
> supervisor/agent, HA is included, outside of the 3 features above, it’s
> like perpetual premium with SQL, etc included.
>
>
>
> See here for specifics:
>
>
>
>
> https://www.cisco.com/c/en/us/products/collateral/unified-communications/cisco-collaboration-flex-plan/datasheet-c78-741220.html
> Table 8.
>
>
>
>
>
> License enforcement is only in UCCX 12.5. Older versions don’t know and
> you end up with Perpetual Premium with HA feature set but with a license
> that expires at the end of your contract term.
>
>
>
> Suffice it to say if you don’t need 12.5 features you could ride the gravy
> train for a while.
>
>
>
> Licensing is still concurrent users.
>
>
>
> There are grace periods so if you need to test something you can make the
> admin a supervisor or something w/o breakage, just remember to remove later.
>
>
>
> If your customer has on-prem premium, with perpetual trade-in credits you
> are close to the cost of SWSS and should probably try and get them to move.
>
>
>
>
>
>
>
>
>
> *Matthew Loraditch**​*
>
> *Sr. Network Engineer*
>
> p: *443.541.1518* <443.541.1518>
>
> w: *www.heliontechnologies.com* <http://www.heliontechnologies.com/>
>
>  |
>
> e: *mloradi...@heliontechnologies.com* 
>
> [image: Helion Technologies] <http://www.heliontechnologies.com/>
>
> [image: Facebook] <https://facebook.com/heliontech>
>
> [image: Twitter] <https://twitter.com/heliontech>
>
> [image: LinkedIn] <https://www.linkedin.com/company/helion-technologies>
>
> *From:* cisco-voip  *On Behalf Of *Anthony
> Holloway
> *Sent:* Monday, May 11, 2020 5:21 PM
> *To:* Pawlowski, Adam 
> *Cc:* Cisco VoIP Group 
> *Subject:* Re: [cisco-voip] UCCX Flex Licensing
>
>
>
> [EXTERNAL]
>
>
>
> But seriously, a premium license to administer the system?  Does this
> include the appadministrator account too?  Do you have first hand
> experience with it?
>
>
>
> On Mon, May 11, 2020 at 4:16 PM Pawlowski, Adam  wrote:
>
> This was the information I heard as well, and the purchase quantities are
> based on feature utilization and concurrency.
>
>
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Brian
> Meade
> *Sent:* Monday, May 11, 2020 5:07 PM
> *To:* Anthony Holloway 
> *Cc:* Cisco VoIP Group 
> *Subject:* Re: [cisco-voip] UCCX Flex Licensing
>
>
>
> Pretty sure when buying as A-Flex-CC that it always just gives you Premium
> licensing on the CCX side.  Had this cause an issue with a customer that
> was staying on Enhanced for the extra CTI ports for many years.
>
>
>
> On Mon, May 11, 2020 at 4:23 PM Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> All,
>
>
>
> Anyone already deal with this themselves?  I am reading/being told
> something I cannot swallow as the truth, because it seems so ridiculous.
>
>
>
> I am being told that you need a Premium license to even login as a
> Supervisor at all.  Like, not for extra functionality (silent monitoring),
> but just as a basic license requirement to even sign in.
>
>
>
> Also, I am being told a Premium license is required for Administrative
> users too.  Like, even the app admin account.  So what, completing a fresh
> install now requires a Premium license?
>
>
>
> Are either of these true?  Can you confirm from your own tests that this
> is in fact how Flex works in UCCX on-prem?
>
> ___
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>
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Re: [cisco-voip] SX80 Camera Tracking and Face Masks

[Brian Meade]

Try drawing a mouth on the mask and see if that helps.

On Tue, Jun 9, 2020 at 1:31 PM ROZA, Ariel 
wrote:

> Hi Guys,
>
>
>
> Did anyone experience problems with speaker tracking while people using
> facemasks?
>
> I received reports from people that verified lack of tracking while using
> a mask, and that it works properly when they take them off.
>
> This is not weird, and somehow expected. But wanted to know if there´s any
> kind of solution/fix/workaround.
>
> Tried looking for bugs in the BST, but does not show anything for SX80
>
>
>
> Regards,
>
>
>
>
>
> *Ariel Roza*
> *Support & Maintenance* *Engineer** | Latam*
>
> t: +54 11 5282-0458 / c: +54 11 5017-4417 / webex:
> https://logicalis-la.webex.com/join/ariel.roza
>
> Av. Belgrano 955 – Piso 20 – CABA – Argentina – C1092AAJ
>
> www.la.logicalis.com
>
> *Business **and technology working as one*
>
> 
> 
> 
> 
> 
>
> Logicalis Argentina S.A. solo puede ser obligado por sus representantes
> legales conforme los límites establecidos en el acto constitutivo y la
> legislación en vigor.
>
> El contenido del presente correo electrónico e inclusive sus anexos
> contienen información confidencial.
>
> El mismo no puede ser divulgado y/o utilizado por cualquiera otro distinto
> al destinatario, ni puede ser copiado de cualquier forma
>
>
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Re: [cisco-voip] Third Party CDR Analysis

[Brian Meade]

Variphy seems to be my favorite so far.  I like it much better than the ISI
offering.

On Wed, Jun 3, 2020 at 4:25 PM UC Penguin  wrote:

> I’m curious what third party CDR Analysis software is commonly used today
> and pros/cons of each?
>
> Looking for something friendly for non-Engineers to run reports.
>
> Thanks in advance
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Re: [cisco-voip] UCCX Flex Licensing

[Brian Meade]

It seems to be the same license consumption as before from what I've seen
but just they only give you Premium licenses now.

On Mon, May 11, 2020 at 5:21 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> But seriously, a premium license to administer the system?  Does this
> include the appadministrator account too?  Do you have first hand
> experience with it?
>
> On Mon, May 11, 2020 at 4:16 PM Pawlowski, Adam  wrote:
>
>> This was the information I heard as well, and the purchase quantities are
>> based on feature utilization and concurrency.
>>
>>
>>
>>
>>
>>
>>
>> *From:* cisco-voip  *On Behalf Of *Brian
>> Meade
>> *Sent:* Monday, May 11, 2020 5:07 PM
>> *To:* Anthony Holloway 
>> *Cc:* Cisco VoIP Group 
>> *Subject:* Re: [cisco-voip] UCCX Flex Licensing
>>
>>
>>
>> Pretty sure when buying as A-Flex-CC that it always just gives you
>> Premium licensing on the CCX side.  Had this cause an issue with a customer
>> that was staying on Enhanced for the extra CTI ports for many years.
>>
>>
>>
>> On Mon, May 11, 2020 at 4:23 PM Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>> All,
>>
>>
>>
>> Anyone already deal with this themselves?  I am reading/being told
>> something I cannot swallow as the truth, because it seems so ridiculous.
>>
>>
>>
>> I am being told that you need a Premium license to even login as a
>> Supervisor at all.  Like, not for extra functionality (silent monitoring),
>> but just as a basic license requirement to even sign in.
>>
>>
>>
>> Also, I am being told a Premium license is required for Administrative
>> users too.  Like, even the app admin account.  So what, completing a fresh
>> install now requires a Premium license?
>>
>>
>>
>> Are either of these true?  Can you confirm from your own tests that this
>> is in fact how Flex works in UCCX on-prem?
>>
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
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Re: [cisco-voip] UCCX Flex Licensing

[Brian Meade]

Pretty sure when buying as A-Flex-CC that it always just gives you Premium
licensing on the CCX side.  Had this cause an issue with a customer that
was staying on Enhanced for the extra CTI ports for many years.

On Mon, May 11, 2020 at 4:23 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> All,
>
> Anyone already deal with this themselves?  I am reading/being told
> something I cannot swallow as the truth, because it seems so ridiculous.
>
> I am being told that you need a Premium license to even login as a
> Supervisor at all.  Like, not for extra functionality (silent monitoring),
> but just as a basic license requirement to even sign in.
>
> Also, I am being told a Premium license is required for Administrative
> users too.  Like, even the app admin account.  So what, completing a fresh
> install now requires a Premium license?
>
> Are either of these true?  Can you confirm from your own tests that this
> is in fact how Flex works in UCCX on-prem?
> ___
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>
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Re: [cisco-voip] UCCX Callback Call Redirect Adding Diversion Header

[Brian Meade]

Ended up switching to a Consult Transfer step instead.  Now it's hold music
rather than true ringback but works pretty well.  I can change the hold
music to fake ringback if it becomes an issue.

On Tue, Apr 21, 2020 at 3:39 PM Brian Meade  wrote:

> I think that should work! I can strip it on the dial-peer side to remove
> the Diversion header for those new dial-peers.
>
> On Tue, Apr 21, 2020 at 3:01 PM Kent Roberts  wrote:
>
>> Can you prepend an access code on it and use a different trunk port/dial
>> peer and strip it that way?
>>
>>
>> Kent
>>
>> On Apr 21, 2020, at 11:41, Brian Meade  wrote:
>>
>> 
>> Hey everyone,
>>
>> I'm working on a UCCX Callback deployment which seems to be a weekly
>> thing lately.
>>
>> But in this customer's environment, I'm noticing the outbound Call
>> Redirect Step is actually causing a diversion header to be sent out the
>> Customer SIP Trunk on the Outgoing Invite.  The phone number is actually
>> the destination number we're doing the callback to.
>>
>> Diversion: > >;reason=deflection;privacy=off;screen=yes
>>
>> They've got Redirecting number outbound checked on the SIP Trunk for
>> call-forwarding/SNR to preserve the original calling number correctly.
>>
>> The carrier is not accepting the call because of the Diversion header and
>> the agent gets a message from the carrier "call cannot be completed as
>> dialed".
>>
>> I've tried sending the call through a translation pattern but that didn't
>> seem to strip out the redirecting number.
>>
>> I'm curious if anyone has seen this with a callback script because I
>> don't remember a Diversion header being normal but maybe I've just always
>> missed it.
>>
>> I can't turn off the Diversion header completely without breaking
>> call-forwarding.
>>
>> Only thing I can think of would be a LUA script to strip the Diversion
>> header if the Calling Number is for an agent extension since those
>> shouldn't be ever forwarded anyways.
>>
>> Anyone got any ideas on how to strip off this Diversion header just for
>> these callback calls?
>>
>> Thanks,
>> Brian Meade
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
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Re: [cisco-voip] UCCX Callback Call Redirect Adding Diversion Header

[Brian Meade]

I think that should work! I can strip it on the dial-peer side to remove
the Diversion header for those new dial-peers.

On Tue, Apr 21, 2020 at 3:01 PM Kent Roberts  wrote:

> Can you prepend an access code on it and use a different trunk port/dial
> peer and strip it that way?
>
>
> Kent
>
> On Apr 21, 2020, at 11:41, Brian Meade  wrote:
>
> 
> Hey everyone,
>
> I'm working on a UCCX Callback deployment which seems to be a weekly thing
> lately.
>
> But in this customer's environment, I'm noticing the outbound Call
> Redirect Step is actually causing a diversion header to be sent out the
> Customer SIP Trunk on the Outgoing Invite.  The phone number is actually
> the destination number we're doing the callback to.
>
> Diversion:  >;reason=deflection;privacy=off;screen=yes
>
> They've got Redirecting number outbound checked on the SIP Trunk for
> call-forwarding/SNR to preserve the original calling number correctly.
>
> The carrier is not accepting the call because of the Diversion header and
> the agent gets a message from the carrier "call cannot be completed as
> dialed".
>
> I've tried sending the call through a translation pattern but that didn't
> seem to strip out the redirecting number.
>
> I'm curious if anyone has seen this with a callback script because I don't
> remember a Diversion header being normal but maybe I've just always missed
> it.
>
> I can't turn off the Diversion header completely without breaking
> call-forwarding.
>
> Only thing I can think of would be a LUA script to strip the Diversion
> header if the Calling Number is for an agent extension since those
> shouldn't be ever forwarded anyways.
>
> Anyone got any ideas on how to strip off this Diversion header just for
> these callback calls?
>
> Thanks,
> Brian Meade
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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[cisco-voip] UCCX Callback Call Redirect Adding Diversion Header

[Brian Meade]

Hey everyone,

I'm working on a UCCX Callback deployment which seems to be a weekly thing
lately.

But in this customer's environment, I'm noticing the outbound Call Redirect
Step is actually causing a diversion header to be sent out the Customer SIP
Trunk on the Outgoing Invite.  The phone number is actually the destination
number we're doing the callback to.

Diversion: ;reason=deflection;privacy=off;screen=yes

They've got Redirecting number outbound checked on the SIP Trunk for
call-forwarding/SNR to preserve the original calling number correctly.

The carrier is not accepting the call because of the Diversion header and
the agent gets a message from the carrier "call cannot be completed as
dialed".

I've tried sending the call through a translation pattern but that didn't
seem to strip out the redirecting number.

I'm curious if anyone has seen this with a callback script because I don't
remember a Diversion header being normal but maybe I've just always missed
it.

I can't turn off the Diversion header completely without breaking
call-forwarding.

Only thing I can think of would be a LUA script to strip the Diversion
header if the Calling Number is for an agent extension since those
shouldn't be ever forwarded anyways.

Anyone got any ideas on how to strip off this Diversion header just for
these callback calls?

Thanks,
Brian Meade
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Re: [cisco-voip] CUCM Interop with MS Teams

[Brian Meade]

It should be pretty similar to most SIP CUBE configurations.  The main
difference is you'll need to do Mutual TLS SIP with Microsoft so you'll
need a public certificate on the CUBE that the SIP-UA can use.  You also
need a public IP on the CUBE and corresponding DNS records that Microsoft
side can reference.

I don't have a sanitized config I can share.  How far have you gotten with
your configuration?

On Wed, Apr 8, 2020 at 11:25 AM Carlos G Mendioroz  wrote:

> Brian,
> would you care to comment "the right configuration" part of this ?
> I've tried to setup it and failed misserably :(
>
> TIA,
> -Carlos
>
> Brian Meade @ 24/03/2020 20:27 -0300 dixit:
> > Cisco CUBE can be used for this as well.
> >
> > Official documentation should be coming in the near future but you can
> > do it now with the right configuration.
> >
> > On Tue, Mar 24, 2020 at 6:07 PM UC Penguin  > <mailto:gen...@ucpenguin.com>> wrote:
> >
> > Does anyone have an experience with setting up interop between
> > CUCM/Microsoft Teams with Direct Routing?
> >
> > If so what (supported) SBC did you use and what if any issues did
> > you encounter?
> >
> > Thanks!
> > ___
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> > cisco-voip@puck.nether.net <mailto: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
> >
>
> --
> Carlos G Mendioroz  
>
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Re: [cisco-voip] FAX into a Cisco shop

[Brian Meade]

OpenText (owner of RightFax) just bought xMedius.  I've only ever heard
good things about xMedius (especially the cloud offering).  I've seen a lot
of mixed opinions on RightFax.

I've also seen GFI FaxMaker several times and also had mixed reviews.

On Mon, Apr 6, 2020 at 11:43 AM Terry Oakley via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> Any suggestions for a software based FAX service/application that works
> seamlessly with CUCM and O365We are hoping to move our legacy FAX
> devices into the 20th century during this 21st century pandemic and push
> to have our FAX services virtual or FAX to email.
>
> I have heard of RightFax, XmediusFax, OpenText but not sure which would be
> the best fit and best to integrate with.
>
>
>
> Thanks
>
>
>
> Terry
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Re: [cisco-voip] [EXTERNAL] Re: Cost-Effective Public Certificate Authority for CUCM certificates

[Brian Meade]

In this case, we're doing public certificates internally as well for CUCM
Tomcat, Unity Connection Tomcat, UCCX Tomcat, and IM&P CUP-XMPP.

Adding the multiple presence domains is pretty easy on the IM&P side and it
will automatically add SAN's for those domains in the CSR.

Expressway-E will also automatically add all domains to the CSR.

On Mon, Mar 30, 2020 at 4:07 PM Jonatan Quezada <
jonatan.quez...@chemeketa.edu> wrote:

> Brian, How challenging was it to do the jabber on all three domains?
>
> Where do you need the multiDomain cert, on the VCS-edge connector right?
> Im looking to see what it would take to get this going for our remote
> workers even though it seems
> like there are few things to make sure are in place first.
>
> for so far its the :
>
> certs for dual domain- how
> provision jabber users
>
>
> On Mon, Mar 30, 2020 at 12:28 PM Brian Meade  wrote:
>
>> I was originally going to go with that wildcard option but this customer
>> has 3 different presence domains to match their email domains which makes
>> the CUP-XMPP cert more complicated.
>>
>> This is my personal email so no access to InCommon certificates
>> unfortunately.
>>
>> On Mon, Mar 30, 2020 at 2:59 PM Matthew Ballard 
>> wrote:
>>
>>> We used to use DigiCert Wildcard which offers that (where you can issue
>>> multiple certificates with different private keys from the same wildcard
>>> cert/purchase).
>>>
>>>
>>>
>>> We switched to using InCommon certificates, which it looks like your
>>> University also subscribes to.  You should be able to get them internally
>>> from whomever licensed that there, as it’s a flat fee service for unlimited
>>> certificates.
>>>
>>>
>>>
>>> Matthew Ballard
>>>
>>> Director of Technology Infrastructure
>>>
>>> Information Systems
>>>
>>> Otis College of Art and Design
>>>
>>> mball...@otis.edu
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* cisco-voip  *On Behalf Of *Brian
>>> Meade
>>> *Sent:* Monday, March 30, 2020 11:42 AM
>>> *To:* cisco-voip voyp list 
>>> *Subject:* [cisco-voip] Cost-Effective Public Certificate Authority for
>>> CUCM certificates
>>>
>>>
>>>
>>> Does anyone know of any public certificate authorities that have cheaper
>>> multi-server SAN certificate options?  I had seen some in the past that let
>>> you buy a wildcard and then can submit CSR's against that still but having
>>> trouble finding that now.
>>>
>>>
>>>
>>> Trying to avoid buying 4 multi-server certificates to cover CUCM
>>> Tomcat/Unity Connection Tomcat/UCCX Tomcat/IM&P XMPP.
>>>
>> ___
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>>
>
>
> --
> During this time of remote work, There will be the need for connectivity
> to other devices such as a cell phone. If you require assistance forwarding
> your desk phone to a remote cell or message phone, please email with desk
> number and where we are forwarding calls. I can do these remotely.
>
> Johnny Q
> Voice Technology Analyst II
> Chemeketa Community College
> johnn...@chemeketa.edu
> Building 22 Room 130
> Work 5033995294
> Cell 5035769873
> FAX 5033995549
>
>
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Re: [cisco-voip] [EXTERNAL] Cost-Effective Public Certificate Authority for CUCM certificates

[Brian Meade]

Namecheap seems to be the cheapest option I've found from some quick
looking.  They seem to resell Comodo certificates but cheaper than Comodo
offers them.

On Mon, Mar 30, 2020 at 2:45 PM Jonatan Quezada <
jonatan.quez...@chemeketa.edu> wrote:

> Im totally looking to update all of mine I think we use digi-cert, pleasea
> let us know what you find out :)
> Cheers!
>
> On Mon, Mar 30, 2020 at 11:43 AM Brian Meade  wrote:
>
>> Does anyone know of any public certificate authorities that have cheaper
>> multi-server SAN certificate options?  I had seen some in the past that let
>> you buy a wildcard and then can submit CSR's against that still but having
>> trouble finding that now.
>>
>> Trying to avoid buying 4 multi-server certificates to cover CUCM
>> Tomcat/Unity Connection Tomcat/UCCX Tomcat/IM&P XMPP.
>> ___
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>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> --
> During this time of remote work, There will be the need for connectivity
> to other devices such as a cell phone. If you require assistance forwarding
> your desk phone to a remote cell or message phone, please email with desk
> number and where we are forwarding calls. I can do these remotely.
>
> Johnny Q
> Voice Technology Analyst II
> Chemeketa Community College
> johnn...@chemeketa.edu
> Building 22 Room 130
> Work 5033995294
> Cell 5035769873
> FAX 5033995549
>
>
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Re: [cisco-voip] Cost-Effective Public Certificate Authority for CUCM certificates

[Brian Meade]

I was originally going to go with that wildcard option but this customer
has 3 different presence domains to match their email domains which makes
the CUP-XMPP cert more complicated.

This is my personal email so no access to InCommon certificates
unfortunately.

On Mon, Mar 30, 2020 at 2:59 PM Matthew Ballard  wrote:

> We used to use DigiCert Wildcard which offers that (where you can issue
> multiple certificates with different private keys from the same wildcard
> cert/purchase).
>
>
>
> We switched to using InCommon certificates, which it looks like your
> University also subscribes to.  You should be able to get them internally
> from whomever licensed that there, as it’s a flat fee service for unlimited
> certificates.
>
>
>
> Matthew Ballard
>
> Director of Technology Infrastructure
>
> Information Systems
>
> Otis College of Art and Design
>
> mball...@otis.edu
>
>
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Brian
> Meade
> *Sent:* Monday, March 30, 2020 11:42 AM
> *To:* cisco-voip voyp list 
> *Subject:* [cisco-voip] Cost-Effective Public Certificate Authority for
> CUCM certificates
>
>
>
> Does anyone know of any public certificate authorities that have cheaper
> multi-server SAN certificate options?  I had seen some in the past that let
> you buy a wildcard and then can submit CSR's against that still but having
> trouble finding that now.
>
>
>
> Trying to avoid buying 4 multi-server certificates to cover CUCM
> Tomcat/Unity Connection Tomcat/UCCX Tomcat/IM&P XMPP.
>
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[cisco-voip] Cost-Effective Public Certificate Authority for CUCM certificates

[Brian Meade]

Does anyone know of any public certificate authorities that have cheaper
multi-server SAN certificate options?  I had seen some in the past that let
you buy a wildcard and then can submit CSR's against that still but having
trouble finding that now.

Trying to avoid buying 4 multi-server certificates to cover CUCM
Tomcat/Unity Connection Tomcat/UCCX Tomcat/IM&P XMPP.
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Re: [cisco-voip] CUCM Interop with MS Teams

[Brian Meade]

Cisco CUBE can be used for this as well.

Official documentation should be coming in the near future but you can do
it now with the right configuration.

On Tue, Mar 24, 2020 at 6:07 PM UC Penguin  wrote:

> Does anyone have an experience with setting up interop between
> CUCM/Microsoft Teams with Direct Routing?
>
> If so what (supported) SBC did you use and what if any issues did you
> encounter?
>
> Thanks!
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Re: [cisco-voip] Expressway MRA registrations

[Brian Meade]

Status->Unified Communications and can look at View Provisioning Sessions
link.

On Tue, Mar 10, 2020 at 11:33 PM SK  wrote:

> Hello,
>
> How can I find how many current registrations are active on the expressway
> for MRA?
>
> Thank you .
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Re: [cisco-voip] Join Webex Button on TOuch 10

[Brian Meade]

The other non-advertised thing is that Webex Teams and Jabber both join
Webex meetings as video calls as well so you're also stuck with that 200
limit if joining from those.

This should be changing with the Unified Client though.

On Thu, Feb 20, 2020 at 9:06 AM Lelio Fulgenzi  wrote:

> Just as an aside... this uses up a “video endpoint” peg count. So, at
> most, 200 attendants this way.
>
> Sent from my iPhone
>
> On Feb 20, 2020, at 7:36 AM, Pawlowski, Adam  wrote:
>
> I set up a dial pattern for this in our UCM, to match the dialing prefix
> set in Webex teams, which currently goes to our site.
>
>
>
> It’s not that great from audio only devices because the meet IVR just
> answers with “enter your meeting number” and then is silent.
>
>
>
> However – you can *start* a meeting this way very easily as you join as a
> video device as far as Webex is concerned.
>
>
>
> *From:* cisco-voip  *On Behalf Of *Brian
> Meade
> *Sent:* Wednesday, February 19, 2020 9:29 PM
> *To:* JASON BURWELL 
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Join Webex Button on TOuch 10
>
>
>
> You can dial any 9-digit meeting number now like 123456...@webex.com.
> Webex changed to make these non-site specific.  So it's easy to make an
> Extron/Crestron interface to enter a 9-digit meeting number than have it
> just add @webex.com and dial.
>
>
>
> The other thing you can do is just have a button for m...@webex.com which
> prompts you to enter the 9-digit meeting number via DTMF.
>
>
>
> On Tue, Feb 18, 2020 at 4:12 PM JASON BURWELL via cisco-voip <
> cisco-voip@puck.nether.net> wrote:
>
> I noticed with CE9.10 they have now added a “Join Webex” button that
> appears on the Touch 10 home screen that allows direct calling to a webex
> meeting without dialing the entire URI. This is a great feature but now I’m
> trying to figure out if anyone is having luck integrating this Quick
> connect to webex meeting feature in to room systems that are Crestron
> controlled and do not have a Touch 10 present. Sure would be nice to be
> able to be able to have this functionality from the Crestron control as
> well. My A/V vendor says they “are working on it” so it could be a while.
> Anyone had luck with this?
>
>
>
> Thanks Jason
>
> ___
> 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] [EXTERNAL] 7832 out of the box - needs to be reset to register

[Brian Meade]

We're starting to see more and more phones that somehow got an ITL on them
new from the factory.  Factory reset doesn't seem to always delete the ITL
either so deleting Security Settings specifically is usually required in
those cases.

On Tue, Feb 18, 2020 at 4:55 PM Lelio Fulgenzi  wrote:

> It was last updated 2017. So, I can’t imagine it’s a new batch.
>
>
>
> *From:* JASON BURWELL 
> *Sent:* Tuesday, February 18, 2020 4:49 PM
> *To:* Lelio Fulgenzi ; voyp list, cisco-voip (
> cisco-voip@puck.nether.net) 
> *Subject:* RE: [EXTERNAL] 7832 out of the box - needs to be reset to
> register
>
>
>
> I just got a 7832 New in the box today and had the exact thing happen.
> Never had this problem before on 7832s. Must be a bug with the default
> firmware they put on this batch.
>
>
>
> Jason
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Lelio
> Fulgenzi
> *Sent:* Tuesday, February 18, 2020 4:09 PM
> *To:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* [EXTERNAL] 7832 out of the box - needs to be reset to register
>
>
>
> *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.*
> * -- *
>
>
>
> Any particular reason that I have to reset all settings on a 7832 before
> it registers?
>
>
>
> Never seen that before.
>
>
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>
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Re: [cisco-voip] Official Cisco E911 Guidance

[Brian Meade]

The big issue with this document is the recommendation of adding a delay as
an option which seems to be a legal gray area.  I asked Cisco about this
and they aren't considering this a legal document and still suggest
checking with your own corporate legal team on guidance.

On Tue, Feb 18, 2020 at 3:04 PM Matthew Loraditch <
mloradi...@heliontechnologies.com> wrote:

> This was just published:
>
>
> https://blogs.cisco.com/collaboration/saying-yes-to-workplace-safety-how-ucm-customers-become-compliant-with-karis-law-and-ray-baums-act
>
>
>
>
> https://www.cisco.com/c/dam/en/us/products/collateral/unified-communications/unified-communications-manager-callmanager/q-and-a-c67-743415.pdf
>
>
>
>
>
>
>
>
>
>
>
>
>
> Matthew Loraditch​
> Sr. Network Engineer
> p: *443.541.1518* <443.541.1518>
> w: *www.heliontechnologies.com*   |
> e: *mloradi...@heliontechnologies.com* 
> [image: Helion Technologies] 
> [image: Facebook] 
> [image: Twitter] 
> [image: LinkedIn] 
> ___
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Re: [cisco-voip] Join Webex Button on TOuch 10

[Brian Meade]

You can dial any 9-digit meeting number now like 123456...@webex.com.
Webex changed to make these non-site specific.  So it's easy to make an
Extron/Crestron interface to enter a 9-digit meeting number than have it
just add @webex.com and dial.

The other thing you can do is just have a button for m...@webex.com which
prompts you to enter the 9-digit meeting number via DTMF.

On Tue, Feb 18, 2020 at 4:12 PM JASON BURWELL via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> I noticed with CE9.10 they have now added a “Join Webex” button that
> appears on the Touch 10 home screen that allows direct calling to a webex
> meeting without dialing the entire URI. This is a great feature but now I’m
> trying to figure out if anyone is having luck integrating this Quick
> connect to webex meeting feature in to room systems that are Crestron
> controlled and do not have a Touch 10 present. Sure would be nice to be
> able to be able to have this functionality from the Crestron control as
> well. My A/V vendor says they “are working on it” so it could be a while.
> Anyone had luck with this?
>
>
>
> 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] Field Notice from Cisco making Secure LDAP mandatory

[Brian Meade]

CUCM doesn't check the names, just that the chain is trusted.

On Sun, Feb 9, 2020 at 5:23 PM Matthew Loraditch <
mloradi...@heliontechnologies.com> wrote:

> Interesting. Our root cert is and has been loaded, but I’m still using
> just the IPs so normally that would make the handshake fail.
>
> Get Outlook for iOS 
>
> Matthew Loraditch​
> Sr. Network Engineer
> p: *443.541.1518* <443.541.1518>
> w: *www.heliontechnologies.com*   |
> e: *mloradi...@heliontechnologies.com* 
> [image: Helion Technologies] 
> [image: Facebook] 
> [image: Twitter] 
> [image: LinkedIn] 
> --
> *From:* Lelio Fulgenzi 
> *Sent:* Sunday, February 9, 2020 5:15:40 PM
> *To:* Matthew Loraditch 
> *Cc:* James Buchanan ; voyp list, cisco-voip (
> cisco-voip@puck.nether.net) 
> *Subject:* Re: [cisco-voip] Field Notice from Cisco making Secure LDAP
> mandatory
>
>
> [EXTERNAL]
>
>
> I couldn’t get secure ldap to work without loading the certificates from
> the AD servers. I also had more luck using the global catalog ports.
>
> Sent from my iPhone
>
> On Feb 9, 2020, at 5:05 PM, Matthew Loraditch <
> mloradi...@heliontechnologies.com> wrote:
>
> I was wondering if they were going to post anything as it’s very unclear
> if ldap over tls was the fix.
>
> Apparently (and amen) it is. Did it on our office system last week to see
> if it would work without any certificate needs. It just worked and during a
> save it will instantly tell you if it worked or not.
>
> Outside of the most regimented environments you should be able to just
> make the change. If it fails talk to your AD team as they would likely have
> something blocked or disabled.
>
> Get Outlook for iOS 
>
> Matthew Loraditch​
> Sr. Network Engineer
> p: *443.541.1518* <443.541.1518>
> w: *www.heliontechnologies.com*   |
> e: *mloradi...@heliontechnologies.com* 
>  
>  
>  
>  
> 
> --
> *From:* cisco-voip  on behalf of
> James Buchanan 
> *Sent:* Sunday, February 9, 2020 4:57:40 PM
> *To:* voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* [cisco-voip] Field Notice from Cisco making Secure LDAP
> mandatory
>
>
> [EXTERNAL]
>
> Hello folks,
>
> I know you all needed some more work. I sure did! So here you are!
>
>
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/trouble/12_5_1/fieldNotice/cucm_b_fn-secure-ldap-mandatory-ad.html
>
>
> I'm interested in any early thoughts on other integrations--vCenter, ISE,
> VPN, TACACS, etc. I assume it applies across the board.
>
> Thanks,
>
> James
>
>
> ___
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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] IM&Presence Persistent Chat My Rooms

[Brian Meade]

I'll probably post it on my GitHub once I get it cleaned up a bit!

On Tue, Dec 17, 2019, 2:40 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> All this mention of the Python file, and yet, no Python file.  What gives
> Brian?  ;)
>
> Thanks for the update!
>
> On Tue, Dec 17, 2019 at 1:36 PM Brian Meade  wrote:
>
>> Just to update everyone on this, these are stored in the pep_node and
>> pep_node_items tables on the actual IM&Presence server.
>>
>> I was able to find that documented here-
>> https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvm46828
>>
>> I was able to use a Python script to export the contents of pep_node and
>> pep_node_items and then insert those records into the new IM&Presence
>> cluster.  The pep_node table references the xcp_user_id which is different
>> than the actual user ID so you have to export those mappings from the old
>> cluster and then find the new xcp_user_id's on the new cluster.  XCP Router
>> must also be restarted after doing the Inserts.
>>
>> I was able to write a Python script to handle the export/import.
>>
>> On Wed, Dec 4, 2019 at 5:17 PM Anthony Holloway <
>> avholloway+cisco-v...@gmail.com> wrote:
>>
>>> It does seem to be held in the external database.  The IM&P server might
>>> just be compiling the my rooms list dynamically for the user at login,
>>> versus it being held in any one spot.
>>>
>>> TC_ROOMS Table
>>>
>>> The TC_ROOMS table contains information for group chat rooms.
>>>
>>> Column Name
>>>
>>> Postgres Datatype
>>>
>>> Oracle Datatype
>>>
>>> Microsoft SQL Datatype
>>>
>>> Not Null
>>>
>>> Description
>>>
>>> ROOM_JID
>>>
>>> VARCHAR (3071)
>>>
>>> VARCHAR2 (3071)
>>>
>>> varchar (3071)
>>>
>>> Yes
>>>
>>> The ID of the room.
>>>
>>> CREATOR_JID
>>>
>>> VARCHAR (3071)
>>>
>>> VARCHAR2 (3071)
>>>
>>> varchar (3071)
>>>
>>> Yes
>>>
>>> The ID of the user who created the room.
>>>
>>> SUBJECT
>>>
>>> VARCHAR (255)
>>>
>>> VARCHAR2 (255)
>>>
>>> varchar (255)
>>>
>>> Yes
>>>
>>> The current subject for the room.
>>>
>>> TYPE
>>>
>>> VARCHAR (32)
>>>
>>> VARCHAR2 (32)
>>>
>>> varchar (32)
>>>
>>> Yes
>>>
>>> The constraint check_type. This value must be either "ad-hoc" or
>>> "persistent".
>>>
>>> CONFIG
>>>
>>> TEXT
>>>
>>> CLOB
>>>
>>> text
>>>
>>> Yes
>>>
>>> The entire packet from the last time the room was configured. This
>>> information enables the room to be reconfigured when the room is recreated
>>> (for example, at start-up).
>>>
>>> SPACKET
>>>
>>> TEXT
>>>
>>> CLOB
>>>
>>> text
>>>
>>> Yes
>>>
>>> The entire packet from the last time the subject was set for the room.
>>> This information enables the room subject to be displayed when the room is
>>> recreated.
>>>
>>> START_MSG_ID
>>>
>>> BIGINT
>>>
>>> NUMBER (19)
>>>
>>> bigint
>>>
>>> Yes
>>>
>>> A sequence number that is used to populate the MSG_ID column in the
>>> TC_MSGARCHIVE table.
>>>
>>> Do not modify this value.
>>>
>>> NEXT_MSG_ID
>>>
>>> BIGINT
>>>
>>> NUMBER (19)
>>>
>>> bigint
>>>
>>> Yes
>>>
>>> A sequence number that is used to populate the MSG_ID column in the
>>> TC_MSGARCHIVE table.
>>>
>>> Do not modify this value.
>>> TC_USERS Table
>>>
>>> The TC_USERS table contains roles and affiliations, alternate names, and
>>> other data associated with group chat room users.
>>>
>>> Column Name
>>>
>>> Postgres Datatype
>>>
>>> Oracle Datatype
>>>
>>> Microsoft SQL Datatype
>>>
>>> Not Null
>>>
>>> Description
>>>
>>> ROOM_JID
>>>
>>> VARCHAR (3071)
>>>
>>> V

Re: [cisco-voip] IM&Presence Persistent Chat My Rooms

[Brian Meade]

Just to update everyone on this, these are stored in the pep_node and
pep_node_items tables on the actual IM&Presence server.

I was able to find that documented here-
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvm46828

I was able to use a Python script to export the contents of pep_node and
pep_node_items and then insert those records into the new IM&Presence
cluster.  The pep_node table references the xcp_user_id which is different
than the actual user ID so you have to export those mappings from the old
cluster and then find the new xcp_user_id's on the new cluster.  XCP Router
must also be restarted after doing the Inserts.

I was able to write a Python script to handle the export/import.

On Wed, Dec 4, 2019 at 5:17 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> It does seem to be held in the external database.  The IM&P server might
> just be compiling the my rooms list dynamically for the user at login,
> versus it being held in any one spot.
>
> TC_ROOMS Table
>
> The TC_ROOMS table contains information for group chat rooms.
>
> Column Name
>
> Postgres Datatype
>
> Oracle Datatype
>
> Microsoft SQL Datatype
>
> Not Null
>
> Description
>
> ROOM_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of the room.
>
> CREATOR_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of the user who created the room.
>
> SUBJECT
>
> VARCHAR (255)
>
> VARCHAR2 (255)
>
> varchar (255)
>
> Yes
>
> The current subject for the room.
>
> TYPE
>
> VARCHAR (32)
>
> VARCHAR2 (32)
>
> varchar (32)
>
> Yes
>
> The constraint check_type. This value must be either "ad-hoc" or
> "persistent".
>
> CONFIG
>
> TEXT
>
> CLOB
>
> text
>
> Yes
>
> The entire packet from the last time the room was configured. This
> information enables the room to be reconfigured when the room is recreated
> (for example, at start-up).
>
> SPACKET
>
> TEXT
>
> CLOB
>
> text
>
> Yes
>
> The entire packet from the last time the subject was set for the room.
> This information enables the room subject to be displayed when the room is
> recreated.
>
> START_MSG_ID
>
> BIGINT
>
> NUMBER (19)
>
> bigint
>
> Yes
>
> A sequence number that is used to populate the MSG_ID column in the
> TC_MSGARCHIVE table.
>
> Do not modify this value.
>
> NEXT_MSG_ID
>
> BIGINT
>
> NUMBER (19)
>
> bigint
>
> Yes
>
> A sequence number that is used to populate the MSG_ID column in the
> TC_MSGARCHIVE table.
>
> Do not modify this value.
> TC_USERS Table
>
> The TC_USERS table contains roles and affiliations, alternate names, and
> other data associated with group chat room users.
>
> Column Name
>
> Postgres Datatype
>
> Oracle Datatype
>
> Microsoft SQL Datatype
>
> Not Null
>
> Description
>
> ROOM_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of the room.
>
> REAL_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of a user in the room. This value is the actual ID of the user,
> rather than an alternate name.
>
> ROLE
>
> VARCHAR (32)
>
> VARCHAR2 (32)
>
> varchar (32)
>
> Yes
>
> The role of the user in the room. This value is constrained to one of the
> following: "none", "hidden", "visitor", "participant", or "moderator".
>
> AFFILIATION
>
> VARCHAR (32)
>
> VARCHAR2 (32)
>
> varchar (32)
>
> Yes
>
> The affiliation of the user in the room. This value is constrained to one
> of the following: "none", "outcast", "member", "admin", or "owner".
>
> NICK_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of the room, plus the alternate name for the user. The format is
> room@tc-server/nick.
>
> REASON
>
> VARCHAR (255)
>
> VARCHAR2 (255)
>
> varchar (255)
>
> Yes
>
> The reason entered when the user's affiliation was last changed.
>
> INITIATOR_JID
>
> VARCHAR (3071)
>
> VARCHAR2 (3071)
>
> varchar (3071)
>
> Yes
>
> The ID of the room in which the configuration change occurred.
>
> Source:
> https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/im_presence/database_setup/11_5_1/cup0_b_database-setup-guide-imp-115/cup0_b_database-setup-guide-imp-115_chapter_0100.html#CUP0_RF_TCBD4E17_00
>
>
>
>
>
> On Wed, Dec 4, 2019 at 3:40 PM Brian Meade  wrote:
>
>> Does anyone know where the My Rooms information is stored?  It doesn't
>> seem to be in the persistent chat database or stored locally on the
>> client.  Is this stored on IM&P somewhere?
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
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[cisco-voip] IM&Presence Persistent Chat My Rooms

[Brian Meade]

Does anyone know where the My Rooms information is stored?  It doesn't seem
to be in the persistent chat database or stored locally on the client.  Is
this stored on IM&P somewhere?
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Re: [cisco-voip] Force Logout Jabber

[Brian Meade]

Yea, I was actually looking at some crash bugs to see if that's an easier
route haha.

On Thu, Nov 14, 2019, 4:27 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Interesting, so you want to change the client behavior, such that you can
> remotely signal the client to return the login screen?
>
> I have never looked into that, but it sounds like a tall order to me.  I'd
> be surprised if that's possible.  But, I'll watch this thread for further
> developments.
>
> BRB, writing a buffer overflow exploit which causes jabber to crash...
>
> On Thu, Nov 14, 2019 at 3:21 PM Brian Meade  wrote:
>
>> Has anyone found a way to force logout all or some Jabber clients on an
>> IM&P Cluster?
>>
>> Restarting XCP Router or un-assigning presence service for the user just
>> makes them spin and show it can't connect to IM&P.
>>
>> ForceLogoutTimerDesktop is in the parameter guide but pushing this out
>> via jabber-config with a short timer still doesn't mean the clients are
>> going to get the new jabber-config right away.
>> ___
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>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
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[cisco-voip] Force Logout Jabber

[Brian Meade]

Has anyone found a way to force logout all or some Jabber clients on an
IM&P Cluster?

Restarting XCP Router or un-assigning presence service for the user just
makes them spin and show it can't connect to IM&P.

ForceLogoutTimerDesktop is in the parameter guide but pushing this out via
jabber-config with a short timer still doesn't mean the clients are going
to get the new jabber-config right away.
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Re: [cisco-voip] Webex room devices with 3rd party meetings

[Brian Meade]

Yea, I'm not a huge fan of the Webex Directory Connector.  It could be a
lot better.  Something like Synergy Sky works much better.

On Thu, Nov 14, 2019 at 12:46 PM Lelio Fulgenzi  wrote:

> Just a heads up that the Hybrid calendar parse tool does _*not*_ work
> with forwarded invites last time I checked. This is a big drag. You can
> copy/paste meeting details into a new meeting I think.
>
>
>
> Or did I already say this?
>
>
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* cisco-voip  *On Behalf Of *Brian
> Meade
> *Sent:* Thursday, November 14, 2019 12:20 PM
> *To:* Myron Young 
> *Cc:* cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] Webex room devices with 3rd party meetings
>
>
>
> The RoomKit mini can act as a USB device and then they can use a laptop to
> join the meeting.
>
>
>
> The other devices right now only support calling in to other meeting
> services via SIP/H.323.  Zoom has Cloud Room Connector licenses you can pay
> for to enable dialing in via SIP/H.323.  With Microsoft Teams, they don't
> allow native SIP/H.323 call-in so you have to use a 3rd party interop
> service such as Pexip/Polycom.  Cisco just announced that they will be
> building an interop service but there's not many details around it yet.
>
>
>
> You'll need to register the Webex devices to CUCM or Webex and that may
> incur some additional costs depending upon your licensing/subscription.
>
>
>
> For one button to push, the Webex Directory Connector will try to parse
> SIP URI's and add the join button.  Sometimes using a 3rd party directory
> connector may be a better option if you run into any issues.  Zoom has one
> for Zoom meetings for Cisco endpoints.  There's other companies like
> Synergy Sky that have alternatives for Cisco endpoints.
>
>
>
> When joining meetings via SIP/H.323, you most likely won't get the full
> native client experience with MS/Zoom such as roster list, being able to
> mute other participants, etc.
>
>
>
> Cisco is working on a WebRTC join experience using the web engine on the
> newer endpoints that will allow you to join other meeting services with a
> more native experience.  For the MS Teams WebRTC experience, you'll be
> limited to single screen though.
>
>
>
> On Wed, Nov 13, 2019 at 4:24 PM Myron Young 
> wrote:
>
> Hello,
>
> Anyone has any good experience in joining non-Webex meetings from the
> Cisco roomkit devices? Trying to deploy a standard for conference rooms
> with roomkit devices and touch 10 but need to make sure user experience is
> easy or similar when joining Webex meetings vs non-webex meetings such as
> Zoom, Blue Jeans or Microsoft meetings.
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
___
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Re: [cisco-voip] Webex room devices with 3rd party meetings

[Brian Meade]

This is a global change in Control Hub.  You can say to not require host
pins at all from cloud-registered video endpoints.  No other changes needed.

On Thu, Nov 14, 2019 at 12:33 PM Lelio Fulgenzi  wrote:

>
>
> You can. But the way to do this is long and arduous. It requires users to
> create a template in Web interface and then use that template in outlook.
>
>
>
> I’m hoping they include this option soon in the options/settings available
> in the outlook plug-in natively.
>
>
>
>
>
> ---
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
>
>
> *From:* Brian Meade 
> *Sent:* Thursday, November 14, 2019 12:13 PM
> *To:* Lelio Fulgenzi 
> *Cc:* Pawlowski, Adam ; James Andrewartha <
> jandrewar...@ccgs.wa.edu.au>; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Webex room devices with 3rd party meetings
>
>
>
> If the endpoints are cloud-registered, you can disable it asking for a
> host pin.
>
>
>
> On Thu, Nov 14, 2019 at 9:56 AM Lelio Fulgenzi  wrote:
>
>
> I think you've hit the nail on the head Adam.
>
> The idea of these room systems are great... they really do making joining
> meetings easy (as long as you know to bring your host PIN, argh!).
>
> But I think we are going to be relegated to providing the best fit for
> all, which, in the end, might be a USB enabled room. ☹
>
> Why Cisco didn't (and doesn't intend to) extend the USB-plug-webcam option
> for their larger units is beyond me. Could be technical, who knows. But
> that would surely open up the justification for these room systems.
>
>
>
> ---
> 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
>
>
>
> -Original Message-
> From: cisco-voip  On Behalf Of
> Pawlowski, Adam
> Sent: Thursday, November 14, 2019 7:52 AM
> To: 'James Andrewartha' ;
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Webex room devices with 3rd party meetings
>
> I don't know if that whole mess is necessary to join a Zoom call. If the
> party who sent you the Zoom invite has the connector service for interop,
> you can usually call the meeting by @zoomcrc.com as  SIP
> call. If they don't, it prompts you to sign in on an app with your account
> (which is paying for a connector license I guess) and then you can join the
> meeting that way.
>
> In my experience with that, I've yet to come across a Zoom meeting I've
> been invited to where someone has the connector on their account, and we
> don't pay for one. In that case the Room Kit ends up being largely useless.
> Yes you can share in a PC but then making use of the camera is expensive.
>
> And before anyone waves in with "Room Kit Mini", the framing on that is
> meant for some very limited room sizes and is not suitable to the type of
> space you'd be installing a proper codec to anyways.
>
> All of these disparate services work best if you can coax the other
> participant(s) into playing in your house with your services.
>
> We've unfortunately decided not to pursue the Webex systems given this,
> and have instead gone with external cameras and microphones that we can
> present to a PC as a USB device. Not for everyone and doesn't fit all cases.
>
>
>
> > -Original Message-
> > From: cisco-voip  On Behalf Of
> > James Andrewartha
> > Sent: Wednesday, November 13, 2019 10:43 PM
> > To: cisco-voip@puck.nether.net
> > Subject: Re: [cisco-voip] Webex room devices with 3rd party meetings
> >
> > Zoom do sell a connector for Cisco room systems, but I don't know how
> > well it works for joining third-party zoom meetings. The setup
> > instructions are quite involved
> > https://support.zoom.us/hc/en-us/articles/115003126346-Zoom-
> > Connector-for-Cisco
> >
> > I've just put in a Logitech Tap Microsoft Teams Room system and the
> > announced compatibility with Webex and Zoom makes me happy we went
> > with them. Someone noted there's been no public agreement between
&g

Re: [cisco-voip] Webex room devices with 3rd party meetings

[Brian Meade]

The RoomKit mini can act as a USB device and then they can use a laptop to
join the meeting.

The other devices right now only support calling in to other meeting
services via SIP/H.323.  Zoom has Cloud Room Connector licenses you can pay
for to enable dialing in via SIP/H.323.  With Microsoft Teams, they don't
allow native SIP/H.323 call-in so you have to use a 3rd party interop
service such as Pexip/Polycom.  Cisco just announced that they will be
building an interop service but there's not many details around it yet.

You'll need to register the Webex devices to CUCM or Webex and that may
incur some additional costs depending upon your licensing/subscription.

For one button to push, the Webex Directory Connector will try to parse SIP
URI's and add the join button.  Sometimes using a 3rd party directory
connector may be a better option if you run into any issues.  Zoom has one
for Zoom meetings for Cisco endpoints.  There's other companies like
Synergy Sky that have alternatives for Cisco endpoints.

When joining meetings via SIP/H.323, you most likely won't get the full
native client experience with MS/Zoom such as roster list, being able to
mute other participants, etc.

Cisco is working on a WebRTC join experience using the web engine on the
newer endpoints that will allow you to join other meeting services with a
more native experience.  For the MS Teams WebRTC experience, you'll be
limited to single screen though.

On Wed, Nov 13, 2019 at 4:24 PM Myron Young 
wrote:

> Hello,
>
> Anyone has any good experience in joining non-Webex meetings from the
> Cisco roomkit devices? Trying to deploy a standard for conference rooms
> with roomkit devices and touch 10 but need to make sure user experience is
> easy or similar when joining Webex meetings vs non-webex meetings such as
> Zoom, Blue Jeans or Microsoft meetings.
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
___
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Re: [cisco-voip] Webex room devices with 3rd party meetings

[Brian Meade]

If the endpoints are cloud-registered, you can disable it asking for a host
pin.

On Thu, Nov 14, 2019 at 9:56 AM Lelio Fulgenzi  wrote:

>
> I think you've hit the nail on the head Adam.
>
> The idea of these room systems are great... they really do making joining
> meetings easy (as long as you know to bring your host PIN, argh!).
>
> But I think we are going to be relegated to providing the best fit for
> all, which, in the end, might be a USB enabled room. ☹
>
> Why Cisco didn't (and doesn't intend to) extend the USB-plug-webcam option
> for their larger units is beyond me. Could be technical, who knows. But
> that would surely open up the justification for these room systems.
>
>
>
> ---
> 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
>
>
>
> -Original Message-
> From: cisco-voip  On Behalf Of
> Pawlowski, Adam
> Sent: Thursday, November 14, 2019 7:52 AM
> To: 'James Andrewartha' ;
> cisco-voip@puck.nether.net
> Subject: Re: [cisco-voip] Webex room devices with 3rd party meetings
>
> I don't know if that whole mess is necessary to join a Zoom call. If the
> party who sent you the Zoom invite has the connector service for interop,
> you can usually call the meeting by @zoomcrc.com as  SIP
> call. If they don't, it prompts you to sign in on an app with your account
> (which is paying for a connector license I guess) and then you can join the
> meeting that way.
>
> In my experience with that, I've yet to come across a Zoom meeting I've
> been invited to where someone has the connector on their account, and we
> don't pay for one. In that case the Room Kit ends up being largely useless.
> Yes you can share in a PC but then making use of the camera is expensive.
>
> And before anyone waves in with "Room Kit Mini", the framing on that is
> meant for some very limited room sizes and is not suitable to the type of
> space you'd be installing a proper codec to anyways.
>
> All of these disparate services work best if you can coax the other
> participant(s) into playing in your house with your services.
>
> We've unfortunately decided not to pursue the Webex systems given this,
> and have instead gone with external cameras and microphones that we can
> present to a PC as a USB device. Not for everyone and doesn't fit all cases.
>
>
>
> > -Original Message-
> > From: cisco-voip  On Behalf Of
> > James Andrewartha
> > Sent: Wednesday, November 13, 2019 10:43 PM
> > To: cisco-voip@puck.nether.net
> > Subject: Re: [cisco-voip] Webex room devices with 3rd party meetings
> >
> > Zoom do sell a connector for Cisco room systems, but I don't know how
> > well it works for joining third-party zoom meetings. The setup
> > instructions are quite involved
> > https://support.zoom.us/hc/en-us/articles/115003126346-Zoom-
> > Connector-for-Cisco
> >
> > I've just put in a Logitech Tap Microsoft Teams Room system and the
> > announced compatibility with Webex and Zoom makes me happy we went
> > with them. Someone noted there's been no public agreement between
> > Webex and Zoom, only between them and MS Teams.
> >
> > --
> > James Andrewartha
> > Network & Projects Engineer
> > Christ Church Grammar School
> > Claremont, Western Australia
> > Ph. (08) 9442 1757
> > Mob. 0424 160 877
> >
> > On 14/11/19 7:13 am, Lelio Fulgenzi wrote:
> > >
> > > The question is whether or not the Zoom devices can dial other systems.
> > >
> > > If you're getting Zoom across the board, that's fine. But if you
> > > can't
> > participate with other systems, then you've further pigeon hole'd
> yourself.
> > >
> > > If that makes sense.
> > >
> > > The fact that Webex/Microsoft signed a deal is even more weight for
> > > Webex
> > devices.
> > >
> > > But I hear what you're saying. If your zoom, hard to say buy webex.
> > > ☹
> > >
> > > ---
> > > 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
> > >
> > >
> > >
> > > -Original Message-
> > > From: Myron Young 
> > > Sent: Wednesday, November 13, 2019 5:52 PM
> > > To: Lelio Fulgenzi 
> > > Cc: cisco-voip voyp list 
> > > Subject: Re: Webex room devices with 3rd party meetings
> > >
> > > Thanks for the insight. I will try with the cloud device to test
> > > this out and see if
> > i can convince the powers that be. Otherwise we may be going full Zoom
> > which I’m not adverse to, but would be very unfamiliar territories.
> > >
> > >> On Nov 13, 2019, at 4:40 PM, Lelio Fulgenzi 
> wrote:
> > >>
> > >>
> > >> I've not had any production experience, only testin

Re: [cisco-voip] UCCX Callback Script

[Brian Meade]

Anthony,

I'm curious how you handle catching when the agent answers the callback
request.

I've got my scripts checking to see if the CallBack contact was answered by
setting some Enterprise Info in my callback queue script but I still have
to check every few seconds to see if that Enterprise Info is set.

I just max out the max steps to account for that.

Thanks,
Brian Meade

On Tue, Nov 5, 2019 at 4:19 PM Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Hi Tim,
>
> I think the idea of a flawless script is in the eyes of the beholder.
>
> I don't personally use the example script from the repo; are you talking
> about the one here:
>
>
> script_respository_902\script_respository\release3\BaseLineAdvQueuing\BaseLineAdvQueuing.aef
>
>
> If so, there a few things wrong with that script.
>
> For example, you said "...despite having Contact Inactive exception error
> handling..."
>
> Yeah, they setup an exception handler at the top for
> ContactInactiveException, but then they never clear it, or reset it, and so
> if and when the caller disconnects while recording their message or
> listening to the "success" prompt, the whole thing falls a part and fails,
> sending script execution down to the ExceptionCIE label.
>
> Another thing wrong with it is that the waiting mechanism for the Agent is
> such that it plays a relatively short prompt, waits 3 seconds for input
> from the Agent, then repeats.
>
> If you consider every application has a max 1,000 steps it can execute,
> and you subtract off the overhead of just getting the call to this point
> (say 21 steps in the most streamlined of scenarios), that leaves you with
> 32 minutes to queue a call, otherwise the call will be aborted.  Since most
> people are only interested in callback when they have queue hold time
> problems, this is likely to cause more issues than it solves.
>
> "...I’ve read that the Call Control Group and Dialog Group should be
> different from the trigger on the originating application..."
>
> Can you link the source?
>
>
> On Tue, Nov 5, 2019 at 10:59 AM Johnson, Tim  wrote:
>
>> Anyone have a callback script that is working flawlessly? We have
>> implemented the solution in Cisco’s Advanced Queueing script and it’s seems
>> to be working, but I’m seeing Contact Inactive Exceptions and Contact
>> Creation errors in syslog each time the callback is used, despite having
>> Contact Inactive exception error handling.
>>
>>
>>
>> It seems that the issue may be related to the Place Call step which calls
>> the trigger of the callback application. I’ve read that the Call Control
>> Group and Dialog Group should be different from the trigger on the
>> originating application (which is what we have setup), but I’m curious if
>> those should also be different from what’s used on the callback
>> application. If so, can I use the same CCG and DG from the original
>> trigger, on the callback trigger?
>>
>>
>>
>> For example, I have the following setup:
>>
>> App_A application has a trigger that uses CCG #8 and Dialog Group #0. In
>> its script, it uses the Place Call step with CCG #25 and Dialog Group #3.
>> This places the call to App_Callback application which has a trigger that
>> uses CCG #25 and Dialog Group #3.
>>
>>
>>
>> Tim Johnson
>>
>> Voice & Video Engineer
>>
>> Central Michigan University
>>
>> Phone: +19897744...@cmich.edu
>>
>> Fax: +19897795900
>>
>> [image: webexemailsig] <https://cmich.webex.com/meet/johns10t>
>>
>>
>> ___
>> 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] Automatic disconnect

[Brian Meade]

Setup the ring no answer to go to voicemail after 10/20 seconds.  In Unity
Connection, set it to play a blank greeting then set to hang up as the
after greeting action.

I'm assuming you want it to ring and then hang up after 10/20 seconds?  Or
will the destination phone answer the call?

On Sun, Nov 3, 2019 at 7:43 AM  wrote:

>
>
>
>
> Can I program a speed dial button on a 7945/7965 to call an extension and
> have it automatically hang-up in say 10/20 seconds.  Looking at using a
> speed dial on a 7945/7965 as a panic button.
>
>
>
>
>
>
>
> Thanks
>
>
>
>
>
>
>
>
>
> *Norm Nicholson*
>
> *Telecom Analyst*
>
> *City of Kitchener*
>
> *(519) 741-2200 x 7000*
>
>
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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[cisco-voip] Webex Proximity Callback to Video Endpoints with CUCM

[Brian Meade]

Hey everyone,

With Webex Meetings video callback to a CUCM-registered video endpoint via
proximity, Webex always tries the d...@domain.com format rather than the
Primary URI configured.

We've confirmed that the video endpoints advertise the d...@domain.com as
well as the Primary URI but Webex Meetings is setup to prefer d...@domain.com.

In our organization and many customer organizations, the B2B Expressways do
not allow d...@domain.com calling.

This issue has become much worse with the new Webex join experience
launched recently.

At ePlus, my coworker opened a Collaboration Idea for resolving this-
https://ciscocollaboration.ideas.aha.io/ideas/COLLAB-I-3255

Webex BU is using this to prioritize fixes and new features.

Can you vote for this issue if it is applicable to yourself or a customer
of yours?  We're really hoping to drive some visibility of this issue.

Thanks,
Brian Meade
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Re: [cisco-voip] UCCx and Jabber - compatible or not?

[Brian Meade]

Be careful with that as UCCX loves to reset all the agent's settings.  It's
worth testing but you may lose all the agent config.

On Fri, Oct 18, 2019 at 3:51 PM Johnson, Tim  wrote:

> As for your “fake extension mobility” suggestion, you could possibly do
> that by having two DNs designated for the user intended for UCCX. Have one
> assigned to the SEP device, the other to the CSF. Have both of those
> assigned to the End User as a Controlled Device, and then setup an AXL call
> to change the IPCC extension on the End User. If I’m thinking right that’ll
> basically give you (and potentially the user) the ability to toggle to
> their preference.
>
>
>
> *From:* Pawlowski, Adam 
> *Sent:* Friday, October 18, 2019 3:43 PM
> *To:* 'Lelio Fulgenzi' ; Matthew Loraditch <
> mloradi...@heliontechnologies.com>; Johnson, Tim ;
> voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* RE: UCCx and Jabber - compatible or not?
>
>
>
> Only in Phone Only or it doesn’t work, or if you are assigning a custom
> configuration or phone.
>
>
>
> In our case (love to toot my own horn), jabber-config.xml turns off phone
> and voicemail. If you want a person to cover a group VM box, or have
> special features (pChat etc) the only way to tell the client what to do is
> to build the CSF device to steer the configuration.
>
>
>
> Regarding this, there’s a community forum post that says if you sign in to
> the phone, then open Jabber, then sign out of the phone, that it “works”
> because CTI still can find the station. I assume this has something to do
> with the order in which the devices are retrieved when you ask for a
> terminal list with CTI but I don’t know.
>
>
>
> It was followed up with a long post of “yeah maybe it works maybe it
> doesn’t but it is not supported”.
>
>
>
> Someone should write some nice wrapped code to move your DN around like a
> fake extension mobility. If you have no other considerations, with AXL it’s
> saving the phone back with that DN out of the line list, and saving it back
> on the CSF device. Probably break a bunch of things but if not I’m sure
> there’s a few bucks to be made off of that.
>
>
>
> Since it is a Friday I don’t intend to touch any of that and demo is since
> I’ll probably end up deleting everyone’s phone line or something.
>
>
>
> Adam
>
>
>
> *From:* cisco-voip  *On Behalf Of *Lelio
> Fulgenzi
> *Sent:* Friday, October 18, 2019 3:36 PM
> *To:* Matthew Loraditch ; Johnson, Tim
> ; voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* Re: [cisco-voip] UCCx and Jabber - compatible or not?
>
>
>
> Ok, now you’re really freaking me out.
>
>
>
> *You don’t need a CSF device to log in with Jabber?*
>
>
>
>
>
> ---
>
> *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:* Matthew Loraditch 
> *Sent:* Friday, October 18, 2019 3:32 PM
> *To:* Johnson, Tim ; Lelio Fulgenzi ;
> voyp list, cisco-voip (cisco-voip@puck.nether.net) <
> cisco-voip@puck.nether.net>
> *Subject:* RE: UCCx and Jabber - compatible or not?
>
>
>
> Yeah just don’t create one if you don’t like having them blank. You don’t
> need a CSF for deskphone control.
>
>
>
>
>
> *Matthew Loraditch**​*
>
> *Sr. Network Engineer*
>
> p: *443.541.1518* <443.541.1518>
>
> w: *www.heliontechnologies.com*
> 
>
>  |
>
> e: *mloradi...@heliontechnologies.com* 
>
> [image: Helion Technologies]
> 
>
> [image: Facebook]
> 
>
> [image: Twitter]
> 

Re: [cisco-voip] Thoughts on options for wireless conference phones

[Brian Meade]

You can also use the 8832 with Wifi and one of those rechargeable power
packs connected via USB-C.

On Thu, Oct 17, 2019 at 4:19 PM Brian Meade  wrote:

> I have some clients that use these-
> https://www.konftel.com/en/products/konftel-300wx
>
> Looks a bit more like a normal conference phone than the Revo Labs setup.
>
> On Wed, Oct 16, 2019 at 10:43 PM Dana Tong 
> wrote:
>
>> Hi all,
>>
>>
>>
>> Anyone have any thoughts on 100% wireless conference phones?
>>
>>
>>
>> Like maybe something that can be docked for charging but you can pick it
>> up and put in on the desk and still dial in ad-hoc conference participants
>> from the device?
>>
>>
>>
>> Maybe some kind of wireless third-party SIP device?
>>
>>
>>
>> Cheers
>>
>> Dana
>>
>>
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Re: [cisco-voip] Thoughts on options for wireless conference phones

[Brian Meade]

I have some clients that use these-
https://www.konftel.com/en/products/konftel-300wx

Looks a bit more like a normal conference phone than the Revo Labs setup.

On Wed, Oct 16, 2019 at 10:43 PM Dana Tong  wrote:

> Hi all,
>
>
>
> Anyone have any thoughts on 100% wireless conference phones?
>
>
>
> Like maybe something that can be docked for charging but you can pick it
> up and put in on the desk and still dial in ad-hoc conference participants
> from the device?
>
>
>
> Maybe some kind of wireless third-party SIP device?
>
>
>
> Cheers
>
> Dana
>
>
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Re: [cisco-voip] 8865s and MRA CUCM registration failover issue

[Brian Meade]

You're may be hitting this limitation-
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvj49486


If not, are all 3 CUCM servers in the _cisco-uds SRV record and resolvable
by the Expressway-C?

On Mon, Oct 14, 2019 at 11:47 AM Erick Bergquist  wrote:

> Has anyone seen where 8865 model phones don't register over MRA  in
> the UCM group if the some servers are not reachable?
>
> 8865s with 12.5.1 SR3 firmware
> 12.5.1 SU1 CUCM
>
> 2 expressway pairs
>
> UCM group order (same as service group),
>
> CCM1
> CCM2
> CCM3
>
> When CCM1 and CCM2 are unreachable the MRA 8865 phone just spins at
> registering.
> Once CCM1 or CCM2 become reachable, the phone comes backs up.
>
> DX 70's and DX 80's register fine when CCM3 is only available over MRA.
>
>
> Thanks,
> Erick
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Re: [cisco-voip] Super Duper Improved Jabber on it's way to you...

[Brian Meade]

Heads up, major issue is having Jabber 12.7 and Webex Meetings app
installed.

By default, Jabber 12.7 tries to launch Webex meetings in Jabber as a SIP
client which obviously isn't the full Webex experience.

Users can change if they join with Webex Meetings app or Jabber app in
dropdown on meeting reminder or in the dropdown at the bottom of the
Meetings tab.

You can change this in the jabber-config.xml under Options section with a
parameter called ConfMediaType-
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_7/cjab_b_parameter-reference-guide-jabber-127/cjab_b_parameter-reference-guide-jabber-127_chapter_0100.html#CJAB_RF_C2BEDA3A_00

You can also change the default layout back to the old style via the
UXModel parameter-
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/12_7/cjab_b_parameter-reference-guide-jabber-127/cjab_b_parameter-reference-guide-jabber-127_chapter_011.html#reference_B422E797A575DD26F869FE1DAFD0CB96





On Fri, Oct 11, 2019 at 9:36 AM Lelio Fulgenzi  wrote:

>
> Are you excited? Cisco sure is For Jabber 12.7 !
>
> At my Cisco Live recap, I talked about how Jabber would be seeing a major
> cosmetic overhaul to match it's other collaboration clients *cough* Webex
> Teams *cough* was Spark *cough*. It was released mid-September, and aside
> from the visual changes, I haven't heard anything bad about it. There will
> be a dot one version released in the next month or so, which we will likely
> use as the roll out version, so we want to get some experience with 12.7 as
> soon as possible.
>
> I'll be testing on my machine today and if there's nothing glaring, I'll
> be pushing it out to the rest of the team as well.
>
> Let me know if you have any issues.
>
> 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] SIP Domain substitution

[Brian Meade]

I don't think DNS SRV records support CNAME.  Even then, it would only
change where it was sent to and not the SIP headers.

On Fri, Oct 4, 2019 at 12:26 PM Lelio Fulgenzi  wrote:

> Yeah – I’d want this to happen all within DNS. But of course, in a
> supported fashion. I’m not interested in spending time modifying
> infrastructure at this time.
>
>
>
> I’ve done some searching, and there’s talk of RR records, but we haven’t
> found much documentation.
>
>
>
>
>
> ---
>
> *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:* Dave Goodwin 
> *Sent:* Friday, October 4, 2019 12:09 PM
> *To:* Lelio Fulgenzi 
> *Cc:* cisco-voip voyp list 
> *Subject:* Re: [cisco-voip] SIP Domain substitution
>
>
>
> Are you wanting this to all happen within DNS instead of happening within
> a SIP UA? As far as I understand, if DNS redirected somewhere (SRV or CNAME
> record for example) it would not change the destination URI the originator
> is trying to reach. The SIP protocol has redirection codes (such as 301 or
> 302) but whether or how you might be able to use them depends on the SIP
> UAs being used.
>
>
>
> You might also be able to use something like a SIP normalization script
> (CUCM), SIP profiles (CUBE), or maybe search pattern replacements
> (Expressway) to just translate the domain as calls flow in/out. I'm
> guessing what might be feasible without knowing more of the picture.
>
>
>
> On Fri, Oct 4, 2019 at 11:10 AM Lelio Fulgenzi  wrote:
>
>
>
> Does SIP allow for domain name substitution?
>
>
>
> By this I mean, instead of advertising or dialing
> coy...@phones.america.acmemanufacturing.com I want to use coy...@zing.com
>
>
>
> But I don’t want to have to reorganize and reprogram anything.
>
>
>
> I just want the DNS to say, “hey, use this domain instead and try again.”
>
> *-sent from mobile device-*
>
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
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Re: [cisco-voip] cisco uccx licensing concurrent or named

[Brian Meade]

The 175 Premium UCCX Seats would be concurrent logged-in agents.

QM/AQM/WFM/CR licenses are based on named user.

On Thu, Oct 3, 2019 at 10:27 PM naresh rathore  wrote:

> hi
>
>
> I can see following when i check license page of uccx. which one of them
> is concurrent and which one of them is named based, how to find out
>
> *Configured Licenses:*
>
> Package: Cisco Unified CCX Premium
>
> Total IVR Port(s): 350
>
> Cisco Unified CCX Premium Seat(s): 175
>
> High Availability : Enabled
>
> Cisco Unified CCX Preview Outbound Dialer: Enabled
>
> Cisco Unified CCX Quality Manager Seat(s): 100
>
> Cisco Unified CCX Advanced Quality Manager Seat(s): 170
>
> Cisco Unified CCX Workforce Manager Seat(s): 170
>
> Cisco Unified CCX Compliance Recording Seat(s): 100
>
> Cisco Unified CCX Maximum Agents: 400
>
> *Recording:*
>
> Cisco Unified CCX Recording Count:100
>
> *Inbound:*
>
> Available Inbound IVR Port(s): 350
>
> *Outbound:* (Predictive and Progressive only)
>
> Cisco Unified CCX Licensed Outbound IVR Port(s): 100
>
> Cisco Unified CCX Outbound IVR Port(s) In Use: 0
>
> Cisco Unified CCX Licensed Outbound Agent Seat(s): 100
>
> Cisco Unified CCX Outbound Agent Seat(s) In Use: 0
>
>
> Regards
>
> Naray
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Re: [cisco-voip] best software/device to simulate ITSP and test CUBE functionality

[Brian Meade]

SIPp will allow you to create your own SIP messages however you want-
http://sipp.sourceforge.net/

We used this to troubleshoot issues in TAC that were difficult to reproduce.

On Thu, Oct 3, 2019 at 8:54 PM naresh rathore  wrote:

> hi
>
>
> what is the best software/device i can use to simulate ITSP and test CUBE
> configuration, my purpose is to change settings on ITSP software/device and
> check sip messages and make corresponding cube config changes for
> troubleshooting and testing purpose
>
>
> Regards
>
> Naray
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Re: [cisco-voip] How are you all dealing with Jabber and 911????

[Brian Meade]

Redsky is the way to go for this.  They have a softphone client add-on that
pops up to ask for address when your location changes.

On Thu, Oct 3, 2019 at 2:14 PM Scott Voll  wrote:

> CER / CM 12.5
>
> See attached PDF of for the diagram.  Starting at the bottom left you see
> all our Jabber stuff internally goes to CM and CER and works as needed.
>
> Moving our way up the diagram you will see a line from the working
> condition going up to remote users.  corporate laptops that are connected
> via the VPN. (problem number 1)
>
> Down from that, is non corp, home users that are using jabber in the
> office via there RDP connection over the VPN. (problem number 2)
>
> then at the top is the jabber clients that have internet access coming in
> via Expressways (problem number 3).
>
> what are others doing about routing 911?  we currently use desk phones in
> the office, but we may be moving to jabber and ditching the desk phones.
> we really need to come up with a way for 911 to work, no matter the
> connectivity.
>
> Back, back in the day, I remember that a very large aircraft company used
> a third party (can't remember who they used) with IP communicator that
> asked for a address and routed 911 out the local 911 call center via a SIP
> trunk.  This is the only way I can think to fix this problem, IF that e911
> provider is still around, and IF they work with Jabber.  Do you guys have
> any other options?
>
> TIA
>
> Scott
>
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Re: [cisco-voip] Microsoft Graph/OAuth 2.0

[Brian Meade]

Unity Connection still just supports EWS.  Microsoft Graph is on the
roadmap last I heard.

On Wed, Oct 2, 2019 at 10:41 AM Benjamin Turner 
wrote:

> Has anyone found any documentation from Cisco regarding Microsoft
> Graph/OAuth 2.0 and Unity authentication for 365 customers? Seems that EWS
> will not be supported.
>
>
>
>
>
>
>
> Thanks,
>
> Ben
>
>
>
>
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Re: [cisco-voip] Jabber MRA with Cisco Umbrella

[Brian Meade]

This is for the Umbrella client installed on all the PCs so it is always
using Umbrela DNS except for any domains specified as internal.

On Thu, Sep 26, 2019, 1:49 PM Scott Voll  wrote:

> are you using an always connected  VPN configuration?  Like Mike said.  in
> our environment, our umbrella VM's point to the internal DNS servers.
> Outside our corporation Umbrella uses the external DNS (hosted elsewhere).
> I don't understand why you are getting the same response both
> internally and externally.
>
> Scott
>
>
> On Tue, Sep 24, 2019 at 6:27 PM Brian Meade  wrote:
>
>> Issue would be a corporate PC with umbrella going off-site.  If you add
>> your internal domains, it would get the _cisco-uds record always rather
>> then _collab-edge.
>>
>> On Tue, Sep 24, 2019, 6:34 PM Norton, Mike 
>> wrote:
>>
>>> Have never used Umbrella for external clients, but I would be very
>>> surprised if it somehow magically exposed your “local” domains to external
>>> clients. Internal clients use the internal Umbrella virtual appliance to
>>> resolve names, and if the request is for a domain defined as “local”, the
>>> virtual appliance then uses the internal DNS server to resolve the name.
>>> External clients would not have access to the internal virtual appliance
>>> nor to the internal DNS server, so it should not be possible for external
>>> clients to get internal answers. IIRC the list of “local” domains is per
>>> “site” and external clients would not be in scope for the site.
>>>
>>> Defining a local domain is probably what you want.
>>>
>>> I could be wrong though - stopped using Umbrella after Cisco bought it
>>> and tried to more than quadruple the pricing on us.
>>>
>>> -mn
>>>
>>>
>>>
>>> *From:* cisco-voip  *On Behalf Of *Brian
>>> Meade
>>> *Sent:* September 24, 2019 12:37 PM
>>> *To:* cisco-voip voyp list 
>>> *Subject:* [cisco-voip] Jabber MRA with Cisco Umbrella
>>>
>>>
>>>
>>> Has anyone been able to get this to work?
>>>
>>>
>>>
>>> Umbrella always finds the _collab-edge SRV record even when internally.
>>> I imagine if we made the voice services domain a local domain we would have
>>> the reverse issue of always seeing _cisco-uds even when external.
>>>
>>>
>>>
>>> Any Umbrella features that could help here?
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Brian Meade
>>>
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Re: [cisco-voip] Jabber MRA with Cisco Umbrella

[Brian Meade]

Issue would be a corporate PC with umbrella going off-site.  If you add
your internal domains, it would get the _cisco-uds record always rather
then _collab-edge.

On Tue, Sep 24, 2019, 6:34 PM Norton, Mike  wrote:

> Have never used Umbrella for external clients, but I would be very
> surprised if it somehow magically exposed your “local” domains to external
> clients. Internal clients use the internal Umbrella virtual appliance to
> resolve names, and if the request is for a domain defined as “local”, the
> virtual appliance then uses the internal DNS server to resolve the name.
> External clients would not have access to the internal virtual appliance
> nor to the internal DNS server, so it should not be possible for external
> clients to get internal answers. IIRC the list of “local” domains is per
> “site” and external clients would not be in scope for the site.
>
> Defining a local domain is probably what you want.
>
> I could be wrong though - stopped using Umbrella after Cisco bought it and
> tried to more than quadruple the pricing on us.
>
> -mn
>
>
>
> *From:* cisco-voip  *On Behalf Of *Brian
> Meade
> *Sent:* September 24, 2019 12:37 PM
> *To:* cisco-voip voyp list 
> *Subject:* [cisco-voip] Jabber MRA with Cisco Umbrella
>
>
>
> Has anyone been able to get this to work?
>
>
>
> Umbrella always finds the _collab-edge SRV record even when internally.  I
> imagine if we made the voice services domain a local domain we would have
> the reverse issue of always seeing _cisco-uds even when external.
>
>
>
> Any Umbrella features that could help here?
>
>
>
> Thanks,
>
> Brian Meade
>
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