Re: [cisco-voip] [External] Voice Gateway Dial-Peer Precedence/Processing Order

[Johnson, Tim]

In the preference list on that page, they list "incoming uri" higher than 
"incoming called-number" for inbound H.323 call legs. So based on that, the 
"incoming uri" dial peer should be chosen first if the string is the same in 
each dial peer. The order in which it appears in the config or the dial peer 
number does not make any difference.

From: Gary Parker 
Sent: Friday, November 11, 2022 8:05 AM
To: Johnson, Tim ; voip puck 
Subject: Re: [External] [cisco-voip] Voice Gateway Dial-Peer 
Precedence/Processing Order

Thanks Tim, I'm not sure fully understand the precedence, still.

If I have to two dial-peers that both match a given inbound call, for example, 
and one is matching on 'incoming uri' and the other is matching on 'incoming 
called-number', for example, will the 'incoming uri' dial-peer always match 
first, regardless of the order it appears in the running config, or the 
dial-peer number?

It's the matches themselves that determine precedence in the matching order?

I note that you can put multiple matches within a dial-peer and only one needs 
to match, it's a shame that compound matches can't be built.

Gary

From: Johnson, Tim mailto:johns...@cmich.edu>>
Date: Friday, 11 November 2022 at 12:57
To: Gary Parker mailto:g.j.par...@lboro.ac.uk>>, voip 
puck mailto:cisco-voip@puck.nether.net>>
Subject: RE: [External] [cisco-voip] Voice Gateway Dial-Peer 
Precedence/Processing Order

** THIS MESSAGE ORIGINATED OUTSIDE LOUGHBOROUGH UNIVERSITY **

** Be wary of links or attachments, especially if the email is unsolicited or 
you don't recognise the sender's email address. **
I believe this is what you're looking for. Order is based on how you have your 
DNIS/ANI pattern configured.

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-dp.html#concept_1ACF9AAF93C24BB988E4A2EE3734C8A6

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Gary Parker
Sent: Friday, November 11, 2022 7:39 AM
To: voip puck mailto:cisco-voip@puck.nether.net>>
Subject: [External] [cisco-voip] Voice Gateway Dial-Peer Precedence/Processing 
Order

Hi folks, feel stupid asking what feels like a newbie question, but I can't 
seem to find an answer online anywhere and I 've never needed to worry about 
this in the past!

In what order are dial-peers checked for a match for calls passing through a 
voice gateway? Is it simply the order they appear in the running-config, does 
the dial-peer number play any part, or is there something else influencing it?

--
Gary Parker
Unified Communications Service Manager
Loughborough University, IT Services
Phone - +441509635635
Teams - g.j.par...@lboro.ac.uk
https://www.osx.ninja/pubkey.txt

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Re: [cisco-voip] [External] Voice Gateway Dial-Peer Precedence/Processing Order

[Johnson, Tim]

I believe this is what you're looking for. Order is based on how you have your 
DNIS/ANI pattern configured.

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-dp.html#concept_1ACF9AAF93C24BB988E4A2EE3734C8A6

From: cisco-voip  On Behalf Of Gary Parker
Sent: Friday, November 11, 2022 7:39 AM
To: voip puck 
Subject: [External] [cisco-voip] Voice Gateway Dial-Peer Precedence/Processing 
Order

Hi folks, feel stupid asking what feels like a newbie question, but I can't 
seem to find an answer online anywhere and I 've never needed to worry about 
this in the past!

In what order are dial-peers checked for a match for calls passing through a 
voice gateway? Is it simply the order they appear in the running-config, does 
the dial-peer number play any part, or is there something else influencing it?

--
Gary Parker
Unified Communications Service Manager
Loughborough University, IT Services
Phone - +441509635635
Teams - g.j.par...@lboro.ac.uk
https://www.osx.ninja/pubkey.txt

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Re: [cisco-voip] [External] RE: Webex App Config XML?

[Johnson, Tim]

I actually got a response back from my CSM and it sounds like these types of 
settings can be set on the back end, but through a provisioning ticket. My CSM 
is submitting a ticket to have the Ultrasound feature disabled. Whether that 
completely locks out the user from changing the setting, or provides 
flexibility, I don’t know but we don’t really have anyone using it with the 
handful of endpoints we have anyway.

I did try to use some of the Jabber parameters, but of course, that only 
functions when it is registered to CUCM. Correct, the EnableProximity parameter 
does not apply to Webex, from my testing.

From: Adam Pawlowski 
Sent: Wednesday, July 27, 2022 10:05 AM
To: Johnson, Tim ; cisco-voip@puck.nether.net
Subject: [External] RE: Webex App Config XML?

Hi Tim,

If it’s registered on premise, it will use the same mechanism to pull the 
configuration as Jabber, but it supports only a handful of parameters.

They’re listed here:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/wbxt/ucmcalling/unified-cm-wbx-teams-deployment-guide/unified-cm-wbx-teams-deployment-guide_appendix_0100.html#id_137099

They were pretty adamant they were not going to let the application control be 
as granular this time, as some of the options paint the thing into a corner 
depending on how they’re used.

Proximity Enable isn’t listed as one of them unfortunately.


Adam Pawlowski
Network Engineer | Network and Communication Services
University at Buffalo Information Technology (UBIT)
243 Computing Center, Buffalo, NY 14260
[contact_line]

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Johnson, Tim
Sent: Tuesday, July 26, 2022 5:02 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] Webex App Config XML?

Hello all,

We just recently migrated to the Webex App and have found some annoyances with 
the settings that are default. Is there a way to customize the settings for the 
Webex app globally, similar to how you could setup a jabber-config.xml file? 
The thing we’re specifically looking for right now is how to disable the “Use 
ultrasound” option under Devices, but I imagine there will be more items we’d 
like to manipulate.

TIA,
Tim Johnson
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[cisco-voip] Webex App Config XML?

[Johnson, Tim]

Hello all,

We just recently migrated to the Webex App and have found some annoyances with 
the settings that are default. Is there a way to customize the settings for the 
Webex app globally, similar to how you could setup a jabber-config.xml file? 
The thing we're specifically looking for right now is how to disable the "Use 
ultrasound" option under Devices, but I imagine there will be more items we'd 
like to manipulate.

TIA,
Tim Johnson
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Re: [cisco-voip] [External] CUCM SIP trunk redundancy with multiple CUBES

[Johnson, Tim]

I have the same setup, but I'm not sure we've run into that scenario. From 
Google, it looks like you could adjust the Route Plan settings under Service 
Parameters (Call Manager) to continue routing on user busy flag. The 
description of the option speaks to intracluster routing, but from a few forum 
posts, it sounds like it applies to regular SIP trunk routing also.

That said, with it sending back a 503, I'm not sure this is going to help. 
Curious what you figure out though.

From: cisco-voip  On Behalf Of Matthew Huff
Sent: Thursday, March 10, 2022 10:12 AM
To: cisco-voip@puck.nether.net
Subject: [External] [cisco-voip] CUCM SIP trunk redundancy with multiple CUBES

We have two cisco ISR 4331 Cube gateways in two different locations. I want to 
be able to route calls to both devices (preferably round-robin). I have the 
router pattern going to a trunk with both cubes defined (with sip options 
keep-alive configured). The issue we are having is that if the call is made to 
CUBE1 and the associated outbound dial-peer in in busyout, the CUBE returns 503 
 Service unavailable and CUCM doesn't try CUBE2. What am I missing?

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] [External] Re: SIP to iTSP best practices

[Johnson, Tim]

Better yet, just keep a PRI for this. It’ll save you many headaches.

From: cisco-voip  On Behalf Of Kent Roberts
Sent: Friday, February 11, 2022 12:14 PM
To: Matthew Huff ; cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] SIP to iTSP best practices


Oh yeah.. one more thing...

Test faxing  a fax test is a min of 10 pages, inbound call and out 
don't just do a page and say your good.  Check T38 if your using it... if you 
have to fail back because of T38 non-compliant, is G711 working?  Does your 
faxing software do/support switchback to 711 if T38 doesn't setup.

If you have a fax machine on a ATA or whater, test to it as well.



Isn't fax dead yet? :)   good luck with your go live.


On 2/11/22 8:52 AM, Matthew Huff wrote:
Thanks for the recommendations. I have a lot to dig into. Question about the 
video disable. We have no video hardware, so  think it would be good to disable 
it before we go live. What’s the best way to disable it globally?

Is it

Voice service voip
  Sip
 Audio forced

?

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | www.ox.com
...

From: Kent Roberts 
Sent: Thursday, February 10, 2022 6:14 PM
To: Matthew Huff ; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] SIP to iTSP best practices





I was part of the team that starting a large scale sip migration almost 10 
years ago.  Have moved thousand's of DID since then.  Run multiple gig circuits 
into the cube.

Recommendations:

on the link to your provider, use address outside of the route able block for 
your company.  (say you use 10.x.x.x  then use 172.16 or 192.168)  If you 
can, don't route the itsp connections on your company network, go direct to the 
routers supporting those links.  (BGP peers I would guess depending on 
carrier/build)   If you can use a dedicated router, unless is a small site  
This is important if you wind up doing any kind of call recording, or if you 
have to enable debugs during the day.

Use dedicated dial peers setup exactly for each itsp SBC link  for in and one 
for out.

Use something like the "voice class uri trunk(x) sip"  or equivalent to bind to 
the dial peers for each SBC.

This will help if you have to add additional carriers, or say acquire a 
company, or need to do special routing...

use full E164 to and from the carrier, they may only want to do 10 digit 
in/out, but that is easy enough.  (uri trunkx will help here, as the inbound 
number will be at the cube, then you can route to cucm with outbound dial peer)
From your CUCM still send the 9 or 8 or whatever for outbound, then strip on 
match in the dialpeer to Itsp.   This will keep call looping etc.

define your voice class codecs on the dialpeers... don't just assume it will 
take the default, or work as you want without it.

if the cube will never see VIDEO, disable the options.  The cube software likes 
to release bugs that cause the cube to go south with video errors.

Depending on your carrier, you may need to force G729 or G711 first, even if 
its not your preferred codec, have seen were the SBC will not negotiate a call, 
if the codecs aren't in the order the carriers SBC wants.

do not assume the carriers network will normalize the calls.  For instance, if 
the destination is on the same carrier, its a direct ip route via the SBC.  If 
that end side can't accept say G729 (cheaper sbc)  the call will just fail.

NEVER user debug ccsip all
debug CCSIP messages is safer, and unless your cube is peeked, it won't add 
to much cpu.

make sure your CPU never exceeds 80% at the max possible peek of routing.

Check how the calls work with MOH.   Inbound and out.  make sure 2 way audio 
remains after the on hold event..

Do you need to force early offer?   (yes sounds silly, but have run into issues 
where some phones had no audio unless this was set)

Ask your carrier, how they handle TFNs outbound, if you pass the ANI from a 3rd 
party. (this is all billing stuff to the TFN owner)
Some may allow calls to process not caring what the number is.
Some may allow you to provide a alternate billing number.
Some will just  603 decline the call if the ANI isn't in your number poll 
assigned to you.
with a 603 the cube will try the next dial peer so you can add a header 
to re-write this with your number.

Diversion headers exist, however most carriers pass them through to the 
destination, and IVRs or Voice Mail systems on the far side will try to process 
that information, and do unexpected things.  (the party your calling doesn't 
exist for example.)

define the default sip control/media source interface, this will be your 
destination from cucm.   

Re: [cisco-voip] [External] Unity Connection - External Calls cut at 49 seconds

[Johnson, Tim]

Tappinghead.gif

On Feb 3, 2022 11:07 PM, Hunter Fuller via cisco-voip 
 wrote:
Have you considered asking your clients to leave messages more quickly?

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

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

On Thu, Feb 3, 2022 at 10:20 AM Riley, Sean via cisco-voip
 wrote:
>
> Unity v 12.5 SU4 with no changes recently and 300 second max message length.
>
>
>
> Voice gateways are CUBEs running isr4300-universalk9.16.09.08.SPA.bin with no 
> changes recently.  2 separate sites having same issue.  Internal calls to 
> voicemail are fine.
>
>
>
> We just had this reported yesterday and I can recreate the issue easily.
>
>
>
> I am hoping someone has run across this or can point me in the right 
> direction.
>
>
>
> Thanks.
>
>
>
> Sean.
>
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Re: [cisco-voip] [External] Re: Small business E911 solution

[Johnson, Tim]

Maybe Intrado? Not sure their minimum requirements but I know we started with 
them when they were West, with only a couple hundred DIDs.

On Dec 9, 2021 2:16 PM, Matthew Huff  wrote:
No, hosted solution isn’t an option as we have a number of custom solutions 
like ring downs, etc…

We already have CUCM and Expressway working fine, I just need directions on the 
simplest solution for E911 for MRA workers.

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | www.ox.com
...

From: Matthew Loraditch 
Sent: Thursday, December 9, 2021 2:00 PM
To: Matthew Huff ; cisco-voip@puck.nether.net
Subject: RE: Small business E911 solution

I’m very curious if you find something. I’m not aware of anything cost 
effective at your size. RedSky’s minimum purchase for a CUCM based system is 
12-14k.

Have you looked at moving to a hosted phone system? Almost every vendor I’m 
aware of includes E911 therein



Matthew Loraditch​
Sr. Network Engineer
(He/Him/His)
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion Technologies]
[Facebook]
[Twitter]
[LinkedIn]
From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Matthew Huff
Sent: Thursday, December 9, 2021 1:34 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Small business E911 solution

[EXTERNAL]

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] [External] Re: UCCX 12.5 Real Time Reporting Tool

[Johnson, Tim]

After some trial and error, I was able to find that it was having an issue with 
temp files in Java. After deleting them, the reports would load properly. Ended 
up just disabling the option to keep temp files.

From: Bill Talley 
Sent: Thursday, November 18, 2021 3:05 PM
To: Johnson, Tim 
Cc: cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] UCCX 12.5 Real Time Reporting Tool

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




On Nov 18, 2021, at 12:58 PM, Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:

Is anyone using UCCX 12.5 SU1 ES01, and able to use the Real Time Reporting 
Tool (not RTMT)? I find it to be handy in troubleshooting on rare occasions, 
but as soon as I upgraded to 12.5 from 12.0 back in August, it stopped working 
for me on multiple clients. It launches, but whenever you attempt to open any 
of the reports it just hangs with a Windows spinning wheel. Once or twice, I’ve 
been able to load one report after I launch the tool but it will stop working 
if I try to look at a different report.

I opened a TAC case a little while ago but the engineer took me down all sorts 
of rabbit holes. I had completely reinstalled Java and it ended up loading but 
then stopped working again within a day after. I didn’t have the time or 
patience then to continue with TAC.

Tim Johnson
Voice & Video Engineer
Central Michigan University
Call me: +19897744406
Video Call me: johns...@cmich.edu
Fax me: +19897795900
Meet me: http://cmich.webex.com/meet/johns10t

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[cisco-voip] UCCX 12.5 Real Time Reporting Tool

[Johnson, Tim]

Is anyone using UCCX 12.5 SU1 ES01, and able to use the Real Time Reporting 
Tool (not RTMT)? I find it to be handy in troubleshooting on rare occasions, 
but as soon as I upgraded to 12.5 from 12.0 back in August, it stopped working 
for me on multiple clients. It launches, but whenever you attempt to open any 
of the reports it just hangs with a Windows spinning wheel. Once or twice, I've 
been able to load one report after I launch the tool but it will stop working 
if I try to look at a different report.

I opened a TAC case a little while ago but the engineer took me down all sorts 
of rabbit holes. I had completely reinstalled Java and it ended up loading but 
then stopped working again within a day after. I didn't have the time or 
patience then to continue with TAC.

Tim Johnson
Voice & Video Engineer
Central Michigan University
Call me: +19897744406
Video Call me: johns...@cmich.edu
Fax me: +19897795900
Meet me: http://cmich.webex.com/meet/johns10t

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Re: [cisco-voip] [External] Re: Jabber Users Prompted To Accept Webex Cert

[Johnson, Tim]

I’ve heard from my help desk that they had a few users report the prompt for 
accepting a cert. Unfortunately, they gathered zero details for me and just had 
the users accept the cert…

Good to know it’s not just us though.


From: cisco-voip  On Behalf Of Jason Aarons
Sent: Thursday, November 11, 2021 10:17 AM
To: Gary Parker 
Cc: cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] Jabber Users Prompted To Accept Webex Cert

Webex clients update switched from the Quovadis Root CA which was older and 
being retired, to the IdenTrust Root CA which it dates back to 2014. The 
IdenTrust Root CA certificate is contained within the default trust store of 
all major operating systems by default.

Not clear why IdenTrust is missing on your computers.

Guessing maybe you disabled automatic root updates at some point or don’t have 
Windows updates running ? 
https://serverfault.com/questions/752146/why-are-many-admins-using-turn-off-automatic-root-certificates-update-policy

Cisco Field Notice we didn’t notice
https://www.cisco.com/c/en/us/support/docs/field-notices/721/fn72120.html

On Thu, Nov 11, 2021 at 6:22 AM Gary Parker 
mailto:g.j.par...@lboro.ac.uk>> wrote:
Morning all, a few years back we had a problem where lots of our managed 
Windows service users were complaining that their Jabber clients had started 
rejecting a certificate offered by idbroker.webex.com

This thread on community.cisco.com 
(https://community.cisco.com/t5/unified-communications/jabber-idbroker-webex-com-certificate-request-during-the-first/td-p/3216376)
 showed we weren’t the only ones, but that it seemed limited to managed clients.

We solved this by adding the EXCLUDED_SERVICES=WEBEX flag to the installer on 
our managed clients.

Fast forward to today and we suddenly have a load of service desk cases from 
users again. Nothing has changed in our configuration of Jabber client, IM 
servers or expressways. The clients haven’t been updated recently, and this 
time we’re also seeing the “Certificate not valid” pop-up on unmanaged Windows 
machines as well as our managed service. The cert that’s being rejected has 
validity start date of late September, so it doesn’t appear to be a cert that’s 
only just been brought into use.

Is anyone else seeing this today?

As a workaround I’ve added:

WEBEX

...to our jabber-config.xml, but that will require users to manually reset 
their clients. Not sure why I hadn’t done earlier ¯\_(ツ)_/¯
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Re: [cisco-voip] [External] Error Processing SAML Response

[Johnson, Tim]

Just for the sake of sanity, all servers are using the same NTP server(s)? And 
if needed, before adjusting NTP, just remember that changing it can alter 
license MAC.

From: Jonathan Charles 
Sent: Friday, September 17, 2021 9:00 AM
To: Kent Roberts 
Cc: Johnson, Tim ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [External] Error Processing SAML Response

The error message in the Cisco traces (SSO) is:

2021-09-15 16:07:43,791 DEBUG [http-nio-81-exec-22] fappend.SamlLogger - 
SAML2Utils.checkConditions: NotOnOrAfter Condition = Wed Sep 15 22:07:44 UTC 
2021   -  this time is 17:07:44 CDT
2021-09-15 16:07:43,791 DEBUG [http-nio-81-exec-22] fappend.SamlLogger - 
SAML2Utils.checkConditions: NotBefore Condition = Wed Sep 15 21:07:44 UTC 2021  
-  this time is 16:07:44 CDT

2021-09-15 15:25:10,642 ERROR [http-nio-81-exec-10] 
authentication.SAMLAuthenticator - Error while processing saml response The 
time in the Assertion's Condition is invalid.
com.sun.identity.saml2.common.SAML2Exception: The time in the Assertion's 
Condition is invalid.

Basically what appears to be occurring is we get a NotBefore of 1 second after 
our request came in (16:07:43) and it gets killed

The real question is what they need to do on the ADFS side to fix this... why 
are they sending us a time in the future? The argument is NTP is off by one 
second for one of the servers (all of them show synched)...


Jonathan

On Thu, Sep 16, 2021 at 8:29 PM Kent Roberts 
mailto:k...@fredf.org>> wrote:
Oh, ok if I mis-understood then, yes a SAML trace would be good, as well as 
knowing is this new or did it work.   Seems similar to what I have seen in UCCE 
with the packet stuff not signed or wrong encryption type… course thats UCCE vs 
CUCM,  but usually cucm just works…



On Sep 16, 2021, at 6:45 PM, Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:

Nah, looks like he said logging into CCM Admin pages, with AD accounts, so all 
areas of the web UI (I believe). The NTP errors that I’ve seen are presented as 
SAML assertion errors.

I’m curious if this is a new SSO config, or if it was working properly and 
something’s changed.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Kent Roberts
Sent: Thursday, September 16, 2021 8:37 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [External] Re: [cisco-voip] Error Processing SAML Response

Remember he said it also was happening on the CUCM Admin account which has 
nothing to do with SSO/SAML.   So means its most likely internal to cucm...

On Sep 16, 2021, at 4:36 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

The logs are pretty clear when its a time difference as the error. I’ve not 
seen it randomly occur but definitely the error will be it’s time and may even 
show the difference.

Its the 4j log file for sso I believe

Get Outlook for iOS<https://aka.ms/o0ukef>

Matthew Loraditch​
Sr. Network Engineer
(He/Him/His)
p: 443.541.1518
w: www.heliontechnologies.com<http://www.heliontechnologies.com/>
 |
e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>
<http://www.heliontechnologies.com/>
<https://facebook.com/heliontech>
<https://twitter.com/heliontech>
<https://www.linkedin.com/company/helion-technologies>

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Thursday, September 16, 2021 4:32:12 PM
To: Jonathan Charles mailto:jonv...@gmail.com>>; Benjamin 
Turner mailto:benmtur...@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Error Processing SAML Response


[EXTERNAL]


Have you been able to confirm the time difference?

I’m not trying to take their side of things, but if it’s minutes off, I 
wouldn’t doubt that’s possible. SSO is highly secure, right? A time difference 
might be enough to throw it off?

Here’s  reference:

https://support.pingidentity.com/s/article/Accounting-for-Time-Drift-Between-SAML-Endpoints50907



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Thursday, September 16, 2021 6:23 PM
To: Benjamin Turner mailto:benmtur...@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Error Processing SAML Response

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<mailto:ith...@uoguelph.ca>

No... TBH, I have never heard of it...

TAC is hyper-asserting that the issue

Re: [cisco-voip] [External] Re: Error Processing SAML Response

[Johnson, Tim]

Nah, looks like he said logging into CCM Admin pages, with AD accounts, so all 
areas of the web UI (I believe). The NTP errors that I’ve seen are presented as 
SAML assertion errors.

I’m curious if this is a new SSO config, or if it was working properly and 
something’s changed.

From: cisco-voip  On Behalf Of Kent Roberts
Sent: Thursday, September 16, 2021 8:37 PM
To: Matthew Loraditch 
Cc: cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] Error Processing SAML Response

Remember he said it also was happening on the CUCM Admin account which has 
nothing to do with SSO/SAML.   So means its most likely internal to cucm...


On Sep 16, 2021, at 4:36 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

The logs are pretty clear when its a time difference as the error. I’ve not 
seen it randomly occur but definitely the error will be it’s time and may even 
show the difference.

Its the 4j log file for sso I believe

Get Outlook for iOS

Matthew Loraditch​
Sr. Network Engineer
(He/Him/His)
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Thursday, September 16, 2021 4:32:12 PM
To: Jonathan Charles mailto:jonv...@gmail.com>>; Benjamin 
Turner mailto:benmtur...@hotmail.com>>
Cc: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Error Processing SAML Response


[EXTERNAL]


Have you been able to confirm the time difference?


I’m not trying to take their side of things, but if it’s minutes off, I 
wouldn’t doubt that’s possible. SSO is highly secure, right? A time difference 
might be enough to throw it off?


Here’s  reference:


https://support.pingidentity.com/s/article/Accounting-for-Time-Drift-Between-SAML-Endpoints50907






From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Thursday, September 16, 2021 6:23 PM
To: Benjamin Turner mailto:benmtur...@hotmail.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Error Processing SAML Response


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


No... TBH, I have never heard of it...


TAC is hyper-asserting that the issue is time mismatch between CUCM/CUC and 
ADFS...




Jonathan


On Thu, Sep 16, 2021 at 4:08 PM Benjamin Turner 
mailto:benmtur...@hotmail.com>> wrote:
Have you tried to run a SAML Tracer?


Sincerely,
Benjamin M. Turner

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Sent: Thursday, September 16, 2021 4:56:48 PM
To: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Error Processing SAML Response


So, users are randomly getting the above error when logging into CUCM UCMUser 
or CUC Inbox... we are also getting it using AD credentials into admin pages 
for CUCM/CUC/etc.


For a user, it will work find repeatedly, then you will get the error, close 
your browser, and reopen, still get the error for a few minutes. Then later it 
will work. When a user is affected, other users work fine.


TAC is saying it is an NTP issue, however, NTP between CUCM 12.5 and IdP (ADFS 
2.0) is fine.


Pings are around 1ms between servers.


Any ideas?




Jonathan






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Re: [cisco-voip] [External] Re: Error upgrade UCS server to ESXi 6.7

[Johnson, Tim]

Thank you for this! I'll give it a shot! I was recently trying the same upgrade 
on some Hyperflex nodes.

From: cisco-voip  On Behalf Of Ryan Huff
Sent: Monday, September 13, 2021 3:52 PM
To: Jonathan Charles ; cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] Error upgrade UCS server to ESXi 6.7

It's possible the vibs aren't in use, making this a harmless process, but I'd 
verify first. There is a nice walk through here:

https://thesleepyadmins.com/2020/08/06/vmware-esxi-6-7-upgrade-missing-dependency-vibs-error/

Get Outlook for iOS

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Sent: Monday, September 13, 2021 3:44:12 PM
To: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Error upgrade UCS server to ESXi 6.7

Getting the following error when trying to upgrade a C240 M5 to VMware 6.7 from 
6.5

[image.png]

How safe is it to just delete these?

I was going to do a esxcli software vib remove -vibname ucs-tool-esxi

I just want to find some documentation that this is safe to do and won't delete 
data or blow up the box...

I did an audit of our UCS C240s and this version of UCS-tools is installed 
everywhere, so I am expecting to get the same error when I upgrade those.


Thanks!

Jonathan Charles
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Re: [cisco-voip] [External] Re: Faxing over CUBE SIP Trunks

[Johnson, Tim]

What's the "go to" for an on-prem fax server these days? I've heard Imagicle is 
good.

On Jun 21, 2021 10:24 PM, Pete Brown  wrote:
XMedius…

[cid:image001.png@01D766E3.8B38BE00]

I still remember our CIO screaming at them on a call (veins popping and all) 
after our implementation.  Some poor fella had to fly in from Canada the next 
day to smooth things over.

Definitely second the 9600 baud part.  Solved a few problems for us.


From: cisco-voip 
cisco-voip-boun...@puck.nether.net 
On Behalf Of Kent Roberts
Sent: Monday, June 21, 2021 6:31 PM
To: JASON BURWELL 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Faxing over CUBE SIP Trunks

sure.   This is what we had to do to get fax working for use on both vg’s. 
(202/322/fxs) and Rightfax/brook trout cards)
Rightfax has the g711 failback enabled.   IE it controls it not the cube.


Under voip service voip..

Fax protocol t38 version 0 ls-redundancy 0 hs-redundancy-fallback none

sip
Midcall-signaling passthru
header-passing

And the dial-peers towards the carriers:
fax-relay ecm disable
fax rate 14400



Good luck.  Fax is an easy but fun/tough thing to make right. Put 
Wireshark on your server, it helps. Also, ,might need to force the max fax 
rate to 9600.



On Jun 21, 2021, at 12:26 PM, JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

Can anyone offer some guidance doing faxing over SIP Trunks? Ever since 
switching to CUBE SIP trunks I’ve been having challenges with outbound faxes. 
With one provider, it rarely works, the other it works most of the time but is 
not reliable. Sometime it fails and I resend it and works fine. I have an 
Xmedius Fax Server using T38 with G711 fallback. Feel like I am missing 
something, I’ve had a few cases open and a mix of fingerpointing 
(XMedius/Cisco/Provider) and dead ends but no solid resolution. Just figured 
I’d ask out here before I escalate these cases again. Both providers use G711 
codec on the trunks. When I point the calls pack to MGCP PRI I have no issues.

Thanks
Jason

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Re: [cisco-voip] [External] Re: Faxing over CUBE SIP Trunks

[Johnson, Tim]

Curious what provider you’re using. Right now I’m using XMedius out to Frontier 
and I’d say it works about 95% of the time for outbound, though the majority of 
our outbound goes through VG202XMs. I know it’s typically recommended to use 
the same signaling through the call flow, so run SIP all the way through on 
your end. Avoid MTP or transcoding.

We have the following set on our dial peers:
fax-relay ecm disable
fax-relay sg3-to-g3
fax-relay ans-disable
fax rate 14400
fax nsf 00
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw

Based on the experience, it doesn’t seem like any of them actually do anything 
other than the fax protocol.

From: cisco-voip  On Behalf Of Jason Aarons
Sent: Monday, June 21, 2021 4:26 PM
To: JASON BURWELL 
Cc: cisco-voip 
Subject: [External] Re: [cisco-voip] Faxing over CUBE SIP Trunks

The providers should support t38 to be reliable.

On Mon, Jun 21, 2021, 2:27 PM JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:
Can anyone offer some guidance doing faxing over SIP Trunks? Ever since 
switching to CUBE SIP trunks I’ve been having challenges with outbound faxes. 
With one provider, it rarely works, the other it works most of the time but is 
not reliable. Sometime it fails and I resend it and works fine. I have an 
Xmedius Fax Server using T38 with G711 fallback. Feel like I am missing 
something, I’ve had a few cases open and a mix of fingerpointing 
(XMedius/Cisco/Provider) and dead ends but no solid resolution. Just figured 
I’d ask out here before I escalate these cases again. Both providers use G711 
codec on the trunks. When I point the calls pack to MGCP PRI I have no issues.

Thanks
Jason

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Re: [cisco-voip] [External] Re: Sip profiles

[Johnson, Tim]

I'm not the most experienced with SIP profiles, so I won't comment on that. 
Maybe you could use translation profiles on the cube instead, to change the 
calling party to something else?

-Original Message-
From: cisco-voip  On Behalf Of 
f...@browardcommunications.com
Sent: Monday, February 1, 2021 10:48 AM
To: cisco-voip@puck.nether.net
Subject: [External] Re: [cisco-voip] Sip profiles



Sent from my iPhone

> On Jan 29, 2021, at 4:22 PM, 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] Non-jabber based Cisco softphones for iPhone and Android

[Johnson, Tim]

Pretty sure you can run Jabber in phone-only mode these days. Not sure on the 
limitations of it though.

-Original Message-
From: cisco-voip  On Behalf Of Matthew Huff
Sent: Monday, February 1, 2021 10:56 AM
To: cisco-voip@puck.nether.net
Subject: [External] [cisco-voip] Non-jabber based Cisco softphones for iPhone 
and Android

We are looking for a solution for softphones that are compatible with 
expressway for both iPhones and android. Jabber is not an option due to Finra & 
SEC compliance (no instant messaging). Specifically we are looking for a voice 
app only with no collaboration/presence features. Any suggestions?
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Re: [cisco-voip] [External] RE: Third Party Softphone w/ TLS

[Johnson, Tim]

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 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Kent Roberts
Sent: Thursday, January 21, 2021 6:35 PM
To: Johnson, Tim mailto:johns...@cmich.edu>>
Cc: cisco-voip@puck.nether.net<mailto: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 
mailto:johns...@cmich.edu>> 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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[cisco-voip] Third Party Softphone w/ TLS

[Johnson, Tim]

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] [EXTERNAL] Remote UCCX users

[Johnson, Tim]

How is NICE in comparison? I’ve been curious about other products but haven’t 
looked too far yet.

From: cisco-voip  On Behalf Of Croft, Keith
Sent: Tuesday, September 8, 2020 7:59 AM
To: Jonatan Quezada ; James Dust 
; Adrian Arevalo-Orozco 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] [EXTERNAL] Remote UCCX users

We used 8811 IP Phones registered over MRA with the users personal line. Then 
Extension Mobility - place the remote ACD line as line1 and the desk phone as 
line 2 and finally got this working in our environment.
We also had success  CSF / Jabber with ACD as line1 and personal line as line 
2. Then add the mac for  CSF profile to the RMCM controlled devices application 
user.

We have since switched to NICE Incontact and deprecated UCCX from our 
environment.

-Keith
From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>>
Date: Tuesday, August 25, 2020 at 4:36 PM
To: James Dust 
mailto:james.d...@charles-stanley.co.uk>>, 
Adrian Arevalo-Orozco 
mailto:adrian.arevalo.oro...@chemeketa.edu>>
Cc: "cisco-voip@puck.nether.net" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] [EXTERNAL] Remote UCCX users

Is this where we add the mac for  CSF profile to the RMCM controlled devices 
application user?

please confirm, I feel like we did this recently for my help desk users also 
running a queue with finesse on my ccx

On Tue, Aug 25, 2020 at 2:29 PM James Dust 
mailto:james.d...@charles-stanley.co.uk>> 
wrote:
I have a number of users working remotely who are part of a queue on UCCX,

They are using jabber on their windows desktop (csf profile on cucm) and are 
connecting into our network via our VPN.

The users would normally just take calls on their desk phones in our office, 
but are working remotely due to COVID.

Thing is they are only able to receive calls when logged into their desk 
phone’s in the office (even though they are fielding the calls via jabber 
whilst working remotely)

Is there any way around this dependancy?

Consider the environment - Think before you print

The contents of this email are confidential to the intended recipient and may 
not be disclosed. Although it is believed that this email and any attachments 
are virus free, it is the responsibility of the recipient to confirm this.

You are advised that urgent, time-sensitive communications should not be sent 
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Details of Charles Stanley group companies and their regulators (where 
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--
Jabber and 
Instructions

Johnny Q
Voice Technology Analyst II
Chemeketa Community College
johnn...@chemeketa.edu
Building 22 Room 130
Work/Jabber 5033995294
Cell 5035769873
FAX 5033995549

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Re: [cisco-voip] [External] Re: Remote Phone Control

[Johnson, Tim]

[The Simpsons: The 10 Worst Things Mr. Burns Has Ever Done, Ranked]

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Friday, August 21, 2020 4:27 PM
To: Hunter Fuller 
Cc: Cisco VoIP Group 
Subject: Re: [cisco-voip] [External] Re: Remote Phone Control

And you would never upload a rick roll audio clip to your CUCM as a mutlitcast 
audio source, and then abuse the join rtp stream function on your co-workers 
phones either.  No...no you wouldn't.  And neither have I.  ;)

On Fri, Aug 21, 2020 at 3:07 PM Hunter Fuller 
mailto:hf0...@uah.edu>> wrote:
Im pretty sure this just completely changed the way we provide remote
help in the COVID era. Since I can just add a customer's phone to my
controlled devices, help them fix/show me some problem remotely, and
then remove it.

--
Hunter Fuller (they)
Router Jockey
VBH Annex B-5
+1 256 824 5331

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

On Fri, Aug 21, 2020 at 3:04 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
>
> It’s a great tool.
>
>
> On Fri, Aug 21, 2020 at 9:13 AM Anthony Holloway 
> mailto:avholloway%2bcisco-v...@gmail.com>> 
> wrote:
>>
>> My add-on was approved to be in the add-on store: 
>> https://addons.mozilla.org/addon/cisco-phone-controller/
>>
>> On Fri, Aug 14, 2020, 11:38 AM Anthony Holloway 
>> mailto:avholloway%2bcisco-v...@gmail.com>> 
>> wrote:
>>>
>>> I published a phone control firefox add-on that I've been sitting on for 
>>> over a decade now: it's not super polished, because it's just for me and my 
>>> friends to use, but I thought, what the hell, make it available publicly. I 
>>> might even polish it up and list it in the add-on store one day. Until 
>>> then: https://github.com/avholloway/cisco-phone-controller
>>>
>>>
>>
>>
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Re: [cisco-voip] [External] IPCC best practice

[Johnson, Tim]

It seems to me that there's not a "best practice" label for most scenarios. 
When I started with UCCX, we went to a call handler first to provide us with an 
easy way to provide a schedule, and a familiar way for the customer to record a 
greeting. Later, we ended up building the schedule into our script and 
directing calls to the trigger. That's my preference, just to involve less 
systems. 

Tim Johnson
Voice & Video Engineer
Central Michigan University
Call me: +19897744406
Video Call me: johns...@cmich.edu
Fax me: +19897795900
Meet me: http://cmich.webex.com/meet/johns10t


-Original Message-
From: cisco-voip  On Behalf Of 
f...@browardcommunications.com
Sent: Wednesday, August 19, 2020 8:19 AM
To: cisco-voip@puck.nether.net
Subject: [External] [cisco-voip] IPCC best practice


Hello, I just have a quick question.
When setting up a call center for a SMB, Is it best practice to have the main 
number go to a unity call handler 1st, with caller input going to uccx 
triggers, or is it considered best practice to have the main number go right to 
CCX?  I have seen both ways.

Thank you.
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Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

[Johnson, Tim]

That bothers me too. I think the only option where that makes sense is for RGs 
and RLs because they cannot share the same name.


From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Thursday, May 28, 2020 3:42 PM
To: Matthew Loraditch 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

Can anyone give a use case where appending or prepending the object type 
identifier on the name is helpful?  E.g., why put -css on a css at all?

On Thu, May 28, 2020 at 2:19 PM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

  1.  PT-, CSS-, etc
  2.  FQDN
  3.  My setups are always distributed. Certainly could have central if it’s 
one site.
  4.  Usually always
  5.  SIP, SIP, SIP
  6.  Unfortunately no, drives my OCD crazy. I hate lower/mixed case naming of 
devices with a passion. I’m also born of a Windows world where case never 
mattered.



Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion Technologies]
[Facebook]
[Twitter]
[LinkedIn]
From: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Thursday, May 28, 2020 3:08 PM
To: James B mailto:james.buchan...@gmail.com>>; 
Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>; 
Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

[EXTERNAL]

Sure I can fire this up


  1.  part_ , like part_local, part_ld, part_ld_privacy , etc.
  2.  FQDN, but, make sure your DNS/NTP/etc works with resiliency.
  3.  Depends on if you’re distributed, using hardware conf, transcoder cause 
you have some people on some sort of twizzler based connection using g729, etc
  4.  Yes, unless you have something on the other side that can’t handle these 
requests coming from the whole group, or again a distributed system.
  5.  SIP. MGCP is nice in a set it and forget it way, but if you want to use 
the gateway to do anything else like custom intercepts, redirection, 
hairpinning, it won’t help you. There are some features that don’t work when 
you go to SIP but whatever.
  6.  Why would you give anything an upper case hostname

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of James B
Sent: Thursday, May 28, 2020 2:28 PM
To: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>; 
Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

I was thinking of the community configuration approach Anthony suggested and 
was thinking of the debates we’d have if we did that:


  1.  Do you use “_PT”, “PT_”, or just the site name? Same for “CSS”, “LOC”, 
and the ever-debated “RGN” or “REG”?.
  2.  FQDN or IP addresses?
  3.  Do all the media resources go into a single MRG or not?
  4.  Do we click “Run on all Nodes” for route lists and trunks or not?
  5.  MGCP, SIP, or H323 (if using PRIs)?
  6.  Can UCCX have upper-case hostnames or not?

The debates would take us so long, version 14.0 would be out, and then we’d 
have to debate about whether a “.0” versoin is stable or not or should we wait 
for “.5”? Still, could be fun!



From: Anthony Holloway
Sent: 28 May 2020 19:15
To: Matthew Loraditch
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Migration from CUCM/UCXN/UCCX 8.5 to 12.5

Keep in mind that PCD network migrations, while awesome for CUCM, do not work 
for other products.

Typically with a project like this, you'll likely have a different approach for 
each app, and not a one size fits all solution.

With the app upgrades, you will also have to change OVA sizes (or want to in 
some cases), and at that point, it might be better to install fresh, and use 
tools like COBRAS, BAT, AXL, ADMIN API, stare & compare, etc. to get data 
exported/imported from old to new.

Or like Kent said, tell yourself it's a toshiba system, and treat it like a 
greenfield.

On Thu, May 28, 2020 at 11:26 AM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
Here’s a fun one. We have taken over support of these ancient servers hosted on 
Esxi 4.1 on UCS-C200-M2s!

Exact Versions are:
8.5 SU2 for CUCM/UCXN
8.5 FCS for UCCX

Each  is a pair of servers.

Have new M5s and flex licensing… need to get to 12.5..  8.5 docs are dead for 
CUCM/UCXN and 8 and 9 docs are dead for UCCX. ISOs I may need are not available 
publicly.

Also fun wrinkle the new host are across the WAN and for many 

Re: [cisco-voip] Which phone is this?

[Johnson, Tim]

Looks like an 8841 with maybe a sticker?

On Apr 14, 2020 9:50 PM, Lelio Fulgenzi  wrote:
I’m watching The Tunnel, Season 3 Episode 1 (a great show by the way) and see 
this phone. Which looks like it has mwi/ringer light on the back? What model is 
this?

[cid:80C6646A-94BC-4543-AF17-AE5DEDE993AA-L0-001]

Sent from my iPhone

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Re: [cisco-voip] Jabber and Finesse

[Johnson, Tim]

Looks like Terry should be okay with that setting, being on 12.5, but for 
others who may try this setting, be aware of this bug: 
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvh48379/?reffering_site=dumpcr

I just ran into it on Monday, being on CUCM 12.0(1.1.10).

From: cisco-voip  On Behalf Of Pawlowski, 
Adam
Sent: Thursday, March 19, 2020 8:56 AM
To: 'Terry Oakley' ; jcolon...@gmail.com
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber and Finesse

Hi Terry - there's one other element to the setup that needs to be set in the 
VCS if it's not.

Maybe this is it?

Step 1
Go to VCS-C.
Step 2
Select VSC-C configuration > Unified Communication > Configuration > SIP Path 
headers and set it to On.

There's not really anything too substantial to configure otherwise. The CSF 
needs to be in the rmjtapi or rmcm or whatever application user's associated 
devices list, and then it "just works"

If you're getting that call cannot be completed message then it almost sounds 
like that line isn't registered and it has no other actions like VM or 
forwarding, which it shouldn't.

Adam

From: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>
Sent: Wednesday, March 18, 2020 10:25 PM
To: jcolon...@gmail.com; Pawlowski, Adam 
mailto:aj...@buffalo.edu>>
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Jabber and Finesse

Thank you for the quick replies.   I know all of you are undergoing immense 
pressure so I truly appreciate the assistance.   I have triple checked that the 
UCCX extension is just on the Jabber Windows client.  When I try and dial the 
extension I get the nice Cisco lady telling me the number cannot be completed 
as dialed.   If I dial the primary extension on the Jabber client it works.
If I put the UCCX extension on a physical set (8851) it will ring.

When I am on the Jabber Windows client I have checked the CSS for the UCCX 
extension it is fine, same as the primary line.  Double checked to make sure 
the extension was an active number. Allow Control of Device from CTI is 
enabled.   There must be some little check box or something that I have missed 
but I have stared at the page so long it all looks the same.

Thanks again

Terry




From: Jose Colon II mailto:jcolon...@gmail.com>>
Sent: Wednesday, March 18, 2020 4:38:26 PM
To: Pawlowski, Adam mailto:aj...@buffalo.edu>>
Cc: Terry Oakley mailto:terry.oak...@rdc.ab.ca>>; 
cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Jabber and Finesse

CAUTION: This email is from an external source. Do not click links or open 
attachments unless you recognize the sender and know the content is safe.
I think that is the key to the issue. UCCX extension can only be registered to 
one device.

On Wed, Mar 18, 2020 at 5:37 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:

Hi Terry,



I had the same problem when I had my CCX extension on multiple items, even when 
unregistered. Clicking on ready resulted in an error, but the first time I made 
a call with it by opening the keypad it started working and I could go ready. 
Since the CCX extension is just an extension, you should be able to dial it 
regardless of what Finesse is doing, assuming it is in a partition that you can 
dial but it may not be.



After I made sure the extension was on nothing but my Jabber client, and I had 
signed out and back in, it began to work fine.



I haven't heard any comments from anyone else and we moved ~75 seats to Jabber 
MRA and Finesse remote this week.



Adam



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Terry Oakley via cisco-voip
Sent: Wednesday, March 18, 2020 6:29 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber and Finesse



We have on prem CUCM running 12.5.1.   We also have IM and Presence and UCCX 
for our phone queues.   I am trying to figure out if I can move our phone 
queues to Jabber and connect to Finesse via remote access (through VM Ware).
I seem to be able to get part way but when I try to make a call to the queue 
the Finesse line will not answer and unless I go off hook first on the Jabber 
app I cannot go to Ready on the Finesse side .   I cannot even dial it just 
directly.   I can use that line and dial out from Jabber but for some reason I 
cannot get the line to be recognized on the Finesse side.   I am sure I 
probably missed something in my haste so if anyone of you have successfully 
done something like this I would appreciate a simple how to.



I hope all of you are safe and your families as well.



Terry





Terry Oakley

Telecommunications Coordinator | Information Technology Services

Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5

work (403) 342-3521   |  FAX (403) 343-4034

[cisco-voip] CMS Loopback Space

[Johnson, Tim]

Does anyone know if it's possible to create a loopback test space within Cisco 
Meeting Server, and maybe how to do it? Or if not, what might be used to create 
one? Although it's fun to just use the Blue Jeans parrot, I've always thought 
it would be cool to host my own.

Tim Johnson
Voice & Video Engineer
Central Michigan University
Call me: +19897744406
Video Call me: johns...@cmich.edu
Fax me: +19897795900
Meet me: http://cmich.webex.com/meet/johns10t

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Re: [cisco-voip] CER Configuration - Guides?

[Johnson, Tim]

I just worked my way through the official configuration guide the other day. It 
was a challenge, but I got it pieced together. The call flow diagram on this 
TAC document would have been super helpful to have!

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Wednesday, February 19, 2020 1:07 PM
To: Matt Taber (mtaber) 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

Thank you, that is super useful


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion Technologies]
[Facebook]
[Twitter]
[LinkedIn]
From: Matt Taber (mtaber) mailto:mta...@cisco.com>>
Sent: Wednesday, February 19, 2020 1:01 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CER Configuration - Guides?

[EXTERNAL]

This guide is a bit more streamlined, with configuration screenshots taken from 
an example deployment:
https://www.cisco.com/c/en/us/support/docs/unified-communications/emergency-responder/211453-Cisco-Emergency-Responder-Integration-wi.html

-Matt

On Feb 19, 2020, at 12:53 PM, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

Any good guides or blogs that go through this in a more real world way than the 
documentation?

I haven’t found anything super helpful yet.

Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com




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Re: [cisco-voip] HTTP Response for HTTP Triggered Script

[Johnson, Tim]

Bingo bango bongo.

From: Matthew Loraditch 
Sent: Tuesday, January 28, 2020 10:40 AM
To: Anthony Holloway ; Johnson, Tim 

Cc: Cisco VoIP Group 
Subject: RE: [cisco-voip] HTTP Response for HTTP Triggered Script

So basically I need 3 separate applications. The HTTP Trigger to take the input 
then pass the input to another application that then places the call into the 
queue that’s in a third application.


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com<http://www.heliontechnologies.com/>
 |
e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>
[Helion Technologies]<http://www.heliontechnologies.com/>
[Facebook]<https://facebook.com/heliontech>
[Twitter]<https://twitter.com/heliontech>
[LinkedIn]<https://www.linkedin.com/company/helion-technologies>
[cid:image006.jpg@01D5D5C7.81067570]
From: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Sent: Tuesday, January 28, 2020 10:24 AM
To: Johnson, Tim mailto:johns...@cmich.edu>>
Cc: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>; 
Cisco VoIP Group mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] HTTP Response for HTTP Triggered Script

[EXTERNAL]

Tim is correct: this is normal from my experience too.

I feel partly responsible for your woes, since I think you found that solution 
from me pointing out the script repository online.

Thinking about it some more, I think I would have the HTTP Triggered 
Application use the Trigger Application step in async mode, to start the 
initial callback feature, and then it can just End almost immediately.

I through together this quick and dirty MS Paint representation of it:

[cid:image007.png@01D5D5C7.81067570]

There's like a dozen ways to solve this, and this is just one of them.  Some 
details are left out of the drawing on purpose, so if you get stuck, let's keep 
this thread going.


On Tue, Jan 28, 2020, 7:14 AM Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:
As far as my experience, this is “normal” for UCCX. Yes, I believe the request 
stays open until the script ends. I believe we opened up a TAC case about it a 
year or two ago but I don’t believe it went anywhere.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Matthew Loraditch
Sent: Tuesday, January 28, 2020 8:06 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] HTTP Response for HTTP Triggered Script

Got the following comment from the developer who is writing an app that will 
queue a call via http trigger:
It seems like the MobileApp integration is accepting my request, spits back 
some HTML, but the HTTP request remains open until the queued call is answered.

Ideally, that HTTP request would be closed with a successful HTTP status code 
(e.g. HTTP 200) when the call was queued, so that I can report back to the user 
that they can wait for a callback.

I understand his problem, but I see no way  of fixing it. The send http 
response command doesn’t seem to have any settings but the doc sent back. Is 
this normal behavior or can I do something with the doc being sent back to make 
it work?


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com<http://www.heliontechnologies.com/>
 |
e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>
 <http://www.heliontechnologies.com/>
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Re: [cisco-voip] HTTP Response for HTTP Triggered Script

[Johnson, Tim]

As far as my experience, this is “normal” for UCCX. Yes, I believe the request 
stays open until the script ends. I believe we opened up a TAC case about it a 
year or two ago but I don’t believe it went anywhere.

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Tuesday, January 28, 2020 8:06 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] HTTP Response for HTTP Triggered Script

Got the following comment from the developer who is writing an app that will 
queue a call via http trigger:
It seems like the MobileApp integration is accepting my request, spits back 
some HTML, but the HTTP request remains open until the queued call is answered.

Ideally, that HTTP request would be closed with a successful HTTP status code 
(e.g. HTTP 200) when the call was queued, so that I can report back to the user 
that they can wait for a callback.

I understand his problem, but I see no way  of fixing it. The send http 
response command doesn’t seem to have any settings but the doc sent back. Is 
this normal behavior or can I do something with the doc being sent back to make 
it work?


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion Technologies]
[Facebook]
[Twitter]
[LinkedIn]
[cid:image006.jpg@01D5D5B2.75C99DA0]

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

[Johnson, Tim]

Yes, it’s bad. I typically download the PDF if I need to Find something.

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, January 21, 2020 1:37 PM
To: Anthony Holloway ; Cisco VoIP Group 

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

I think it has to do with results in non-visible text, i.e. the code itself.

But, yes, I have experienced this. Sometimes, using a different browser helps.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Tuesday, January 21, 2020 11:57 AM
To: Cisco VoIP Group 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Cisco's Website and Find on Page (CTRL+F) Issue

What's the problem?

Go to this page:
https://www.cisco.com/c/en/us/support/docs/cloud-systems-management/smart-call-home/119144-config-sch-00.html

Then CTRL+F for the search term: cert

There should be 21 matches, and not a single one of them are actually rendered 
on the page.

I was trying to figure out which certs on CUCM are for Smart Call Home, because 
we (and some of you too) have a VeriSign cert expiring next month, and I wanted 
to double check that it was for smart call home.

Anyway, the post is about the website functionality, and not smart call home, 
you could easily run into this issue on any page and with any search term.  If 
you're using Chrome, you may have experienced the issue where you can see that 
there are matches, but you cannot cycle through to any of them; it appears suck.

Does anyone else deal with this?  If so, let's get it fixed.

I already submitted this via the feedback link at the bottom of the page, but 
if one of you knows how to work around this issue, that would be great.

Through some basic web skills, I was able to determine that the search terms 
are in the source code for the page, but inside of elements which are hidden 
via CSS; like the drop down menus.

Also, if you can confirm the issue, and then submit the same feedback, maybe 
through sheer volume, they'll get their web team to fix it sooner rather than 
later.

You just need to click the Feedback link at the very bottom of the page, and 
I'll even paste some copy here for you, just in case you didn't want to write 
your own:

---

Hello Web Team!

When I and several others perform research in the course of performing our job 
duties, we are often in need of searching within your documentation by way of 
using the browser's find on page feature (CTRL+F).

However, due to the way in which your web page is structured, we are hindered 
from finding valuable content, because non-visible content is being searched; 
such as drop down menus.

Example: Navigate to:

https://www.cisco.com/c/en/us/support/docs/cloud-systems-management/smart-call-home/119144-config-sch-00.html

Press CTRL+F, followed by typing: cert

You should see that there are 21 matches, but none of them appear anywhere on 
the page.  That is just one such example, and this happens on many, if not all 
of your pages.

Your consideration for this issue is greatly appreciated.  Thank you.

---

Thanks Cisco VoIPers!
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Re: [cisco-voip] 988 Suicide Hotline

[Johnson, Tim]

Oh hey, 989, that’s me! :)

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, December 13, 2019 9:16 AM
To: cisco-voip voyp list 
Subject: Re: [cisco-voip] 988 Suicide Hotline

Pretty sure the dial plan would need to be updated to be able to use the router 
filter service clause. It’s always been advertised as X11.

It’s the urgent priority that I’m hoping will override anything else. It makes 
sense it would.

I will say this though, there are gonna be a bunch of misdials in certain 
areas….

980 area code - North Carolina
984 area code - North Carolina
985 area code - Louisiana
986 area code - Idaho
989 area code - Michigan


---
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]

From: James Buchanan 
mailto:james.buchan...@gmail.com>>
Sent: Friday, December 13, 2019 3:20 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: cisco-voip voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] 988 Suicide Hotline

I think you could add this to a route filter nonetheless but believe the 988 
will override if you give it urgent priority.

On Friday, December 13, 2019, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

I’m hoping a simple 988 route pattern will work here and 9.@ route patterns 
won’t interfere. Otherwise, would we see a COP file update sometime?

https://www.bostonglobe.com/2019/12/12/business/fcc-votes-set-up-3-digit-988-suicide-hotline-number/
-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 | 
le...@uoguelph.ca

www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
Twitter and Facebook


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Re: [cisco-voip] Server-groups and failover...

[Johnson, Tim]

What’s your dial peer configuration look like? Curious if you have ‘huntstop’ 
configured.

From: cisco-voip  On Behalf Of Pawlowski, 
Adam
Sent: Tuesday, November 19, 2019 8:07 AM
To: 'Jonathan Charles' ; Anthony Holloway 

Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Server-groups and failover...

I think I had to make some adjustments to our timers as well to get this to 
work before network timeout or similar:

sip-ua
retry invite 2
timers trying 100
!

I know I also goofed this up between dial peer group and server group, one of 
the two will retry within the group, the other sure doesn’t.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Monday, November 18, 2019 11:56 PM
To: Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Server-groups and failover...

I pasted the wrong part of the script (to manually change it)...

Here is the actual config:



voice class server-group 1
 ipv4 172.31.120.43
 ipv4 172.31.125.43 preference 2
 description Verizon SIP
!

Jonathan

On Mon, Nov 18, 2019 at 10:22 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
First off, I'm wondering why it says "no ipv4" in front of your two addresses.  
That might be your problem right there.

Secondly, I'd recommend putting an explicit preference on your entries, it's 
just better for everyone, and you don't get a credit back from Cisco for saving 
on a few ascii characters by implicitly using the default.  Plus, if the 
default is 0, which it is, then your next preference should be technically 1.  
But then having nothing and 1 seems silly, because if pref 1 is actually pref 
2, then well, might as well call them pref nothing and pref 8.  I digress.

You might not have failed over, because you might not have provided the system 
with the correction conditions to failover...E.g., you didn't wait long enough.

No seriously, by default SIP failover occurs after 30 seconds.  Unless, did you 
lower the retry count under sip-ua?  Or did you enable SIP options?  If you 
enabled SIP options, have your confirmed that it's turned on correctly?

Can you share the output of the following commands:

show run | section sip-ua|sip.options-keepalive

show dial-peer voice summary

Feel free to redact what you need to, in terms of IPs or usernames/passwords.  
I am only looking for the features and settings for retries and keepalives.


On Mon, Nov 18, 2019 at 9:26 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
Using session server groups on outbound dial-peers and it does not appear to be 
failing over:


voice class server-group 1
 no ipv4 172.31.125.43  preference 2
 no ipv4 172.31.120.43
 description Verizon SIP
!

We had the 172.31.20.43 go down (no response to invites) and we did NOT 
failover to the second (.125.43)...

What is needed to force a failover to the next configured SBC?


Jonathan
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Re: [cisco-voip] UCCX Callback Script

[Johnson, Tim]

Thanks for all of the feedback here. Maybe I’ll try using the same CCG & DG to 
see how it goes. Here’s also where it was mentioned about using separate CCG: 
https://community.cisco.com/t5/contact-center/contact-inactive-when-getting-channel-call-back-from-q/m-p/2091873/highlight/true#M63875

As for this new secret, I’m not quite sure I understand how that would be done, 
but I am intrigued.

From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Tuesday, November 5, 2019 10:24 PM
To: Tanner Ezell 
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: Re: [cisco-voip] UCCX Callback Script


Your abbreviation of easy peasy has won me over.
-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 | 
le...@uoguelph.ca<mailto:le...@uoguelph.ca>

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

[University of Guelph Cornerstone with Improve Life tagline]

On Nov 5, 2019, at 5:31 PM, Tanner Ezell 
mailto:tanner.ez...@gmail.com>> wrote:
PssshhhtI'll share a "secret" for playing the agent menu only when the 
agent answers..

Pass the contact to the agent script, then play your agent menu after they 
connect.

Ez pz.

Regards,
Tanner Ezell



On Tue, Nov 5, 2019 at 2:54 PM Brian Meade 
mailto:bmead...@vt.edu>> wrote:
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 
mailto:avholloway%2bcisco-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 
mailto:johns...@cmich.edu>> 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<mailto:+19897744...@cmich.edu>
Fax: +19897795900
[webexemailsig]<https://cmich.webex.com/meet/johns10t>

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[cisco-voip] UCCX Callback Script

[Johnson, Tim]

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
[webexemailsig]

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Re: [cisco-voip] UCCx and Jabber - compatible or not?

[Johnson, Tim]

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 
; Johnson, Tim ; voyp 
list, cisco-voip (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 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Friday, October 18, 2019 3:36 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>; 
Johnson, Tim mailto:johns...@cmich.edu>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto: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<mailto:le...@uoguelph.ca>

www.uoguelph.ca/ccs<https://nam05.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7Cjohns10t%40cmich.edu%7C0167e165f99b401edd6008d754035322%7Cc871bc6e7cc64a57a4eb22309fc34963%7C1%7C0%7C637070245575507428=Cyo0An0ExLGeAXrnx1pXGkdupHYrRYnrCki8Iscfz%2Bg%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

From: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Friday, October 18, 2019 3:32 PM
To: Johnson, Tim mailto:johns...@cmich.edu>>; Lelio 
Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, cisco-voip 
(cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) 
mailto: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
w: 
www.heliontechnologies.com<https://nam05.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.heliontechnologies.com%2F=02%7C01%7Cjohns10t%40cmich.edu%7C0167e165f99b401edd6008d754035322%7Cc871bc6e7cc64a57a4eb22309fc34963%7C1%7C0%7C637070245575517423=GKUkcVRWPdajscHjw02JeaQgOjDvvqruFX7r%2FF1gKIQ%3D=0>
 |
e: mloradi...@heliontechnologies.com<mailto:mloradi...@heliontechnologies.com>
[Helion 
Technologies]<https://nam05.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.heliontechnologies.com%2F=02%7C01%7Cjohns10t%40cmich.edu%7C0167e165f99b401edd6008d754035322%7Cc871bc6e7cc64a57a4eb22309fc34963%7C1%7C0%7C637070245575517423=GKUkcVRWPdajscHjw02JeaQgOjDvvqruFX7r%2FF1gKIQ%3D=0>
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Re: [cisco-voip] UCCx and Jabber - compatible or not?

[Johnson, Tim]

Yeah, just don’t have a CSF device assigned to that user. Or, if you do have a 
CSF device for that user, make sure their IPCC extension is not assigned to it.


From: cisco-voip  On Behalf Of Lelio 
Fulgenzi
Sent: Friday, October 18, 2019 3:30 PM
To: Matthew Loraditch ; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
Subject: Re: [cisco-voip] UCCx and Jabber - compatible or not?

Interesting. Devices without DNs just freak me out!

---
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]

From: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Friday, October 18, 2019 3:28 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

Correct. If you have agents who want to  use Jabber and have a hardphone DO NOT 
put their agent DN on any other device. If you search the DN in CUCM you need 
to see that DN associated to only a single device




Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: 
www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion 
Technologies]
[Facebook]
[Twitter]
[LinkedIn]
From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Friday, October 18, 2019 3:26 PM
To: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>; 
voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

Yes jabber in deskphone control mode is ok. Jabber is not operating as a 
softphone in that mode and breaks no UCCX “rules”

In this case, do I have to make sure not to program a DN on 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

[University of Guelph Cornerstone with Improve Life tagline]

From: Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Friday, October 18, 2019 3:24 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; voyp list, 
cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: UCCx and Jabber - compatible or not?

LOL

Yes Historically you were good with the hardphone and CIPC b/c you can use ext. 
mobility on both to get you extension and from a UCCX perspective said 
extension was only on one device in CUCM.

Yes jabber in deskphone 

Re: [cisco-voip] UCCx and Jabber - compatible or not?

[Johnson, Tim]

Right, so you can have Jabber configured for an agent, but that’s the only 
thing they’ll be able to use otherwise you’re in an unsupported configuration.

[cid:image002.png@01D585C6.596E5740]

From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Friday, October 18, 2019 3:07 PM
To: Lelio Fulgenzi ; voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
Subject: Re: [cisco-voip] UCCx and Jabber - compatible or not?

The problem is workers who have a deskphone and also need to use a softphone 
occasionally or vice-versa.

An agent extension can only be programmed on one device in all of CUCM.

Previously the above situation could be handled by CIPC as extension mobility 
is supported.

Jabber does not and will not support extension mobility as a softphone.

UCCX says that supporting an agent extension on multiple devices is extremely 
difficult to fix

I can’t remember if you were in those sessions at Cisco Live with me or not but 
it was a fun little it’s not my problem scenario from the two BUs in some of 
the sessions asking how they were going to fix this.

Jabber CTI control is kosher and has no bearing on this issue.


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: 
www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion 
Technologies]
[Facebook]
[Twitter]
[LinkedIn]
From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Friday, October 18, 2019 2:55 PM
To: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] UCCx and Jabber - compatible or not?


I’ve heard all the rumblings with UCCx and Jabber. But I’m hoping someone can 
explain to me what exactly the problem is. From what I understand, there is not 
an actual incompatibility with UCCx and Jabber, just with some deployment 
methods and some restrictions UCCx places on things.

So, in simplest terms, if you have a desktop agent, with Jabber (4Win) running 
and only one (agent) extension and no other device has that extension, things 
work.

I believe it’s when you try to do anything else other than this simple 
deployment that things go awry. Say, have both Jabber and a hard phone and log 
into your queue using a phone agent on the hard phone but want to use Jabber to 
answer. Or you want Jabber to control your hard phone. Etc.

Can I get some ideas as to whether this is true and where people have seen 
trouble? And what is actually supported?

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] CCX Editor Step Properties

[Johnson, Tim]

Yeah, it’s still an issue in 12.0. Right now it’s opening properties for me in 
1-2 seconds, other times it’s 10.

Would be nice if they did a full makeover of the tool, but I don’t expect it.

From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Tuesday, September 24, 2019 11:11 AM
To: Matthew Loraditch 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CCX Editor Step Properties

I would like to know this too!  Seems to be different lengths of delay 
depending on a few factors, however, I have not nailed down what those factors 
are.  If it happened more often, I'd put some time into it, but since it's 
generally quick-ish, I ignore it.

I have one system I access exclusively via AnyConnect, and it's running 
11.6(2), and it's god awful slow.  Just clicking Add on the Set Enterprise Call 
Info step takes like 10-20 seconds for the dialog box to pop open.

On Tue, Sep 24, 2019 at 9:51 AM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
Is there anyway on this earth to make this load faster???  Currently painfully 
going through a script and setting up new parameters and slowly dying of 
impatience as I click properties and wait a seeming eternity for the window to 
open.
This has been a pet peeve forever.

Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: 
www.heliontechnologies.com
 |
e: mloradi...@heliontechnologies.com
[Helion 
Technologies]
[Facebook]
[Twitter]
[LinkedIn]
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Re: [cisco-voip] UCCX Call Escalation/Tiered CSQs

[Johnson, Tim]

You’re right. It would still be considered a supervised transfer, having them 
wait for the additional digits to be sent before completing the transfer. Yes, 
using commas for the delay in the string.

From: Anthony Holloway 
Sent: Monday, August 5, 2019 12:05 PM
To: Johnson, Tim 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX Call Escalation/Tiered CSQs

Oh I see, a secret door.

A speed dial on the phone would not produce a blind transfer though, right?  
Unless I'm missing something.  The Agent would press transfer, then the caller 
goes on network hold, then the Agent presses the speed dial and waits for all 
of the digits to be sent (assuming you're using commas as delay?) and then the 
Agent must press transfer again to complete the transfer.

On Mon, Aug 5, 2019 at 10:35 AM Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:
Thanks for the response.

Yeah, you’re right. I hadn’t thought too deep into how it would affect 
reporting, but splitting into a second application does provide better options 
there. This is the way we’ve done it with another call center that we provide 
service for, but I just wasn’t sure if there was a “better” way.

With the Get Digit String, I was thinking we could use their “welcome” prompt 
on that step. There would be no indication during the prompt that it was 
available so it would kind of be somewhat of a “hidden” menu. The 1st tier 
agents would be able to call their main line, and press an expected digit 
string that I could apply IF logic to and process the call differently. The 
blind transfer could be handled just by adding a speed dial on their phones to 
include a wait and then the DTMF digits. That being said though, the call 
center is thinking that a phase two of this change will be to add a menu for 
the 1st tier agents to select from to route to agents that specialize in 
different areas.

From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Monday, August 5, 2019 10:55 AM
To: Johnson, Tim mailto:johns...@cmich.edu>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] UCCX Call Escalation/Tiered CSQs

I wouldn't worry too much about cluttering up your scripts (or apps, or 
triggers).

First and foremost, I would worry about reporting.  Does the solution allow the 
stock reports to adequately meet the needs of the reporting personnel?

Second, I would worry about ease of execution for the Agent.  Are the Agents 
able to easily execute this escalation?

If I had to pick one of your three options, Option 1 would be my choice.  This 
would allow Application level reporting (real-time and historical), not just 
called number and/or CSQ level.

I didn't quite understand why your Option 3 uses a Get Digit String.  Can you 
explain that?  Also, this option would not allow Agents to take advantage of a 
blind transfer (a speedier option for escalating calls).

On Mon, Aug 5, 2019 at 9:25 AM Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:
Hello,

I’m looking for ideas on how people setup their applications/scripts when 
handling call centers with multiple tiers of support. More specifically, how do 
you handle 1st tier agents queueing calls for 2nd tier agents? Below, I’ve 
provided three ways I can think of to achieve it, but I’m curious if someone 
has a better idea.


  *   An application/trigger for each tier. The 1st tier agent transfers the 
call to the trigger for the next tier. Benefit of this being that the scripts 
are broken out and there’s not as much clutter in each.
  *   A second trigger on the application, and a Get Call Contact Info step to 
grab the called number and queue the call for 2nd tier CSQ based on that. The 
1st tier agent would transfer the call two the secondary trigger. This makes 
for a more cluttered script, but you don’t have to cross reference anything.
  *   A Get Digit String that is used at the “welcome” prompt, which can be 
used by the 1st tier agent when they do a supervised transfer to the trigger on 
the application. This again makes for a more cluttered script than doing two 
applications/triggers, but maybe makes it easier to manage and do reporting on?

Thanks!

Tim Johnson

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Re: [cisco-voip] UCCX Call Escalation/Tiered CSQs

[Johnson, Tim]

Thanks for the response.

Yeah, you’re right. I hadn’t thought too deep into how it would affect 
reporting, but splitting into a second application does provide better options 
there. This is the way we’ve done it with another call center that we provide 
service for, but I just wasn’t sure if there was a “better” way.

With the Get Digit String, I was thinking we could use their “welcome” prompt 
on that step. There would be no indication during the prompt that it was 
available so it would kind of be somewhat of a “hidden” menu. The 1st tier 
agents would be able to call their main line, and press an expected digit 
string that I could apply IF logic to and process the call differently. The 
blind transfer could be handled just by adding a speed dial on their phones to 
include a wait and then the DTMF digits. That being said though, the call 
center is thinking that a phase two of this change will be to add a menu for 
the 1st tier agents to select from to route to agents that specialize in 
different areas.

From: Anthony Holloway 
Sent: Monday, August 5, 2019 10:55 AM
To: Johnson, Tim 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] UCCX Call Escalation/Tiered CSQs

I wouldn't worry too much about cluttering up your scripts (or apps, or 
triggers).

First and foremost, I would worry about reporting.  Does the solution allow the 
stock reports to adequately meet the needs of the reporting personnel?

Second, I would worry about ease of execution for the Agent.  Are the Agents 
able to easily execute this escalation?

If I had to pick one of your three options, Option 1 would be my choice.  This 
would allow Application level reporting (real-time and historical), not just 
called number and/or CSQ level.

I didn't quite understand why your Option 3 uses a Get Digit String.  Can you 
explain that?  Also, this option would not allow Agents to take advantage of a 
blind transfer (a speedier option for escalating calls).

On Mon, Aug 5, 2019 at 9:25 AM Johnson, Tim 
mailto:johns...@cmich.edu>> wrote:
Hello,

I’m looking for ideas on how people setup their applications/scripts when 
handling call centers with multiple tiers of support. More specifically, how do 
you handle 1st tier agents queueing calls for 2nd tier agents? Below, I’ve 
provided three ways I can think of to achieve it, but I’m curious if someone 
has a better idea.


  *   An application/trigger for each tier. The 1st tier agent transfers the 
call to the trigger for the next tier. Benefit of this being that the scripts 
are broken out and there’s not as much clutter in each.
  *   A second trigger on the application, and a Get Call Contact Info step to 
grab the called number and queue the call for 2nd tier CSQ based on that. The 
1st tier agent would transfer the call two the secondary trigger. This makes 
for a more cluttered script, but you don’t have to cross reference anything.
  *   A Get Digit String that is used at the “welcome” prompt, which can be 
used by the 1st tier agent when they do a supervised transfer to the trigger on 
the application. This again makes for a more cluttered script than doing two 
applications/triggers, but maybe makes it easier to manage and do reporting on?

Thanks!

Tim Johnson

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[cisco-voip] UCCX Call Escalation/Tiered CSQs

[Johnson, Tim]

Hello,

I'm looking for ideas on how people setup their applications/scripts when 
handling call centers with multiple tiers of support. More specifically, how do 
you handle 1st tier agents queueing calls for 2nd tier agents? Below, I've 
provided three ways I can think of to achieve it, but I'm curious if someone 
has a better idea.


  *   An application/trigger for each tier. The 1st tier agent transfers the 
call to the trigger for the next tier. Benefit of this being that the scripts 
are broken out and there's not as much clutter in each.
  *   A second trigger on the application, and a Get Call Contact Info step to 
grab the called number and queue the call for 2nd tier CSQ based on that. The 
1st tier agent would transfer the call two the secondary trigger. This makes 
for a more cluttered script, but you don't have to cross reference anything.
  *   A Get Digit String that is used at the "welcome" prompt, which can be 
used by the 1st tier agent when they do a supervised transfer to the trigger on 
the application. This again makes for a more cluttered script than doing two 
applications/triggers, but maybe makes it easier to manage and do reporting on?

Thanks!

Tim Johnson

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