Re: [cisco-voip] SMS message

[Ryan Huff]

Unity doesn't have a natively dedicated way to use the SMS protocol, however, 
most tier 1 cell phone carriers, transport their SMS messaging through or 
within the SMTP protocol.

Verizon for example, uses xxx...@vtext.com, so in theory, you could use 
Unity to send to a standard SMTP address, which, in theory, would be delivered 
as an SMS text to the recipient.

Thanks,

-R

Thanks,

Ryan Huff

From: cisco-voip  on behalf of harbor235 

Sent: Friday, July 21, 2023 9:19:58 AM
To: Cisco VOIP 
Subject: [cisco-voip] SMS message

Hi everyone,

Is it possible to send an SMS message after receipt of a voicemail. I can 
scrape the mail relay to send a SMS message but was wondering how to do this on 
Unity?


Mike
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Re: [cisco-voip] [External] Re: Certificate issue and I am rubbish at certificates. (full disclosure)

[Ryan Huff]

Sovereign Citizen. That’s just funny.

Thanks,

Ryan Huff

From: cisco-voip  on behalf of Hunter 
Fuller 
Sent: Wednesday, May 24, 2023 12:14:27 PM
To: Matthew Loraditch 
Cc: Terry Oakley ; voip puck 

Subject: Re: [cisco-voip] [External] Re: Certificate issue and I am rubbish at 
certificates. (full disclosure)

2028 is WAY too far in the future. No modern browser trusts a
publicly-issued certificate that is valid that far in the future. How
did you even get that certificate.

If you did a self signed, then that would explain why no browser
trusts it. Self signed is the "sovereign citizen" of certificates. You
need to get a certificate authority to sign your CSR.

https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fknowledge.digicert.com%2Fgeneralinformation%2F2-year_Certificate_Availability.html=05%7C01%7C%7C33aae16f4f824da959ec08db5c72202d%7C84df9e7fe9f640afb435%7C1%7C0%7C638205417463181216%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C=F3nhWssXTK3oZj0mDi%2BySMTvinQ2iJcDRiQvQIMOVto%3D=0<https://knowledge.digicert.com/generalinformation/2-year_Certificate_Availability.html>

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

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

On Wed, May 24, 2023 at 11:01 AM Matthew Loraditch
 wrote:
>
> It sounds like something is different between the old and new certs (besides 
> the dates). As far as clients accessing Unity via a browser, the 
> callmanager-trust certs are not involved. I’m not even sure they are used at 
> all on a Unity server. I’ve never touched them.
>
>
>
> I would take a look at the old and new certs and make sure the subject and 
> SAN fields are all the same. There can be a lot of reasons for cert errors 
> and the errors are all similar and hard to diagnose without access to the 
> browser throwing the error, but that’s the first thing I would check.
>
>
>
>
>
>
> Matthew Loraditch
> Sr. Network Engineer
> direct: 443.541.1518
> e: mloradi...@heliontechnologies.com
> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.heliontechnologies.com%2F=05%7C01%7C%7C33aae16f4f824da959ec08db5c72202d%7C84df9e7fe9f640afb435%7C1%7C0%7C638205417463181216%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C=9WGDmNKbNXHrjDes9vllJS%2FN9u4u5uEOOHMOeF4e5xk%3D=0<http://www.heliontechnologies.com/>

>
> From: cisco-voip  On Behalf Of Terry 
> Oakley
> Sent: Wednesday, May 24, 2023 11:35 AM
> To: 'voip puck' 
> Subject: [cisco-voip] Certificate issue and I am rubbish at certificates. 
> (full disclosure)
>
>
>
> [EXTERNAL]
>
>
>
> On our Unity Connection server the certificates for Tomcat and Tomcat trust 
> expired over the weekend, my oversight.   I regenerated the certificates and 
> both are now year 2028 expiry date.   But we still get the same error if 
> someone is trying to access their inbox  (https://server/inbox/)  (error is 
> You cannot visit server right now because the website uses HSTS)
>
>
>
> I noticed that there is a CallManager-Trust certificate that expired on the 
> same day as the Tomcat certs.   The CallManager-Trust certificate is issued 
> by the CA (CA signed) but when I go to Generate a CSR I don’t have the option 
> to choose CallManager-Trust or Trust .  I have Tomcat, Tomcat ecdsa or ipsec. 
>   The common name for the expired CallManager-Trust certificate is the 
> UnityConnection server that users cannot get too.   Little confused as to 
> where this CallManager Trust certificate can be generated from.
>
>
>
>
>
> Thank you
>
>
>
> Terry
>
>
>
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[cisco-voip] FXS ordering question

[Ryan Huff]

Does any know if the base NIM-4FXS SKU has the PVDM4 already on the NIM, or do 
I need to buy the PVDM4 and add it in to the NIM? I assume it does come with it 
because unlike the MFT T1 there is no other use for the NIM-4FXS without DSP.

Thanks,

Ryan Huff
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Re: [cisco-voip] FX0 pool exhuastion

[Ryan Huff]

There isn’t any great “baked in” way to monitor aggregate usage, however, some 
things that might help:


  *   Log the output of “debug vpm signal” to the syslog. If you’re time 
stamping the syslog entries then you could reasonably piece together the times 
the FXO port’s experience a fxols_onhook_ringing or fxols_proc_voice event.

  *   Use some sort of a traffic grapher like Paessier (PRTG). These platforms 
tend to have a knack for being able to login to CLIs and execute command 
structures. If you can do that (one way or the other), frequently capture the 
output of "show voice port summary" which gives you a real-time / at-the-moment 
state of each FXO/S voice port

Absent that, you'd likely be relegated to the above, in an "on demand / manual" 
way.

Thanks,

Ryan

Thanks,

Ryan Huff

From: cisco-voip  on behalf of harbor235 

Sent: Monday, April 17, 2023 11:15:06 AM
To: Cisco VOIP 
Subject: [cisco-voip] FX0 pool exhuastion

Hi,

I have a CISCO ISR4331 voice bundle (CME) setup with a 4FX0 using 4 analog 
lines. The number of phones and users have increased and would like to verify 
justification to add SIP trunks. If we are experiencing no free FX0 lines for 
inbound or outbound calls does a log message get generated to observe the need 
to augment the analog line pool or add a sip trunk? I would like an observable 
data point to justify adding more capacity?


thanks in advance,

Mike


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Re: [cisco-voip] Voicemail alternative with cisco CME

[Ryan Huff]

In theory, you could make a SIP trunk connection to “whatever” and make 
blind/supervised transfer to “whatever”. As far as a tightly integrated or 
native solution however, an old CUE or CUC is your best bet.

Thanks,

Ryan Huff

From: cisco-voip  on behalf of harbor235 

Sent: Friday, April 7, 2023 9:28:44 AM
To: Cisco VOIP 
Subject: [cisco-voip] Voicemail alternative with cisco CME

Hi all,

Is there a way to avoid using Cisco voicemail "Unity Connection" with Cisco CME?
Just curious if a non-cisco product can provide that functionality. I am trying 
to avoid the monthly charge for the current unity connection product.

thanks in advance,


Mike
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Re: [cisco-voip] MRA voice quality issues with Optimum/Altice

[Ryan Huff]

I know you didn’t ask for solutions… however, you might consider throwing an 
SDWAN appliance in front of the Expressway. The forward error correction 
capabilities might help with the jitter / lack of end to end EF queue.

Get Outlook for iOS

From: cisco-voip  on behalf of Matthew Huff 

Sent: Monday, October 11, 2021 12:38:42 PM
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] MRA voice quality issues with Optimum/Altice


A continuing problem we have is voice quality issues with our users  on 
Optimum/Altice. Garbled and dropped speech are common. I’ve played around with 
different codes without much success. Opus appears to work as well as any 
others. We are directly peered with Lightpath, and the traceroutes show a 
direct path, so it’s not an issue with congestion at a peering point. We don’t 
have any issue with Verizon or Spectrum users.



Not really looking for any solutions, as it’s likely a issue with Optimum, but 
I was curious if this is a common issue.



BTW, don’t get me started with the Altice One service. The issues with their 
router alone is horrid.



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: Issue with Legit DIDs Flagged as "Potential SPAM"

[Ryan Huff]

CNAME dips are done by the called party carrier. Nothing you can do to change 
that other than change the calling party ANI or work with the calling/called 
party carrier to correct the database entry.

Get Outlook for iOS

From: cisco-voip  on behalf of JASON 
BURWELL via cisco-voip 
Sent: Saturday, March 27, 2021 9:53:09 PM
To: Kent Roberts 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] [EXTERNAL] Re: Issue with Legit DIDs Flagged as 
"Potential SPAM"

I’m having the same issue with a couple of the TFNs as well.

Jason

On Mar 27, 2021, at 9:44 PM, Kent Roberts  wrote:


CAUTION: This email originated outside of Founders Federal Credit Union. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

What if you send TF number, instead of a DID?


On Mar 27, 2021, at 6:49 PM, JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

Hello All,

An issue has come up again and I was hoping maybe someone here had delt with it 
and possibly had some advice. More and more, the DID numbers we send out as 
calling party number are being flagged as “Possible SPAM” or other types of 
“Probable SPAM” messages with the mobile phone carriers. As a result customers 
wont answer the call. These are legitimate customer calls and not any kind of 
cold calling or telemarketing. I took this up with Verizon and AT reps last 
year and got a lot of excuses and basically they told me there is no way to 
“whitelist” list numbers, the systems use some kind of AI processing to decide 
what calls to flag. So the issue has come up again because customers are 
requesting callback and when we call them back sometimes its flagged as 
possible SPAM so they don’t answer the call. Has anyone else found a way to 
deal with this? Hopefully once STIR/SHAKEN is fully implemented, they will stop 
doing this but until then its causing us quite a bit of headache.

Thanks
Jason
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Re: [cisco-voip] CUCM call set up issue after migration

[Ryan Huff]

This would be discovered in the “utils diagnose test” however, I have seen on 
more than one occasion, in a “node-move-IP-change” situation, where either the 
reverse DNS entry didn’t get changed, or more scrupulously, the incorrect one 
exists along with the correct one.

-Ryan

On Jan 12, 2021, at 16:55, Wes Sisk (wsisk) via cisco-voip 
 wrote:

 On UCM cli ‘utils diagnose test’ to ensure things look mostly okay from OS 
and low level perspective.

After that in UCMadmin ensure system->server shows the correct IP/name for each 
server.

After that ensure ccm process restarted (or whole server rebooted). Based on 
your description SDL links on tcp:8002 are not coming up because remote name/ip 
is not recognized against what is the the database and hosts file for name 
resolution.

-w

On Jan 12, 2021, at 4:05 PM, Riley, Sean 
mailto:sri...@robinsonbradshaw.com>> wrote:

This past weekend we migrated 2 CUCM servers to a new datacenter.  This 
involved changing the IP address on these 2 CUCM nodes.  These 2 nodes consist 
of the Publisher and 1 Subscriber.  We have another Sub at a remote datacenter 
that was not touched this past weekend.

Node configuration:

DC A
CM1: Pub which was re-ip’d
CM2: Sub which was re-ip’d

DC B:
CM3: Sub at remote site that was not changed

Phones are at many sites, but issue is independent of the phone type, phone 
location or subnet.  Also, Expressway phones have the same issue.

The issue is any phone that is registered to CM3 cannot call phones registered 
to CM1 or CM2 and vice versa.  The phones do not see the call coming in.  If 
SNR is configured, the call will ring to the remote destination. Phones 
registered to CM3 can make outbound PSTN calls without issue, but not receive 
inbound from PSTN (probably because the gateway is handing off to CM1 or CM2).  
While the gateways are not unique to the issue, they are running H323.

If the phones are both registered to CM3, they can call each other, but not 
phones registered to CM1 or CM2.

I have had my network team verify there is not anything they can see in the 
network causing this behavior. Database replication checks out OK and I can 
ping from/to each node.

Anyone able to point me in the right direction to figure this out?

Thanks.
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Re: [cisco-voip] Different MOH for multi site

[Ryan Huff]

A little more cumbersome, but common device configuration (CDC) should offer 
you this flexibility.

Sent from my iPhone

> On Jan 5, 2021, at 15:18, f...@browardcommunications.com wrote:
> 
> That’s where I’m confused, I don’t see where you can set a specific audio 
> source per site, or device pool, in the device pool settings, you can just 
> set a mrgl.
> 
> There will be many sites, more than 2, and the idea is if I call a phone at 
> site A, I would get the Site A MOH audio source file, if I call site B I 
> would get the site B audio source file, and so on.
> 
> 
> 
> Sent from my iPhone
> 
>> On Jan 5, 2021, at 3:05 PM, Kent Roberts  wrote:
>> 
>> Multicast is easy since it’s 2 different sites
>> 
>> Otherwise use unicast. Setup your music in cucm, one fir each site and 
>> assign it-in the device pool or on each device 
>> 
>> 
>> Kent
>> 
 On Jan 5, 2021, at 12:52, f...@browardcommunications.com wrote:
>>> 
>>> Does anyone have a recommendation on how to have different MOH depending 
>>> on what number / site is called?
>>> Cucm 12.5
>>> 2 node cluster
>>> Multi site
>>> 
>>> Thank you
>>> 
>>> Sent from my iPhone
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Re: [cisco-voip] List still active?

[Ryan Huff]

Yes, it has declined in volume.

Sent from my iPhone

> On Dec 25, 2020, at 14:30, Bill Talley  wrote:
> 
> Thanks for the confirmation Ryan.  Are you also seeing a significant decline 
> in volume from the group?   
> 
> Hope all the usual (and even casual) participants are staying healthy and 
> employed.  Hope those aren’t reasons for the decline in forum usage. 
> 
> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
> Please excude my typtos.
> 
>> On Dec 25, 2020, at 1:28 PM, Ryan Huff  wrote:
>> 
>> I still see you.
>> 
>> Sent from my iPhone
>> 
>>>> On Dec 25, 2020, at 14:28, Bill Talley  wrote:
>>> 
>>> I stopped receive list emails.   Is the list dead or was I banned? 
>>> 
>>> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
>>> Please excude my typtos.
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Re: [cisco-voip] List still active?

[Ryan Huff]

I still see you.

Sent from my iPhone

> On Dec 25, 2020, at 14:28, Bill Talley  wrote:
> 
> I stopped receive list emails.   Is the list dead or was I banned? 
> 
> Sent from an iPhone mobile device with very tiny touchscreen input keys.  
> Please excude my typtos.
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Re: [cisco-voip] Unity xfer Getting 603 from carrier

[Ryan Huff]

Codec preservation tells CUBE to try not to renegotiate the codec in the middle 
of a call. Not sure of your setup, but most major carriers won’t typically do 
this anyway. I can’t really tell you what would happen in your environment by 
changing that, other than it would allow CUBE to participate in codec 
renegotiations mid call.

“media-change” helps reduce reINVITES from reaching the carrier unless there is 
a reason for it to (like an actual media change). Sometimes call forwards that 
involve Unity Connection will generate reINVITES with a null (0.0.0.0) media 
address (no media change). If that reINVITE reaches the carrier, the carrier 
may “pay attention” to it and try to renegotiate the media to 0.0.0.0 which it 
can’t... so the call drops.

Some carriers will ignore these reINVITES (they are supposed to) and there is 
no issue, other carriers (AT is particularly offensive with this in my 
experience) do not, and quirks like this creep up.

You can read up on the command here: 
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_proto/configuration/15-mt/cube-proto-15-mt-book/voi-cube-midcall-reinvite.html#GUID-F5641A05-14F7-4396-B04B-5F1D4F527457

Thanks,

Ryan

On Jul 29, 2020, at 08:25, "f...@browardcommunications.com" 
 wrote:

 Thank you.
Currently that setting under sip is set to:
midcall-signaling preserve-codec

If that gets changed what do you think the impact would be?

This customer has had many hands on their environment and is a bit of a mess, 
so I feel that if I change one thing, 10 other things will break.


Sent from my iPhone

On Jul 29, 2020, at 8:18 AM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Try adding “midcall-signaling passthru media-change” under voice service voip 
-> sip.

Thanks,

Ryan

Sent from my iPhone

On Jul 29, 2020, at 07:45, 
"f...@browardcommunications.com<mailto:f...@browardcommunications.com>" 
mailto:f...@browardcommunications.com>> wrote:


Greetings all, this might be simple fix, I just haven’t dealt with this in a 
while.

We have a unity AA that, when external callers call, select menu options, etc. 
Unity will send the call back to. CtiRp, which then CFA to an external number. 
You hear the Unity transfer message, 1 second of MoH, then 10 seconds of 
nothing, then the call drops.

Internal calls to the CFA ctirp works

Internal calls to the unity ctirp then back to the CFA ctirp works.

I tried having unity call the pstn number directly with same results

I think the issue is with the calling number is why the carrier is sending the 
603, but it is next to impossible to get them to tell us that.

Call flow:
Pstn>sipt>cucm >unity>cucm>ctirp-CFA-pstn

Where would I change the calling number being CFA’ed from Unity to the PSTN?

Any ideas?

Thank you.

/FW

Sent from my iPhone
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Re: [cisco-voip] Unity xfer Getting 603 from carrier

[Ryan Huff]

Try adding “midcall-signaling passthru media-change” under voice service voip 
-> sip.

Thanks,

Ryan

Sent from my iPhone

On Jul 29, 2020, at 07:45, "f...@browardcommunications.com" 
 wrote:


Greetings all, this might be simple fix, I just haven’t dealt with this in a 
while.

We have a unity AA that, when external callers call, select menu options, etc. 
Unity will send the call back to. CtiRp, which then CFA to an external number. 
You hear the Unity transfer message, 1 second of MoH, then 10 seconds of 
nothing, then the call drops.

Internal calls to the CFA ctirp works

Internal calls to the unity ctirp then back to the CFA ctirp works.

I tried having unity call the pstn number directly with same results

I think the issue is with the calling number is why the carrier is sending the 
603, but it is next to impossible to get them to tell us that.

Call flow:
Pstn>sipt>cucm >unity>cucm>ctirp-CFA-pstn

Where would I change the calling number being CFA’ed from Unity to the PSTN?

Any ideas?

Thank you.

/FW

Sent from my iPhone
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Re: [cisco-voip] how to disable backup warning message on cucm

[Ryan Huff]

No way to disable the alert message that I’m aware of. I think DRS is an 
unavoidable assumption (and by extension m, the alert) in the modern versions.

As you know, this isn’t a great strategy. It’s a little more than, “not 
recommended”, it’s actually not supported by Cisco to backup this way. Veeam 
has been known to cause CPU spikes, kernel panics.. etc in CUCM (while powered 
on)... not a good strategy at all. DRS is the path to reinforce ;)

From my understanding, it’s a pretty simplistic check... just looking for a 
backup device, and the the XML file for the backup set in the backup device’s 
location.

They might be able to run one manual DRS, and then just keep modifying the 
dates in the XML for the backup set. Seems like something that could be 
scripted fairly easily too.

To me though, that’s a lot of work to intentionally do it the wrong way. It’s 
been my experience that when customers invite the Devil to dinner, he usually 
shows up.

- Ryan

On Jul 29, 2020, at 03:12, naresh rathore  wrote:


hi,


One of our Customer running version cucm 12.5.1.12900-115 (upgraded from 10). 
they had backup enabled, they decided to do veem backup (even though not 
recommended by Cisco). they deleted backup device and schedule configuration 
and also disabled DRF Master and DRF local and restarted tomcat but still we 
see  message of 32 days without backup. is there a way to disable this warning?




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Re: [cisco-voip] DRS Message Restore & DRS File Decrypter Updated

[Ryan Huff]

Cool man! Take a victory lap cause Cisco is no longer going to play “hide the 
cheese” with you each time they tweak the salt functions in new versions. Think 
of all the free time!

Lol... cheers Pete.

Sent from my iPhone

On May 11, 2020, at 16:23, Pete Brown  wrote:


Yes, all that’s been ironed out.  Jeff had to deal with a ridiculous amount of 
red tape to make it happen.  Plus he did a great job of cleaning up the code 
(packaging, logging, error handing, etc).

I’m just happy that it’s now a “blessed” utility instead of something that’s 
frowned on in the Community forums.

Sent from 
Mail
 for Windows 10

From: Anthony Holloway
Sent: Monday, May 11, 2020 12:20 PM
To: Pete Brown
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DRS Message Restore & DRS File Decrypter Updated

Are these based off of your tools?  If so, are you credited in any way?

On Sat, May 9, 2020 at 12:36 PM Pete Brown 
mailto:j...@chykn.com>> wrote:
Just a heads up, Cisco has made some updates to the DRS file decryption tool 
and released it for customer use.  Jeff Lindborg has also updated the DRS 
Message Restore (fka DRS Message Fisher) tool .  Both are available here.

http://www.ciscounitytools.com/Applications/CxN/DRSMessageFisher/DRSMessageFisher.html

I’m leaving my tools page and GitHub repo up, but I’d recommend using Cisco’s 
version going forward.  It’s been fun!  


Sent from 
Mail
 for Windows 10

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Re: [cisco-voip] sip 404 not found for incoming calls

[Ryan Huff]

Thanks.

The 404 is sourced from the gi1/0/1 interface IP address. If CUCM initiated the 
404 I would expect CUCM’s IP address to be the source of the 404.

It seems like this is config as “sip on a stick” (a router with a single 
interface used for inbound and outbound SIP traffic). A little more 
challenging, not impossible, but more challenging.

It would be outstanding if you could make the gi0/0/0 interface an “outside” 
interface in a different network, then bind the dial peers appropriately... 
again, not material to the issue outhand.

Once you have the inout debuts, please send it ove.

Sent from my iPhone

On Apr 30, 2020, at 09:06, naresh rathore  wrote:


hi


Pls find the attached ccsip messages and cube config. i will send voice ccapi 
inout config tomorrow. outgoing call works fine but cucm respond with 404 
message during incoming call attempt


Regards




From: Ryan Huff 
Sent: Thursday, April 30, 2020 10:50 PM
To: naresh rathore 
Cc: Amit Kumar ; cisco-voip@puck.nether.net 

Subject: Re: [cisco-voip] sip 404 not found for incoming calls

Naresh,

Any chance you could send a ccsip messages and a voice ccapi inout debug from a 
failed inbound call?

Sent from my iPhone

On Apr 30, 2020, at 06:28, naresh rathore  wrote:


hi


I tried both, via translation pattern or directly pointing a particular number 
to phone but still the same result.

Regards



From: Amit Kumar 
Sent: Thursday, April 30, 2020 6:41 PM
To: naresh rathore 
Cc: James B ; cisco-voip@puck.nether.net 

Subject: Re: [cisco-voip] sip 404 not found for incoming calls

Are you having a dn, exact to called number, of you are doing some translation, 
then make sure route pattern incoming css shoold have access to xlate pt. Nd 
xlate css shoud have access to phones pt.

On Thu, Apr 30, 2020, 2:02 PM naresh rathore 
mailto:nare...@hotmail.com>> wrote:
hi,


Thanks for the reply.


pls see following snapshot and attached gateway config. outgoing dialpeer 
(200,201) is currently matching correctly to cucm (for incoming call)

[cid:9c8082fb-a47a-4859-8d73-bd22200523c5]
[cid:f721c217-c6cf-4fd5-bbc6-10962f7f7269]
[cid:2300f744-b399-4d90-ab4b-133bb37a3611]
[cid:a2643290-7308-42d9-b6bc-699c2df07a6f]


Regards

Naray

From: James B mailto:james.buchan...@gmail.com>>
Sent: Thursday, April 30, 2020 5:39 PM
To: naresh rathore mailto:nare...@hotmail.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] sip 404 not found for incoming calls


Hello,



Can you send your gateway configuration and a screenshot of your CUCM trunk 
configuration? That’d give us more to go off of.



Thanks,



James







From: naresh rathore<mailto:nare...@hotmail.com>
Sent: 30 April 2020 08:36
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] sip 404 not found for incoming calls



hi,







i have cucm version 12.0. outgoing call is working without any issue. but 
incoming call is failing. the call request is received by cucm but its 
responding with 404 not found. i checked CSS and also pointed call directly to 
ip phone using significant digits and incoming css but still the same issue. 
also sip uri have called number. not sure why 404 not found msg is sent by cucm 
to cube.





Regards



Naray



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Re: [cisco-voip] sip 404 not found for incoming calls

[Ryan Huff]

Naresh,

Any chance you could send a ccsip messages and a voice ccapi inout debug from a 
failed inbound call?

Sent from my iPhone

On Apr 30, 2020, at 06:28, naresh rathore  wrote:


hi


I tried both, via translation pattern or directly pointing a particular number 
to phone but still the same result.

Regards



From: Amit Kumar 
Sent: Thursday, April 30, 2020 6:41 PM
To: naresh rathore 
Cc: James B ; cisco-voip@puck.nether.net 

Subject: Re: [cisco-voip] sip 404 not found for incoming calls

Are you having a dn, exact to called number, of you are doing some translation, 
then make sure route pattern incoming css shoold have access to xlate pt. Nd 
xlate css shoud have access to phones pt.

On Thu, Apr 30, 2020, 2:02 PM naresh rathore 
mailto:nare...@hotmail.com>> wrote:
hi,


Thanks for the reply.


pls see following snapshot and attached gateway config. outgoing dialpeer 
(200,201) is currently matching correctly to cucm (for incoming call)

[cid:9c8082fb-a47a-4859-8d73-bd22200523c5]
[cid:f721c217-c6cf-4fd5-bbc6-10962f7f7269]
[cid:2300f744-b399-4d90-ab4b-133bb37a3611]
[cid:a2643290-7308-42d9-b6bc-699c2df07a6f]


Regards

Naray

From: James B mailto:james.buchan...@gmail.com>>
Sent: Thursday, April 30, 2020 5:39 PM
To: naresh rathore mailto:nare...@hotmail.com>>; 
cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] sip 404 not found for incoming calls


Hello,



Can you send your gateway configuration and a screenshot of your CUCM trunk 
configuration? That’d give us more to go off of.



Thanks,



James







From: naresh rathore
Sent: 30 April 2020 08:36
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] sip 404 not found for incoming calls



hi,







i have cucm version 12.0. outgoing call is working without any issue. but 
incoming call is failing. the call request is received by cucm but its 
responding with 404 not found. i checked CSS and also pointed call directly to 
ip phone using significant digits and incoming css but still the same issue. 
also sip uri have called number. not sure why 404 not found msg is sent by cucm 
to cube.





Regards



Naray



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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Ryan Huff]

Yes, RFC 2833 is the older RFC and 4733 is the newer. I believe CUCM still 
references 2833 because 4733 could potentially result in a non supported DTMF 
scenario that would appear supported if CUCM stated it supported 4733 (Ex. a 
SBC not supporting all the same events that CUCM would require perhaps).

I haven’t dug into the topic in depth however, so I may not be correct.

Here is an excerpt from 4734 that sums it up; “ This document provides a number 
of clarifications to the original document. However, it specifically differs 
from RFC 2833 by removing the requirement that all compliant implementations 
support the DTMF events. Instead, compliant implementations taking part in 
out-of-band negotiations of media stream content indicate what events they 
support. This memo adds three new procedures to the RFC 2833 framework: 
subdivision of long events into segments, reporting of multiple events in a 
single packet, and the concept and reporting of state events.“


Sent from my iPhone

On Apr 24, 2020, at 12:01, Anthony Holloway  
wrote:


Ding ding ding!  Winner!  I wonder why Cisco doesn't update the CUCM UI.  I was 
looking for DTMF support in a Telepresence Admin Guide for like an SX20 or 
something, and I couldn't find RFC2833 mentioned anywhere, but it did mention 
RFC4733.  Anyway, that's all the trivia I have for now.

On Fri, Apr 24, 2020 at 10:17 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
RCF 4733, I believe.

Sent from my iPhone

On Apr 24, 2020, at 10:58, Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:


Actually you don't want to set rfc2833 (pop quiz: rfc2833 is not the real RFC 
number.  What's the real RFC number?  Don't google it, but reply if you know!) 
on your CUCM SIP Trunk to CUBE.  You want No Preference.  It's a setting right 
on the SIP Trunk, just scroll to the bottom of the settings page.  Also, on 
your CUBE dial-peers you don't want solely rtp-nte either, you want both 
rtp-nte and at least one Out of Band (OOB) option, like sip-kpml or sip-notify 
(thought the latter requires a SIP Trunk Security Profile change from default 
to allow unsolicited NOTIFY).


On Fri, Apr 24, 2020 at 9:35 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks


From: Jason Aarons mailto:scubajas...@gmail.com>>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

I am doubtful porting had anything to do with it. Was it tested fully before 
the port?

Under dial peers is dtmf-relay rtp-nte set? In CUCM trunks is rfc2833 set? How 
is Unity integrated with CUCM ? SIP? CXN Version?

Without some debugs /traces I suspect you won't find much.

On Thu, Apr 23, 2020, 3:52 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Ryan Huff]

RCF 4733, I believe.

Sent from my iPhone

On Apr 24, 2020, at 10:58, Anthony Holloway  
wrote:


Actually you don't want to set rfc2833 (pop quiz: rfc2833 is not the real RFC 
number.  What's the real RFC number?  Don't google it, but reply if you know!) 
on your CUCM SIP Trunk to CUBE.  You want No Preference.  It's a setting right 
on the SIP Trunk, just scroll to the bottom of the settings page.  Also, on 
your CUBE dial-peers you don't want solely rtp-nte either, you want both 
rtp-nte and at least one Out of Band (OOB) option, like sip-kpml or sip-notify 
(thought the latter requires a SIP Trunk Security Profile change from default 
to allow unsolicited NOTIFY).


On Fri, Apr 24, 2020 at 9:35 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks


From: Jason Aarons mailto:scubajas...@gmail.com>>
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

I am doubtful porting had anything to do with it. Was it tested fully before 
the port?

Under dial peers is dtmf-relay rtp-nte set? In CUCM trunks is rfc2833 set? How 
is Unity integrated with CUCM ? SIP? CXN Version?

Without some debugs /traces I suspect you won't find much.

On Thu, Apr 23, 2020, 3:52 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Ryan Huff]

Given the statement, “We ported our numbers to SIP yesterday. Now, their main 
menu is not transferring numbers correctly”, I’m taking the change in ingress 
signaling as the change agent and assuming nothing was changed in CUC/CUCM.

I’d suspect DTMF to be the cause in this case, as this can be a common symptom 
when switching to SIP from (I’m assuming this as well) TDM (PRI).

If DTMF were an issue, a likely and more immediate fix (though I’d only 
consider this temporary, I wouldn’t leave it this way) would be to check the 
“MTP Required” option on the ingress SIP trunk(s), then save/reset the SIP 
trunk(s) and test.

Back in the day, this was thought of as a solution, but it’s not, it’s just (if 
it works) masking the issue. It’s the difference between sweeping a dirty 
floor, or just laying new carpet on top of a dirty floor. Additional there are 
resource considerations within the CUCM cluster that you’d want to be concerned 
with because “MTP Required” in a scenario like this, would cause the media 
stream in every single call leg between the phone and (assuming CUBE) to 
terminate with CUCM.

If this were to work, then a DTMF mis-match would likely be the issue and that 
could a misconfiguration with EO, codecs... etc.

Thanks,

Ryan

On Apr 24, 2020, at 09:58, Anthony Holloway  
wrote:


The reason I ask is that the troubleshooting is a little different for each 
issue.

DTMF

You would know if it's DTMF if for example, you push the button and the voice 
recording just keeps on going.  Most recordings are set such that if you barge 
in on them, the recording ends abruptly to process your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC 
processes it twice, as in double digits.  This might be a little harder to tell 
from UX, but it might be easier if you setup a test number to the Opening 
Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue.  I.e., You 
press the button, the recording stops, or even says, "Wait while I transfer 
your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few 
settings on CUBE and within CUCM which can affect how a transfer functions, 
thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question 
Anthony?

thanks
Hamu


From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as "Potential SPAM"

[Ryan Huff]

That is correct, however, your carrier has far more capabilities to compel 
other carrier than you do. Just calling 1-800–my-carrier and opening a support 
ticket probably won’t cut it. It’s a good and likely necessary start, but 
ultimately, in my experience, things like this with carriers only get solved 
when you get the money folks (account managers) involved.

No one in the carrier world cares about you till money is involved.

Sent from my iPhone

On Apr 3, 2020, at 13:58, JASON BURWELL  wrote:


The problem is my carrier says they just deliver the call, its up to the called 
party end carrier to do the CID Name dip to deliver the CID name, City/State 
-or- in this case, substitute it with “Potential SPAM” and delver with the 
call. Which as I understand it, is correct information. On the end carrier side 
I have spoken with Verizon and AT and they basically said there is nothing 
they can do and pointed the blame to 3rd party app providers which as I said 
before, I know is not the truth based on my own personal experience. I guess 
I’ll try to get it escalated at Verizon. I think I have it about up as far as I 
can with AT Thanks for all the input from everyone.

Jason

From: Ryan Huff 
Sent: Friday, April 3, 2020 1:52 PM
To: JASON BURWELL 
Cc: cisco-voip@puck.nether.net
Subject: Re: [EXTERNAL] Re: [cisco-voip] PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

You need to become a thorn in the side of the AM for your upstream carrier. 
It’s a carrier -2- carrier fight at that point.
Sent from my iPhone


On Apr 3, 2020, at 13:49, JASON BURWELL 
mailto:jason.burw...@foundersfcu.com>> wrote:

Thanks for all the replies thus far. To answer a couple of the questions that 
have come up, we are using valid, working DID numbers that we own for all 
outbound Calling Number Masks. And none of the DIDs forward to other carriers, 
they are all pointed from the PSTN to our various gateways.

One thing that was mentioned is that a SPAM autodialer bot has at some point 
spoofed some of our numbers causing them to be flagged as SPAM which is 
certainly a possibility and nothing we can do about that. I regularly get calls 
even on my cell phone with the whole “hey I missed a call form you” from the 
caller and they get irritated when I tell them, sorry I did not call you.

I know there is nothing we can do from a configuration perspective. I was just 
hoping there was some managed whitelist these carriers used that I was unaware 
of. I know there are various 3rd party apps that do this but its definitely 
something being done at the carrier level as well because I frequently get 
these messages as well on a Verizon phone and I do not have and SPAM apps or 
subscriptions.

As more and more numbers are spoofed for SPAM calls I imagine at some point all 
numbers will be flagged at potential SPAM at this rate.

So unless I missed something, it sounds like there is really nothing we can do 
about it?

Jason



From: Ryan Huff mailto:ryanh...@outlook.com>>
Sent: Friday, April 3, 2020 12:30 PM
To: JASON BURWELL 
mailto:jason.burw...@foundersfcu.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [EXTERNAL] Re: [cisco-voip] PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

CAUTION: This email originated outside of Founders Federal Credit Union. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

I’ve seen this happen on my Verizon cell recently. Was very surprised, it was 
the first time I had ever seen it.

CNAME dips and presentation are done by the called party’s carrier, so there 
isn’t anything (functionally) the calling party’s PBX can do to influence that. 
CNAME inserts are done by your upstream carrier, so if something has actually 
been modified in the CNAME database for your ANI, your upstream carrier would 
have done it.

The only real actionable thing I think you can do (besides changing your ANI to 
something else), is what you’ve done. Call your upstream carrier and give them 
call samples where your call was delivered by the called party’s carrier and 
masked with incorrect ANI. Let the carriers fight each other on the carrier 
level.
Sent from my iPhone



On Apr 3, 2020, at 12:13, JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

More and more I have users reporting that their outbound PSTN calls are showing 
as “Potential SPAM” on called party phones. Its causing some real problems 
because these are legitimate calls that the customer in many cases has 
requested but they are ignoring it due to the message and if they don’t have 
voicemail set up or its full they have the perception we are not returning 
calls. I’m assuming the Caller ID name in the national Database is being 
substituted with this message by the wireless carriers. We don’t do any 
telemarketing so there is no reason why our calls should be

Re: [cisco-voip] [EXTERNAL] Re: PSTN Calls Incorrectly Flagged as "Potential SPAM"

[Ryan Huff]

You need to become a thorn in the side of the AM for your upstream carrier. 
It’s a carrier -2- carrier fight at that point.

Sent from my iPhone

On Apr 3, 2020, at 13:49, JASON BURWELL  wrote:


Thanks for all the replies thus far. To answer a couple of the questions that 
have come up, we are using valid, working DID numbers that we own for all 
outbound Calling Number Masks. And none of the DIDs forward to other carriers, 
they are all pointed from the PSTN to our various gateways.

One thing that was mentioned is that a SPAM autodialer bot has at some point 
spoofed some of our numbers causing them to be flagged as SPAM which is 
certainly a possibility and nothing we can do about that. I regularly get calls 
even on my cell phone with the whole “hey I missed a call form you” from the 
caller and they get irritated when I tell them, sorry I did not call you.

I know there is nothing we can do from a configuration perspective. I was just 
hoping there was some managed whitelist these carriers used that I was unaware 
of. I know there are various 3rd party apps that do this but its definitely 
something being done at the carrier level as well because I frequently get 
these messages as well on a Verizon phone and I do not have and SPAM apps or 
subscriptions.

As more and more numbers are spoofed for SPAM calls I imagine at some point all 
numbers will be flagged at potential SPAM at this rate.

So unless I missed something, it sounds like there is really nothing we can do 
about it?

Jason



From: Ryan Huff 
Sent: Friday, April 3, 2020 12:30 PM
To: JASON BURWELL 
Cc: cisco-voip@puck.nether.net
Subject: [EXTERNAL] Re: [cisco-voip] PSTN Calls Incorrectly Flagged as 
"Potential SPAM"

CAUTION: This email originated outside of Founders Federal Credit Union. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

I’ve seen this happen on my Verizon cell recently. Was very surprised, it was 
the first time I had ever seen it.

CNAME dips and presentation are done by the called party’s carrier, so there 
isn’t anything (functionally) the calling party’s PBX can do to influence that. 
CNAME inserts are done by your upstream carrier, so if something has actually 
been modified in the CNAME database for your ANI, your upstream carrier would 
have done it.

The only real actionable thing I think you can do (besides changing your ANI to 
something else), is what you’ve done. Call your upstream carrier and give them 
call samples where your call was delivered by the called party’s carrier and 
masked with incorrect ANI. Let the carriers fight each other on the carrier 
level.
Sent from my iPhone


On Apr 3, 2020, at 12:13, JASON BURWELL via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:

More and more I have users reporting that their outbound PSTN calls are showing 
as “Potential SPAM” on called party phones. Its causing some real problems 
because these are legitimate calls that the customer in many cases has 
requested but they are ignoring it due to the message and if they don’t have 
voicemail set up or its full they have the perception we are not returning 
calls. I’m assuming the Caller ID name in the national Database is being 
substituted with this message by the wireless carriers. We don’t do any 
telemarketing so there is no reason why our calls should be flagged with SPAM. 
I’ve reached out and received little help from Verizon or AT Wondering what 
other are doing to get numbers “white listed” as I’m sure I’m not the only one 
facing this. Thanks Jason


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Re: [cisco-voip] PSTN Calls Incorrectly Flagged as "Potential SPAM"

[Ryan Huff]

I’ve seen this happen on my Verizon cell recently. Was very surprised, it was 
the first time I had ever seen it.

CNAME dips and presentation are done by the called party’s carrier, so there 
isn’t anything (functionally) the calling party’s PBX can do to influence that. 
CNAME inserts are done by your upstream carrier, so if something has actually 
been modified in the CNAME database for your ANI, your upstream carrier would 
have done it.

The only real actionable thing I think you can do (besides changing your ANI to 
something else), is what you’ve done. Call your upstream carrier and give them 
call samples where your call was delivered by the called party’s carrier and 
masked with incorrect ANI. Let the carriers fight each other on the carrier 
level.

Sent from my iPhone

On Apr 3, 2020, at 12:13, JASON BURWELL via cisco-voip 
 wrote:


More and more I have users reporting that their outbound PSTN calls are showing 
as “Potential SPAM” on called party phones. Its causing some real problems 
because these are legitimate calls that the customer in many cases has 
requested but they are ignoring it due to the message and if they don’t have 
voicemail set up or its full they have the perception we are not returning 
calls. I’m assuming the Caller ID name in the national Database is being 
substituted with this message by the wireless carriers. We don’t do any 
telemarketing so there is no reason why our calls should be flagged with SPAM. 
I’ve reached out and received little help from Verizon or AT Wondering what 
other are doing to get numbers “white listed” as I’m sure I’m not the only one 
facing this. Thanks Jason


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[cisco-voip] Hey everyone

[Ryan Huff]

Totally not “voice related” however...

I created a MineCraft server for my kids this evening, and I added a public 
domain to it. S, if your "co workers" are stressing you out for the next 
couple weeks, and they happen to have MineCraft Java edition, have them add the 
multiplayer server quarantine.huffytown.com. I know my kids would love meeting 
new friends "in world" and, I can also add that there WILL BE 
moderation/parental supervision on this game server .

The server is running 24/7/365 and can host 20 simultaneous players ... have 
fun!

Thanks,

Ryan
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Re: [cisco-voip] Help with Test Calls

[Ryan Huff]

Yeah, sounds like you may have the problem sorted/known, and if it truly is 
that other carrier’s have an offending/stale route that need updated... its a 
pain cause sometimes the issue isn’t always with the LEC, it can be 
upstream to the LEC too.

Having users open tickets with their carrier’s is really the only way.. and it 
can be a grueling, support ticket generating nightmare cause there is really 
nothing you can do until the call is on your upstream provider’s network.

I called from a Verizon Mobile (Atlanta Metro/GA) just a few moments ago and 
got a “Thank You for calling FastTrack Communications” prompt.

Thanks,

Ryan

On Feb 23, 2020, at 23:47, Natambu Obleton  wrote:


Hey group,

I am a member of this list because we do operate a CUCM cluster and apologize 
for misusing this list for a non-cisco related issue.

I need help troubleshooting an incoming calling issue. I run an SS7 connected 
Metaswitch in Colorado, USA and am having issues with incoming calls from USA 
based mobile phones. We did call trap with Centurylink, our tandem provider, 
and problem calls are not reaching them. They recommended having the callers 
open tickets with their local providers.  This hasn’t resulted in any success 
with getting a carrier to address this issue. This started on Thursday, 
2/20/2020, and we have been unable to reproduce it with our local phones and 
have been relying on other callers to open the tickets.

I would appreciate the USA based members of this could try to call my auto 
attendant at 970-247-3366. If you are unable to reach the recording, could you 
please open a ticket with your local carrier? Our customers are reporting 
ring-no-answer on these calls that never hit my switch.

Thank you for your time.

--

Natambu Obleton
CISSP #370414 CCIE #38491 R+Security
Director of Network Engineering and Operations
FastTrack Communications, Inc.
970.247.3366

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Re: [cisco-voip] CUC 6 Second Intervals of Silence

[Ryan Huff]

It would’ve interesting to know;

- vCPU (1 or 2)
- OVA template size
- Is unified messaging configured

Sent from my iPhone

On Feb 15, 2020, at 00:09, Anthony Holloway  
wrote:


There's post on the ciscoUC subreddit about CUC greetings sounding "garbled" 
and it turns out there is a small bit of silence being inserted on the wire by 
CUC.  Has anyone here experienced this before?  This has been confirmed on 
11.0, 11.5, 12.0 and 12.5 so far.

https://www.reddit.com/r/ciscoUC/comments/f3xi5m/unity_connection_inserts_silence_every_6_seconds/


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Re: [cisco-voip] CUCM Cluster Expansion

[Ryan Huff]

For 11.x, but I've found this helpful: 
https://www.cisco.com/web/software/283088407/126036/cucm-11.0.ova.readme.txt

Thanks,

Ryan

From: Matthew Loraditch 
Sent: Thursday, February 13, 2020 5:24 PM
To: Ryan Huff ; cisco-voip@puck.nether.net 

Subject: RE: CUCM Cluster Expansion


Yeah, I’m just trying to understand (as I read the ovf file) what the actual 
difference is between the 1000/2500 user OVA. I seem to be missing something 
(or maybe not). CPU is actually 1 less starting but same reservation, same RAM, 
same HDD.




Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: 
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From: Ryan Huff 
Sent: Thursday, February 13, 2020 5:21 PM
To: Matthew Loraditch ; 
cisco-voip@puck.nether.net
Subject: Re: CUCM Cluster Expansion



[EXTERNAL]



I wouldn't see a reason not to just up-size the two nodes you have now to the 
2.5k OVA (use 2 vCPU on each node). For the 15 pieces of flair, I'd then add in 
a 3rd 2.5k OVA w/o the CCM service enabled and run TFTP.. etc on it and give 
the pub a break.



-Ryan





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>>
Sent: Thursday, February 13, 2020 5:10 PM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] CUCM Cluster Expansion



One of my biggest customers is experiencing issues that appear to be related to 
resource utilization. I’ve never had a customer who needed more than a 2 node 
1000 user cluster.



They are getting close to some of the capacity levels listed in the sizing 
guides.



I’m looking for some opinions on what the best way to deal with this. I have 
the hardware capacity for either method.



Add a Third 1000 user Subscriber and turn off call processing and tftp on the 
Pub?



Rebuild both existing servers to 2500 user OVAs?



Add a third and do the rebuild also?



Can I just make the existing server be the 2500 capacity level? I actually 
don’t understand the difference between the 2500 and 1000 user OVAs, the 2500 
appears to actually be lesser capacity by default (1 less cpu). So go to 7500?



I’d appreciate any opinions out there. Going to be doing some reading over the 
next few days to try and figure this out.



Thanks all!



Matthew Loraditch​

Sr. Network Engineer

p: 443.541.1518

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Re: [cisco-voip] CUCM Cluster Expansion

[Ryan Huff]

I wouldn't see a reason not to just up-size the two nodes you have now to the 
2.5k OVA (use 2 vCPU on each node). For the 15 pieces of flair, I'd then add in 
a 3rd 2.5k OVA w/o the CCM service enabled and run TFTP.. etc on it and give 
the pub a break.

-Ryan


From: cisco-voip  on behalf of Matthew 
Loraditch 
Sent: Thursday, February 13, 2020 5:10 PM
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] CUCM Cluster Expansion


One of my biggest customers is experiencing issues that appear to be related to 
resource utilization. I’ve never had a customer who needed more than a 2 node 
1000 user cluster.



They are getting close to some of the capacity levels listed in the sizing 
guides.



I’m looking for some opinions on what the best way to deal with this. I have 
the hardware capacity for either method.



Add a Third 1000 user Subscriber and turn off call processing and tftp on the 
Pub?



Rebuild both existing servers to 2500 user OVAs?



Add a third and do the rebuild also?



Can I just make the existing server be the 2500 capacity level? I actually 
don’t understand the difference between the 2500 and 1000 user OVAs, the 2500 
appears to actually be lesser capacity by default (1 less cpu). So go to 7500?



I’d appreciate any opinions out there. Going to be doing some reading over the 
next few days to try and figure this out.



Thanks all!


Matthew Loraditch​
Sr. Network Engineer
p: 443.541.1518
w: 
www.heliontechnologies.com
   |  e: 
mloradi...@heliontechnologies.com
[Helion 
Technologies]
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Re: [cisco-voip] [EXT] Re: Expressway Cluster failover for MRA...

[Ryan Huff]

I think your red boxed note may be specifically apply to the part about UDS.

Sent from my iPhone

On Feb 8, 2020, at 09:16, Lelio Fulgenzi  wrote:



And what exactly is this note supposed to mean?




Sent from my iPhone

On Feb 8, 2020, at 8:56 AM, Jeffrey McHugh 
mailto:jmch...@fidelus.com>> wrote:

This has some failover info it:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-5/exwy_b_mra-expressway-deployment-guide/exwy_b_mra-expressway-deployment-guide_chapter_01.html<https://nam04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2Fexpressway%2Fconfig_guide%2FX12-5%2Fexwy_b_mra-expressway-deployment-guide%2Fexwy_b_mra-expressway-deployment-guide_chapter_01.html=02%7C01%7C%7C200453c8a19f42b080ab08d7aca179b8%7C84df9e7fe9f640afb435%7C1%7C0%7C637167681839179756=Scz9wNxzK%2Fe2l7kF0c2t3tXoGiwhKjWOL2XFyzNixbs%3D=0>

I think it’s fairly new for expressway 12.5.x guides, I don’t remember seeing 
it in 8.11 guides


Jeffrey McHugh | Practice Manager, Collaboration Services

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From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Friday, February 7, 2020 11:22 PM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [EXT] Re: [cisco-voip] Expressway Cluster failover for MRA...

So I changed the SRV records to be equal priority and weight... and everything 
works fine.

If I put a C & E pair at a site in maintenance mode, we do NOT see automatic 
reregistration of phone services to the other C/E pair at the other site.

If you log out and back in, it does automatically reregister.

If we put one of the 4 Expressways in maintenance mode, it fails over 
automatically.

How do we achieve automatic failover for MRA?


Jonathan


On Wed, Jan 29, 2020 at 3:17 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
I mean an outbound flow... offnet Jabber calls PSTN... I need it to go to 
primary DC... the only way I can force that is by lowering priority of the 
collab-edge SRV record.

To force a failover, I put the primary in maintenance mode, then Jabber times 
out and dies... log out, log ba

Re: [cisco-voip] 12.5 Upgrade files posted on CCO

[Ryan Huff]

Totally see your point.

While I guess it may be a possibility, TAC has never, in my experience, come 
close to digging that deep to look for an exit from a support request (which is 
conceivably why they would look that deep since it poses no functional risk). 
In my 20+ years of experience with TAC and Cisco, as long as the “thing” is 
under a support contract, TAC is far more willing to assist you than to look 
for an exit.

Sent from my iPhone

On Feb 6, 2020, at 12:42, Charles Goldsmith  wrote:


Agreed, but I've never done this for a customer, and here is my reasoning.  
From my understanding, when you install/upgrade, the md5 of the iso used is 
written into the logs or a file on the system.  If TAC were so inclined, they 
could tell if you installed from valid media or not.

Granted, I've had TAC supplied bootable media fail the media test and not match 
md5, but was cleared in writing to use it.

I've never had TAC check my files before, but there is always the possibility.

PUT can be a bit delayed (at least it has been for me), but never more than 
about 24 hours.  Seems we get most orders in within 12 - 18 hours.


On Thu, Feb 6, 2020 at 11:17 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I can handle the bootable issue far faster and more efficiently than the PUT 
process can, which is one of the reasons why I’ve found PUT to not be super 
useful to me.

Sent from my iPhone

On Feb 6, 2020, at 11:33, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:


As far as I know, ordering from PUT creates a sales order number that is used 
for entitlement when migrating your licenses. It also gives you a bootable ISO 
or should.



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Nick via cisco-voip
Sent: Thursday, February 6, 2020 11:20 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] 12.5 Upgrade files posted on CCO

The upgrade files for CUCM 12.5 both SU1 and SU2 both state the following

For upgrades from 12.x only. Upgrades from 11.x or earlier are requested via PUT

Is this just incorrect wording as far as i am aware there has never been any 
different files for upgrades from 11 or 12, I have ordered 12.5.1 SU1 from PUT 
and the upgrade file is identical to the one you can download from CCO 
supposedly for 12 only?

Anyone able to confirm these are the same files?
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Re: [cisco-voip] 12.5 Upgrade files posted on CCO

[Ryan Huff]

Perhaps, less inefficiencies of the tool and rather my disposition. I have 
every ISO since 7.5.x. It’s far easier for me to address offline and go, than 
to download from eDelivery (which isn’t PUT, but rather a result of the PUT 
process), so perhaps misplaced blame..

Sent from my iPhone

On Feb 6, 2020, at 12:39, Wes Sisk (wsisk)  wrote:

 When last I looked PUT fulfillment was by electronic posting and download in 
countries where that is legal. Some geo’s require physical media and the 
creates a shipping limitation.

What inefficiency do you experience with PUT?

Thanks,
Wes

On Feb 6, 2020, at 12:17 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

I can handle the bootable issue far faster and more efficiently than the PUT 
process can, which is one of the reasons why I’ve found PUT to not be super 
useful to me.

Sent from my iPhone

On Feb 6, 2020, at 11:33, Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:


As far as I know, ordering from PUT creates a sales order number that is used 
for entitlement when migrating your licenses. It also gives you a bootable ISO 
or should.



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Nick via cisco-voip
Sent: Thursday, February 6, 2020 11:20 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: [cisco-voip] 12.5 Upgrade files posted on CCO

The upgrade files for CUCM 12.5 both SU1 and SU2 both state the following

For upgrades from 12.x only. Upgrades from 11.x or earlier are requested via PUT

Is this just incorrect wording as far as i am aware there has never been any 
different files for upgrades from 11 or 12, I have ordered 12.5.1 SU1 from PUT 
and the upgrade file is identical to the one you can download from CCO 
supposedly for 12 only?

Anyone able to confirm these are the same files?
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Re: [cisco-voip] 12.5 Upgrade files posted on CCO

[Ryan Huff]

I can handle the bootable issue far faster and more efficiently than the PUT 
process can, which is one of the reasons why I’ve found PUT to not be super 
useful to me.

Sent from my iPhone

On Feb 6, 2020, at 11:33, Pawlowski, Adam  wrote:


As far as I know, ordering from PUT creates a sales order number that is used 
for entitlement when migrating your licenses. It also gives you a bootable ISO 
or should.



From: cisco-voip  On Behalf Of Nick via 
cisco-voip
Sent: Thursday, February 6, 2020 11:20 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 12.5 Upgrade files posted on CCO

The upgrade files for CUCM 12.5 both SU1 and SU2 both state the following

For upgrades from 12.x only. Upgrades from 11.x or earlier are requested via PUT

Is this just incorrect wording as far as i am aware there has never been any 
different files for upgrades from 11 or 12, I have ordered 12.5.1 SU1 from PUT 
and the upgrade file is identical to the one you can download from CCO 
supposedly for 12 only?

Anyone able to confirm these are the same files?
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Re: [cisco-voip] [EXTERNAL] 12.5 Upgrade files posted on CCO

[Ryan Huff]

The PUT has been useless to me outside of some upgrade entitlement scenarios.

Sent from my iPhone

On Feb 6, 2020, at 11:29, JASON BURWELL via cisco-voip 
 wrote:


I’ve always wondered the same thing so I’ll be interested to hear some other 
answers. PUT has never worked properly for me and no one has ever been able to 
tell me why so I pretty much just gave up on using that and I’ve downloaded 
software from CCO for upgrades.

When I went from 10.5 to 11.5 I opened up a TAC case asking about that same 
message you are asking about and was told to disregard it as it was the same 
file I would get from PUT but not sure if that holds true for all upgrade 
packages.

Jason

On Feb 6, 2020, at 11:21 AM, Nick via cisco-voip  
wrote:


CAUTION: This email originated outside of Founders Federal Credit Union. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

The upgrade files for CUCM 12.5 both SU1 and SU2 both state the following

For upgrades from 12.x only. Upgrades from 11.x or earlier are requested via PUT

Is this just incorrect wording as far as i am aware there has never been any 
different files for upgrades from 11 or 12, I have ordered 12.5.1 SU1 from PUT 
and the upgrade file is identical to the one you can download from CCO 
supposedly for 12 only?

Anyone able to confirm these are the same files?
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Re: [cisco-voip] Cisco website downloads

[Ryan Huff]

Working now.


From: Lelio Fulgenzi 
Sent: Sunday, February 2, 2020 12:05 PM
To: Anthony Holloway 
Cc: Ryan Huff ; cisco-voip voip list 

Subject: Re: [cisco-voip] Cisco website downloads


I just checked. Looks like it’s fixed?


-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<https://eur04.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Cf7349059403046ebc39508d7a80214f2%7C84df9e7fe9f640afb435%7C1%7C0%7C637162599220133081=VRdUTB2ob5IXO4Cw%2F0oSOeZxe5ln8fJWDJsaHEPoxiM%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook



[University of Guelph Cornerstone with Improve Life tagline]

On Feb 2, 2020, at 2:54 AM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

Yep.  Just tried it.  Looks like the EULA API call is returning a 400 Bad 
Request to the page.

On Sat, Feb 1, 2020 at 9:10 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Anyone else having issues with the downloads section for the Cisco website? 
I've tried multiple browsers and OSs and I get the same thing each time. I 
attempt to download an image file and get the popup modal asking to accept the 
license agreement and then after I click the accept button, the screen opacity 
decreases (as if another window opened or is about to) and then nothing 
happens. I've waited up to 10 minutes and nothing happens.

I thought perhaps an entitlement issue, but I get the same issue when trying to 
download images that I have very recently been entitled to / downloaded.



Thanks,

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

[Ryan Huff]

That seems correct. It seems like you’re speaking about an outbound flow and 
Lelio is speaking about an inbound flow.

The traversal client cluster (the CS) should know about all the peers in the 
traversal server cluster (the Es).

Sent from my iPhone

On Jan 29, 2020, at 15:21, Jonathan Charles  wrote:


OK, maybe I am misunderstanding... I have th E's paired together as a cluster 
and the C's paired together as a cluster... I have the C's initiating a UCM 
traversal client to both E's... is this not correct?


Jonathan

On Tue, Jan 28, 2020 at 7:59 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
I could be wrong here, but from what was explained to me...

You may be able to control the initial connection from off-prem device to the E 
of your choosing, but you cannot control which C that E talks to. And 
vice-versa.

So, you could point people to Ea, but they could easily be sent to Cs. And that 
traffic back from Cs could easily be sent to Es.

I was told at one time, the only option would be to put hosts in maintenance 
mode or something like that. But it wasn’t advised.

I’d love to hear other suggestions.

-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
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On Jan 28, 2020, at 8:49 PM, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

We have two pairs of Expressway clusters (C/E) at two different locations 
(primary and DR)...

The cluster is up, however, we want to make sure that we are in Active/Standby.

Currently, we have one of our SRV records for collab-edge set at 5 (the backup 
is at 10) with the same weight.

The clustering guide says we should set the priority and weight on both SRV 
records the same, which will cause half of the registrations to go to the DR 
site. It is far away and has less capability.

How do we:

1 - Make sure the primary site handles all MRA registrations and the DR site is 
only used when the primary is down.
2 = Make sure failover occurs automatically... currently Jabber users have to 
log out and back in to connect to the DR site.


Thanks!


Jonathan

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

[Ryan Huff]

Well at least the docs bumped the WAN tolerances to 80Ms instead of the 30Ms it 
used to be... so... progress? Lol

Honestly, I don’t see them changing intra-clustering communications anymore 
than it is today.

Sent from my iPhone

On Jan 29, 2020, at 13:28, Charles Goldsmith  wrote:


Yes, but when installing distributed systems (across geographically diverse 
DC's), this is better than having Core1 talk to Edge2 scenario all the time.

Hopefully this is resolved soon and we can go back to clustering.


On Wed, Jan 29, 2020 at 11:59 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
But without clustering, if Core1 fails, Edge1 will still be active and Jabber 
clients will still see Edge1 running and attempt to connect through it!

De: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
En nombre de Charles Goldsmith
Enviado el: martes, 28 de enero de 2020 23:18
Para: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
CC: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Asunto: Re: [cisco-voip] Expressway Cluster failover for MRA...

We've built them as individual pairs (Edge/Core) and then use DNS to control 
which one goes where.  Without the cluster, we know that Edge1 will always talk 
to Core1.

I get the feeling that clustering was always meant to be in the same DC, and 
for redundancy purposes in the same DC.

If you have two DC's, either a cluster at each DC, or just a pair at each DC, 
depending on the business needs.

On Tue, Jan 28, 2020 at 8:11 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

How does no. 2 actually solve the problem of having to log back in?

Is this a supported/suggested deployment method?

It’s been a while since I first looked at things and don’t recall things 
mentioning using the cluster name in the SRV records.

I’m intrigued. And interested!


-sent from mobile device-

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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 | 
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 | @UofGCCS on Instagram, Twitter and Facebook


On Jan 28, 2020, at 9:03 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
1.) It used to be in previous versions that all cluster nodes could technically 
be active at any time and SRV weights and priorities could influence the path 
selection but not guarantee it end-to-end when all cluster nodes are up and 
running.

I believe this behavior has changed/improved and I think you are supposed to be 
able to control that now with SRV weights and priorities, but I could be wrong. 
I haven’t played with Expressway clustering in a bit.

2.) As far as the Jabber registration goes; what I’ve done before in the edge 
is have the collab-edge SRV point to the edge cluster FQDN as the target. Then 
I create round robin A records for the cluster FQDN (one resolving your each 
edge server). The for the edge certs, just make sure the edge cluster fqdn is 
in the SAN.

This way if one of the edge server goes down, the Jabber client is ultimately 
still trying to resolve the same MRA FQDN via SRV lookup (this a key to Jabber 
client failover for MRA).

Thanks,

Ryan


On Jan 28, 2020, at 20:50, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


We have two pairs of Expressway clusters (C/E) at two different locations 
(primary and DR)...

The cluster is up, however, we want to make sure that we are in Active/Standby.

Currently, we have one of our SRV records for collab-edge set at 5 (the backup 
is at 10) with the same weight.

The clustering guide says we should set the priority and weight on both SRV 
records the same, which will cause half of the registrations to go to the DR 
site. It is far away and has less capability.

How do we:

1 - Make sure the primary site handles all MRA registrations and the DR site is 
only used when the primary is down.
2 = Make sure failover occurs automatically... currently Jabber users have to 
log out and back in to connect to the DR site.


Thanks!


Jonathan

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

[Ryan Huff]

Correct, the edge cluster and the control cluster are very much separate 
entities apart from themselves and load balance within their own respective 
clusters.

As long as clustering has been implemented correctly, in that scenario, the 
edge cluster would still be able to load balance and distribute resources 
within its own cluster (edge 1 answered the request but edge two transported 
the request to control 2 because control 1 was unreachable).

In that type of a degraded “brown out” scenario, you should manually put edge 1 
in maintenance mode. Not sure if you could trigger maintenance mode via API but 
if you could, then its conceivable that one could write a polling “heart beat” 
script to detect those unique “brown out” states and then automatically make 
edge 1 unserviceable by placing it into maintenance mode.

Thanks,

Ryan

On Jan 29, 2020, at 12:59, ROZA, Ariel  wrote:


But without clustering, if Core1 fails, Edge1 will still be active and Jabber 
clients will still see Edge1 running and attempt to connect through it!

De: cisco-voip  En nombre de Charles 
Goldsmith
Enviado el: martes, 28 de enero de 2020 23:18
Para: Lelio Fulgenzi 
CC: cisco-voip@puck.nether.net
Asunto: Re: [cisco-voip] Expressway Cluster failover for MRA...

We've built them as individual pairs (Edge/Core) and then use DNS to control 
which one goes where.  Without the cluster, we know that Edge1 will always talk 
to Core1.

I get the feeling that clustering was always meant to be in the same DC, and 
for redundancy purposes in the same DC.

If you have two DC's, either a cluster at each DC, or just a pair at each DC, 
depending on the business needs.

On Tue, Jan 28, 2020 at 8:11 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

How does no. 2 actually solve the problem of having to log back in?

Is this a supported/suggested deployment method?

It’s been a while since I first looked at things and don’t recall things 
mentioning using the cluster name in the SRV records.

I’m intrigued. And interested!


-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>

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 | @UofGCCS on Instagram, Twitter and Facebook


On Jan 28, 2020, at 9:03 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
1.) It used to be in previous versions that all cluster nodes could technically 
be active at any time and SRV weights and priorities could influence the path 
selection but not guarantee it end-to-end when all cluster nodes are up and 
running.

I believe this behavior has changed/improved and I think you are supposed to be 
able to control that now with SRV weights and priorities, but I could be wrong. 
I haven’t played with Expressway clustering in a bit.

2.) As far as the Jabber registration goes; what I’ve done before in the edge 
is have the collab-edge SRV point to the edge cluster FQDN as the target. Then 
I create round robin A records for the cluster FQDN (one resolving your each 
edge server). The for the edge certs, just make sure the edge cluster fqdn is 
in the SAN.

This way if one of the edge server goes down, the Jabber client is ultimately 
still trying to resolve the same MRA FQDN via SRV lookup (this a key to Jabber 
client failover for MRA).

Thanks,

Ryan


On Jan 28, 2020, at 20:50, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


We have two pairs of Expressway clusters (C/E) at two different locations 
(primary and DR)...

The cluster is up, however, we want to make sure that we are in Active/Standby.

Currently, we have one of our SRV records for collab-edge set at 5 (the backup 
is at 10) with the same weight.

The clustering guide says we should set the priority and weight on both SRV 
records the same, which will cause half of the registrations to go to the DR 
site. It is far away and has less capability.

How do we:

1 - Make sure the primary site handles all MRA registrations and the DR site is 
only used when the primary is down.
2 = Make sure failover occurs automatically... currently Jabber users have to 
log out and back in to connect to the DR site.


Thanks!


Jonathan

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

[Ryan Huff]

1.) It used to be in previous versions that all cluster nodes could technically 
be active at any time and SRV weights and priorities could influence the path 
selection but not guarantee it end-to-end when all cluster nodes are up and 
running.

I believe this behavior has changed/improved and I think you are supposed to be 
able to control that now with SRV weights and priorities, but I could be wrong. 
I haven’t played with Expressway clustering in a bit.

2.) As far as the Jabber registration goes; what I’ve done before in the edge 
is have the collab-edge SRV point to the edge cluster FQDN as the target. Then 
I create round robin A records for the cluster FQDN (one resolving your each 
edge server). The for the edge certs, just make sure the edge cluster fqdn is 
in the SAN.

This way if one of the edge server goes down, the Jabber client is ultimately 
still trying to resolve the same MRA FQDN via SRV lookup (this a key to Jabber 
client failover for MRA). 

Thanks,

Ryan

> On Jan 28, 2020, at 20:50, Jonathan Charles  wrote:
> 
> 
> We have two pairs of Expressway clusters (C/E) at two different locations 
> (primary and DR)...
> 
> The cluster is up, however, we want to make sure that we are in 
> Active/Standby.
> 
> Currently, we have one of our SRV records for collab-edge set at 5 (the 
> backup is at 10) with the same weight.
> 
> The clustering guide says we should set the priority and weight on both SRV 
> records the same, which will cause half of the registrations to go to the DR 
> site. It is far away and has less capability.
> 
> How do we:
> 
> 1 - Make sure the primary site handles all MRA registrations and the DR site 
> is only used when the primary is down.
> 2 = Make sure failover occurs automatically... currently Jabber users have to 
> log out and back in to connect to the DR site.
> 
> 
> Thanks!
> 
> 
> Jonathan 
> 
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Re: [cisco-voip] [EXTERNAL] Finesse dropping calls upon answer

[Ryan Huff]

To add on to this, you may also want to verify that if you are doing EO in 
cucm’s trunk sip profile, that you also don’t have EO set on the cube. That 
would potentially cause this issue too.

Sent from my iPhone

On Jan 17, 2020, at 17:37, Kent Roberts  wrote:

 Additional to Nicks’ response….  Check the SIP configuration and make sure 
that you have BEST Effort MTP enabled.   That way if it needs the MTP and it 
can’t get one, DTMF might not work, but your call remains.

Also, I have seen this with Delayed/Early offer on some setups and phones.   We 
had to force early offer to resolve it.  (But that could be what a carrier is 
doing as well)


ALSO!!!

If this is a new setup…. Check the codec order!!!

When we set ours up, G711 was perferred, but when the carrier sent a G729 call 
it term’d.  We were forced to swap the codec order to resolve.   (This was 8 
years ago, and things have gotten better, but…)


DEBUG CCSIP MESSAGING is your friend on the cube….   Don’t use debug ccsip all, 
if can really mess things up at the CPU side.


CUCM logs to your trunk should tell you what’s going on.



On Jan 17, 2020, at 3:30 PM, Nick Britt 
mailto:nickolasjbr...@gmail.com>> wrote:

Are you using external media resources  MTP's  (i.e. on the cube) or are you 
simply using the software MTP's on the CUCM ?

In my experience, I have seen a leg of the call from CUCM to UCCX via SIP 
provider invoke an MTP due to a DTMF Mismatch.

Then CUCM software MTP resources can randomly drop calls even when they have 
the capacity, which matches what you are stating.

If it was a codec mismatch I would imagine all calls would be dropped.

I would check software MTP services (System - > servicer parameter - > select 
active server - > Cisco IP Voice Media Streaming App (active) then the call 
count and run flag under Media termination point.

If this is enabled and you are not using external mtp resources this could be 
your issue.


On Fri, Jan 17, 2020 at 2:18 PM Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>> wrote:
outstanding, thank you all for insight on where maybe to look, or rather start 
looking.

ill see the CTI command from finess via logs, though yeah? ill have to set up 
capture I think.

for codec mismatches? Im setting those up on my cube, MediaResourcesInstance 
also on my cube?

MTP on the SIP trunk checked.

On Fri, Jan 17, 2020 at 2:09 PM Kent Roberts 
mailto:k...@fredf.org>> wrote:
Remember, call drops is not a finesse issue. Finesse is just a control 
piece for the user.Look into  CUCM/ivr.What is sending the disco 
request?Should be carrier/IVR/or UCM.Now if you see finesse send a CTI 
command to hang up the phone, thats a different issue.

Most of the time on answer drops can be a sign of codec mismatch.

Post some cucm logs, with call examples.

On Jan 17, 2020, at 2:52 PM, Nick Britt 
mailto:nickolasjbr...@gmail.com>> wrote:

What's the call volume? is MTP required ticked on the CUCM SIP trunk?

In RTMT can you check your software MTP resources?

On Fri, Jan 17, 2020 at 1:48 PM Jonatan Quezada 
mailto:jonatan.quez...@chemeketa.edu>> wrote:

UCCX version: 10.6.1.11003-29 (ES02-11)

CUCM Version: 11.5.1.15900-18


caller--->xx7899--(cube)-->CUCM>uccxTriggerAndApplication--->HelpDeskDevices>all88xx
 phones>calHandled


upon answering the call drops, but not every call! Very challenging


I imagine there are plenty of reasons why this is a thing, but in our case its 
hard to diagnose since no network changes are in effect nor have we changed 
configuration for the help desk.


what are we seeing as the top 5 or 6 reasons or rather things to re-verify 
regarding the configuration.


please advise

--
For immediate assistance please reach out to our IT help Desk 5033997899
or visit our IT Help center, located here:
https://projects.chemeketa.edu/servicedesk/customer/portals

Johnny Q
Voice Technology Analyst II
Chemeketa Community College
johnn...@chemeketa.edu
Building 22 Room 130
Work 5033995294
SIP 5035406689
FAX 5033995549

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Re: [cisco-voip] Voice traffic over SDWAN

[Ryan Huff]

If your SD-WAN solution has packet duplication (aka 0 time failover) my 
experience is that it tends to work well.

Other SD-WAN solutions may work better for VoIP if you can pin SIP/s RTP/sRTP 
to one side or the other (active/passive over active/active).

Depending on the VoIP and SD-WAN scenario, sometimes packets from call session 
A that started on network path A may end up on network path B in the SD-WAN 
appliance (when that traffic is load balanced/inspected/classified by the 
SD-WAN) and that isn’t always tolerated well by some real-time applications 
(Cisco Expressway can be uniquely sensitive to this).

Typical half duplex tcp traffic tolerates this fairly well and the 
sending/receiving applications may not even notice or see it as minor packet 
loss and initiate TCP retransmission. Full duplex traffic like media (and by 
extension real-time applications) generally do not handle that well.

Thanks,

Ryan

On Dec 19, 2019, at 15:59, Kent Roberts  wrote:

 I didn’t notice anything different then from the regular wan


Kent

On Dec 19, 2019, at 13:50, Fares Alsaafani  wrote:


Hi all,

Did anyone came recently cross deploying or POC voice over SDWAN solution.
I’m looking for thoughts , experiences , tests , things to avoid?

Fares,

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Re: [cisco-voip] Inbound MGCP call dead-air

[Ryan Huff]

Is this issue on the back of a TN port or anything like that? Have you engaged 
the carrier?

Maybe the gateway is actually working just fine?

Sent from my iPhone

On Dec 19, 2019, at 11:01, Jonathan Charles  wrote:


It is a T1 CAS and the controller is up with no errors it is 1977 over 
there.


Jonathan

On Thu, Dec 19, 2019 at 3:03 AM daniele visaggio 
mailto:visaggio.dani...@gmail.com>> wrote:
can you do a "show isdn status" ?

Is the t1 completely up?

Il giorno gio 19 dic 2019 alle ore 03:14 Jonathan Charles 
mailto:jonv...@gmail.com>> ha scritto:
We rebooted the CCM cluster and the problem persisted...

Traces show no sign of the call int he calllog...



Jonathan

On Wed, Dec 18, 2019 at 8:00 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Have you pulled a ccm trace and debug voice ccapi inout, to verif you’re not 
seeing any weirdness there?

“no mgcp / mgcp” has been known to fix weird things; I assume you’ve tried that 
though.

Sent from my iPhone

On Dec 18, 2019, at 19:59, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


Inbound call (to x2475) on an MGCP T1-CAS  E Wink

Gateway is a 4331 running 16.06.02.SPA, CCM is 11.5.1.2900-21

Get dead air and see no MGCP setup message to CCM, after 60 seconds, call times 
out (debug vpm signal and mgcp packet, below):

006469: Dec 18 18:23:38.037: htsp_dsp_message: SEND_SIG_STATUS: state=0xC 
timestamp=35130 systime=1042275951
006470: Dec 18 18:23:38.037: htsp_process_event: [0/1/1:1(21), EM_ONHOOK, 
E_DSP_SIG_1100]em_onhook_offhook
006471: Dec 18 18:23:38.037: htsp_timer - 50 msec
006472: Dec 18 18:23:38.088: htsp_process_event: [0/1/1:1(21), 
EM_QUALIFY_SEIZURE, E_HTSP_EVENT_TIMER]em_qualify_seizure_timeouthtsp_setup_ind
006473: Dec 18 18:23:38.088: [0/1/1:1(21)] get_local_station_id calling num= 
calling name= calling time=12/18 18:23  orig called=
006474: Dec 18 18:23:38.088: htsp_timer - 3000 msec
006475: Dec 18 18:23:38.090: htsp_process_event: [0/1/1:1(21), 
EM_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]em_wait_setup_ack_get_ack
006476: Dec 18 18:23:38.090: htsp_timer_stop interdigit timer cfgd to 3000
006477: Dec 18 18:23:38.090: htsp_timer2 - 127 msec
006478: Dec 18 18:23:38.217: htsp_process_event: [0/1/1:1(21), 
EM_WAIT_SETUP_ACK, E_HTSP_EVENT_TIMER2]em_wait_prewink_timer
006479: Dec 18 18:23:38.217: em_offhook (0)vnm_dsp_set_sig_state:[recEive and 
transMit0/1/1:1(21)] set signal state = 0x8em_onhook 
(200)vnm_dsp_set_sig_state:[recEive and transMit0/1/1:1(21)] set signal state = 
0x0
006480: Dec 18 18:23:38.611: htsp_digit_ready: digit = 32
006481: Dec 18 18:23:38.611: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006482: Dec 18 18:23:38.751: htsp_digit_ready: digit = 34
006483: Dec 18 18:23:38.751: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006484: Dec 18 18:23:38.892: htsp_digit_ready: digit = 37
006485: Dec 18 18:23:38.892: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006486: Dec 18 18:23:39.032: htsp_digit_ready: digit = 35
006487: Dec 18 18:23:39.032: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect

006488: Dec 18 18:23:41.335: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838278 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006489: Dec 18 18:23:41.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838278
<---

006490: Dec 18 18:23:56.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838279 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006491: Dec 18 18:23:56.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838279
<---

006492: Dec 18 18:24:11.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838280 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006493: Dec 18 18:24:11.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838280
<---

006494: Dec 18 18:24:26.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838281 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006495: Dec 18 18:24:26.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838281
<---

Configs:


controller T1 0/1/1
 framing sf
 clock source line primary
 linecode ami
 cablelength long 0db
 ds0-group 1 timeslots 1-24 type e
 description Local T1 CAS
!
mgcp
mgcp call-agent 10.48.41.80 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability mf-package
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp package-capability fm-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp bind control source-interface GigabitEthernet0/0/0
mgcp bind media source-interface GigabitEthernet0/0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior com

Re: [cisco-voip] Inbound MGCP call dead-air

[Ryan Huff]

Have you pulled a ccm trace and debug voice ccapi inout, to verif you’re not 
seeing any weirdness there?

“no mgcp / mgcp” has been known to fix weird things; I assume you’ve tried that 
though.

Sent from my iPhone

On Dec 18, 2019, at 19:59, Jonathan Charles  wrote:


Inbound call (to x2475) on an MGCP T1-CAS  E Wink

Gateway is a 4331 running 16.06.02.SPA, CCM is 11.5.1.2900-21

Get dead air and see no MGCP setup message to CCM, after 60 seconds, call times 
out (debug vpm signal and mgcp packet, below):

006469: Dec 18 18:23:38.037: htsp_dsp_message: SEND_SIG_STATUS: state=0xC 
timestamp=35130 systime=1042275951
006470: Dec 18 18:23:38.037: htsp_process_event: [0/1/1:1(21), EM_ONHOOK, 
E_DSP_SIG_1100]em_onhook_offhook
006471: Dec 18 18:23:38.037: htsp_timer - 50 msec
006472: Dec 18 18:23:38.088: htsp_process_event: [0/1/1:1(21), 
EM_QUALIFY_SEIZURE, E_HTSP_EVENT_TIMER]em_qualify_seizure_timeouthtsp_setup_ind
006473: Dec 18 18:23:38.088: [0/1/1:1(21)] get_local_station_id calling num= 
calling name= calling time=12/18 18:23  orig called=
006474: Dec 18 18:23:38.088: htsp_timer - 3000 msec
006475: Dec 18 18:23:38.090: htsp_process_event: [0/1/1:1(21), 
EM_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]em_wait_setup_ack_get_ack
006476: Dec 18 18:23:38.090: htsp_timer_stop interdigit timer cfgd to 3000
006477: Dec 18 18:23:38.090: htsp_timer2 - 127 msec
006478: Dec 18 18:23:38.217: htsp_process_event: [0/1/1:1(21), 
EM_WAIT_SETUP_ACK, E_HTSP_EVENT_TIMER2]em_wait_prewink_timer
006479: Dec 18 18:23:38.217: em_offhook (0)vnm_dsp_set_sig_state:[recEive and 
transMit0/1/1:1(21)] set signal state = 0x8em_onhook 
(200)vnm_dsp_set_sig_state:[recEive and transMit0/1/1:1(21)] set signal state = 
0x0
006480: Dec 18 18:23:38.611: htsp_digit_ready: digit = 32
006481: Dec 18 18:23:38.611: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006482: Dec 18 18:23:38.751: htsp_digit_ready: digit = 34
006483: Dec 18 18:23:38.751: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006484: Dec 18 18:23:38.892: htsp_digit_ready: digit = 37
006485: Dec 18 18:23:38.892: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect
006486: Dec 18 18:23:39.032: htsp_digit_ready: digit = 35
006487: Dec 18 18:23:39.032: htsp_process_event: [0/1/1:1(21), EM_OFFHOOK, 
E_VTSP_DIGIT]em_offhook_digit_collect

006488: Dec 18 18:23:41.335: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838278 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006489: Dec 18 18:23:41.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838278
<---

006490: Dec 18 18:23:56.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838279 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006491: Dec 18 18:23:56.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838279
<---

006492: Dec 18 18:24:11.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838280 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006493: Dec 18 18:24:11.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838280
<---

006494: Dec 18 18:24:26.336: MGCP Packet sent to 10.48.41.81:2427--->
NTFY 331838281 *@ORHC-VG4331-01.orhcnet.local MGCP 0.1
X: 0
O:
<---

006495: Dec 18 18:24:26.337: MGCP Packet received from 10.48.41.81:2427--->
200 331838281
<---

Configs:


controller T1 0/1/1
 framing sf
 clock source line primary
 linecode ami
 cablelength long 0db
 ds0-group 1 timeslots 1-24 type e
 description Local T1 CAS
!
mgcp
mgcp call-agent 10.48.41.80 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability mf-package
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp package-capability fm-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp bind control source-interface GigabitEthernet0/0/0
mgcp bind media source-interface GigabitEthernet0/0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
ccm-manager fallback-mgcp
ccm-manager redundant-host 10.48.41.81
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager config server 10.48.41.81
ccm-manager config

Any ideas on why MGCP will not send call setup to CCM?



Jonathan


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Re: [cisco-voip] best way to move CUCM Publisher from one host/DC to another

[Ryan Huff]

With ELM/PLM, I’ve not had an issue.

Sent from my iPhone

On Dec 11, 2019, at 11:47, Anthony Holloway  
wrote:


No issue, but also don't do it very much.  I just take the one that's randomly 
generated and use that as the static assignment, then redo the license request.

On Wed, Dec 11, 2019 at 9:55 AM Ryan Ratliff (rratliff) via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:
Has anyone run into problems setting a static MAC on your ELM/PLM vm? Dynamic 
mac addresses can definitely bite you but I’m curious how this workaround (that 
we do document as best practice) works in the real world.


  *   Ryan

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Charles Goldsmith mailto:w...@woka.us>>
Date: Wednesday, December 11, 2019 at 10:37 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-voip list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] best way to move CUCM Publisher from one host/DC to 
another

Yes sir, moving to new hosts in the same DC, both hosts plugged into the same 
Nexus on 10gbit.

To UC Penguin's point, it was on 6.0

On Wed, Dec 11, 2019 at 7:46 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Interesting, was it 10GB end2end (nics and all)? I’ve done it on a 1GB end2end 
and got close to 700 mbps (if I recall correctly, 680-682 was the highest it 
hit).

Not disagreeing, just interesting... it would be worth some investigating 
someday.

Sent from my iPhone


On Dec 11, 2019, at 01:50, Charles Goldsmith 
mailto:w...@woka.us>> wrote:
I'm a big fan of SCP as well, but it's limited to 1 vCPU on the encryption, so 
that seems to limit it more than the links.  I know this because trying to move 
VM's over 10gbit connections and was only getting about 400 mbps.

If you have a middle pc/jump box, I'm a big fan of simple export/import if you 
don't have a vCenter in the picture.  That way, you get a backup of the VM.  
vCenter is nice, but migration moves it, doesn't copy, even with different 
storage.

I have never tried to use vmkstools, may have to investigate that the next time 
I migrate.

On Tue, Dec 10, 2019 at 8:26 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Yes, SCP is beholden to the line rate between the hosts. Though VMWare doesn’t 
“recommend” it, I can say I’ve also never had a problem with it, FWIW... and 
yeah, super convenient.

Is you have shared storage between the hosts and can migrate the storage and 
compute, I’d power off the VM and just do that.
Sent from my iPhone


On Dec 10, 2019, at 21:20, Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
SCP is so slow and not recommended by VMware*, but damn if it's not convenient.

Ovftool is super fast but I think it requires a middle PC to be ran from.

It would be awesome if you could have the best of both worlds. Like run ovftool 
right on ESXi. I wonder.

I have used Veeam free backup to move VMs, which is as fast as ovftool, but a 
huge install for a one time move.

*To prevent performance and data management related issues on ESX, avoid the 
use of using scp, cp, or mv for storage operations; instead use vmkfstools, 
VMware's virtual machine Importer tool.
https://kb.vmware.com/s/article/1000936<https://nam10.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F1000936=02%7C01%7C%7C80e22d72f2af4a897d0008d77e59dcc9%7C84df9e7fe9f640afb435%7C1%7C0%7C637116796724447124=o2zWRU73ZWZAlaZQOI2Smin4HK50SFbd6UvJfkRXheA%3D=0>


On Tue, Dec 10, 2019, 7:50 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I don’t think vMotion would change the MAC address, UUID.. etc and I think 
you’d be fine (not while the VM is powered on though).

Typically, what I do is power the VM down and SCP the VM folder to the target 
host from the source host (requires SSH server/client be enabled and excluded 
in the host firewall for the hosts). Then in the target host, add the .vmx file 
into inventory and power on. You’ll initially be asked if you moved or copied 
the VM and you’ll want to select move (if you select copy, then it will 
randomize a few things like nic MAC .. etc).

Lastly, remove the source VM from inventory and after you’re sure the target VM 
is healthy and running fine, delete the source VM from storage on the source 
host.
Thanks,

Ryan


On Dec 10, 2019, at 20:42, naresh rathore 
mailto:nare...@hotmail.com>> wrote:
hi


We have to migrate our Voice VMs from one host/DC to another host/DC. i think 
if we clone or do vmotion, mac address gets changed and we have to apply for 
license, we may face database corruption.


Is there a way which Cisco recommends to do migration, if we have to migrate 
Voice VMs from one host to another?


Regards


Naray
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Re: [cisco-voip] best way to move CUCM Publisher from one host/DC to another

[Ryan Huff]

Interesting, was it 10GB end2end (nics and all)? I’ve done it on a 1GB end2end 
and got close to 700 mbps (if I recall correctly, 680-682 was the highest it 
hit).

Not disagreeing, just interesting... it would be worth some investigating 
someday.

Sent from my iPhone

On Dec 11, 2019, at 01:50, Charles Goldsmith  wrote:


I'm a big fan of SCP as well, but it's limited to 1 vCPU on the encryption, so 
that seems to limit it more than the links.  I know this because trying to move 
VM's over 10gbit connections and was only getting about 400 mbps.

If you have a middle pc/jump box, I'm a big fan of simple export/import if you 
don't have a vCenter in the picture.  That way, you get a backup of the VM.  
vCenter is nice, but migration moves it, doesn't copy, even with different 
storage.

I have never tried to use vmkstools, may have to investigate that the next time 
I migrate.

On Tue, Dec 10, 2019 at 8:26 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Yes, SCP is beholden to the line rate between the hosts. Though VMWare doesn’t 
“recommend” it, I can say I’ve also never had a problem with it, FWIW... and 
yeah, super convenient.

Is you have shared storage between the hosts and can migrate the storage and 
compute, I’d power off the VM and just do that.

Sent from my iPhone

On Dec 10, 2019, at 21:20, Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:


SCP is so slow and not recommended by VMware*, but damn if it's not convenient.

Ovftool is super fast but I think it requires a middle PC to be ran from.

It would be awesome if you could have the best of both worlds. Like run ovftool 
right on ESXi. I wonder.

I have used Veeam free backup to move VMs, which is as fast as ovftool, but a 
huge install for a one time move.

*To prevent performance and data management related issues on ESX, avoid the 
use of using scp, cp, or mv for storage operations; instead use vmkfstools, 
VMware's virtual machine Importer tool.
https://kb.vmware.com/s/article/1000936<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F1000936=02%7C01%7C%7Ceb93b37ca4fb479b4b6008d77e0670b0%7C84df9e7fe9f640afb435%7C1%7C0%7C637116438430643299=mFju5OvyMPfidnEhicwZeqm2AVIkyG6fcymST%2B5Txl4%3D=0>


On Tue, Dec 10, 2019, 7:50 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I don’t think vMotion would change the MAC address, UUID.. etc and I think 
you’d be fine (not while the VM is powered on though).

Typically, what I do is power the VM down and SCP the VM folder to the target 
host from the source host (requires SSH server/client be enabled and excluded 
in the host firewall for the hosts). Then in the target host, add the .vmx file 
into inventory and power on. You’ll initially be asked if you moved or copied 
the VM and you’ll want to select move (if you select copy, then it will 
randomize a few things like nic MAC .. etc).

Lastly, remove the source VM from inventory and after you’re sure the target VM 
is healthy and running fine, delete the source VM from storage on the source 
host.

Thanks,

Ryan

On Dec 10, 2019, at 20:42, naresh rathore 
mailto:nare...@hotmail.com>> wrote:


hi


We have to migrate our Voice VMs from one host/DC to another host/DC. i think 
if we clone or do vmotion, mac address gets changed and we have to apply for 
license, we may face database corruption.


Is there a way which Cisco recommends to do migration, if we have to migrate 
Voice VMs from one host to another?


Regards


Naray
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Re: [cisco-voip] best way to move CUCM Publisher from one host/DC to another

[Ryan Huff]

Yes, SCP is beholden to the line rate between the hosts. Though VMWare doesn’t 
“recommend” it, I can say I’ve also never had a problem with it, FWIW... and 
yeah, super convenient.

Is you have shared storage between the hosts and can migrate the storage and 
compute, I’d power off the VM and just do that.

Sent from my iPhone

On Dec 10, 2019, at 21:20, Anthony Holloway  
wrote:


SCP is so slow and not recommended by VMware*, but damn if it's not convenient.

Ovftool is super fast but I think it requires a middle PC to be ran from.

It would be awesome if you could have the best of both worlds. Like run ovftool 
right on ESXi. I wonder.

I have used Veeam free backup to move VMs, which is as fast as ovftool, but a 
huge install for a one time move.

*To prevent performance and data management related issues on ESX, avoid the 
use of using scp, cp, or mv for storage operations; instead use vmkfstools, 
VMware's virtual machine Importer tool.
https://kb.vmware.com/s/article/1000936<https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F1000936=02%7C01%7C%7Cab0b7bc019a5498b47cb08d77de0b304%7C84df9e7fe9f640afb435%7C1%7C0%7C63711627633116=knBH29jcaYeLJNpDcAsbUb4KgtvLDFYZdpU0FSnhy5Y%3D=0>


On Tue, Dec 10, 2019, 7:50 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I don’t think vMotion would change the MAC address, UUID.. etc and I think 
you’d be fine (not while the VM is powered on though).

Typically, what I do is power the VM down and SCP the VM folder to the target 
host from the source host (requires SSH server/client be enabled and excluded 
in the host firewall for the hosts). Then in the target host, add the .vmx file 
into inventory and power on. You’ll initially be asked if you moved or copied 
the VM and you’ll want to select move (if you select copy, then it will 
randomize a few things like nic MAC .. etc).

Lastly, remove the source VM from inventory and after you’re sure the target VM 
is healthy and running fine, delete the source VM from storage on the source 
host.

Thanks,

Ryan

On Dec 10, 2019, at 20:42, naresh rathore 
mailto:nare...@hotmail.com>> wrote:


hi


We have to migrate our Voice VMs from one host/DC to another host/DC. i think 
if we clone or do vmotion, mac address gets changed and we have to apply for 
license, we may face database corruption.


Is there a way which Cisco recommends to do migration, if we have to migrate 
Voice VMs from one host to another?


Regards


Naray
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Re: [cisco-voip] best way to move CUCM Publisher from one host/DC to another

[Ryan Huff]

I don’t think vMotion would change the MAC address, UUID.. etc and I think 
you’d be fine (not while the VM is powered on though).

Typically, what I do is power the VM down and SCP the VM folder to the target 
host from the source host (requires SSH server/client be enabled and excluded 
in the host firewall for the hosts). Then in the target host, add the .vmx file 
into inventory and power on. You’ll initially be asked if you moved or copied 
the VM and you’ll want to select move (if you select copy, then it will 
randomize a few things like nic MAC .. etc).

Lastly, remove the source VM from inventory and after you’re sure the target VM 
is healthy and running fine, delete the source VM from storage on the source 
host.

Thanks,

Ryan

On Dec 10, 2019, at 20:42, naresh rathore  wrote:


hi


We have to migrate our Voice VMs from one host/DC to another host/DC. i think 
if we clone or do vmotion, mac address gets changed and we have to apply for 
license, we may face database corruption.


Is there a way which Cisco recommends to do migration, if we have to migrate 
Voice VMs from one host to another?


Regards


Naray
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Re: [cisco-voip] CUCM 12.5.1.11900 active partition high usage

[Ryan Huff]

Apologies.. misread..

Are you seeing higher than normal CPU utilization also? Is this on the back of 
an upgrade or a new install? Was the underlying hypervisor recently upgraded?

In the 10.5/11.0 days this could have been due to SELinux denials with VMWare 
tools, but not prevalent in 12.5 to my knowledge. You could flip “utils os 
secure” to the permissive state to see if brings the consumption down a little.

https://quickview.cloudapps.cisco.com/quickview/bug/CSCux90747

Sent from my iPhone

On Dec 4, 2019, at 14:11, Erick Bergquist  wrote:


Those items are for the  logging partition, not active.

No core dumps, no kernel dumps and trace files are set to default settings.

Anyone know how to trim down usage of the root / partition, it is 13.5 gb in 
size and showing 13.0 gig used.


On Wed, Dec 4, 2019 at 11:41 AM Kent Roberts 
mailto:k...@fredf.org>> wrote:
How about crash dumps?


> On Dec 4, 2019, at 11:38 AM, Ryan Huff 
> mailto:ryanh...@outlook.com>> wrote:
>
> Are the trace collection thresholds higher than normal? Call Management 
> Records enabled?
>
> Sent from my iPhone
>
>> On Dec 4, 2019, at 12:37, Erick Bergquist 
>> mailto:erick...@gmail.com>> wrote:
>>
>> Has anyone seen where the active partition on 12.5.1.11900 is 96% full?
>>
>> 2 vCPU, 8 gig RAM, 110gig disk size.
>>
>> Seeing the free disk space slowly decrease on the active partition.
>> Not finding any bugs in bug toolkit on 12.5 for this.
>>
>> Thanks,
>> Erick
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Re: [cisco-voip] CUCM 12.5.1.11900 active partition high usage

[Ryan Huff]

Are the trace collection thresholds higher than normal? Call Management Records 
enabled?

Sent from my iPhone

> On Dec 4, 2019, at 12:37, Erick Bergquist  wrote:
> 
> Has anyone seen where the active partition on 12.5.1.11900 is 96% full?
> 
> 2 vCPU, 8 gig RAM, 110gig disk size.
> 
> Seeing the free disk space slowly decrease on the active partition.
> Not finding any bugs in bug toolkit on 12.5 for this.
> 
> Thanks,
> Erick
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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

[Ryan Huff]

Adam,

I've not encountered the streaming issue with a SIP trunk, just H.323.. 
interesting.

Out of curiosity, what code level in the ISR G2 did you go to when you 
encountered this? The behavior of simplex streaming is to send a null/fake IP 
address in the logic channel multimedia control message, which for anything 
required to check incoming RTP packets against the IP/port would fail (creating 
the silent MoH condition) which I was under the impression was only the G3 ISRs.

Thanks,

Ryan

From: cisco-voip  on behalf of Pawlowski, 
Adam 
Sent: Monday, December 2, 2019 8:50 AM
To: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks


After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.



I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn’t have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.



Adam







From: cisco-voip  On Behalf Of Mark H. 
Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



Does the problem go away if you force an MTP?



Can you provide a debug ccsip messages of an external call?



Can you share your sanitized CUBE config?



Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I’ve been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

[Ryan Huff]

Additionally, If it’s a custom MoH (not the built in Cisco tune), make sure the 
file is uploaded to each CUCM server individually; the working phone 2 phone 
scenario could be happening on nodes where the moh file is, and the trunk 
registered to a node where it is not.

Thanks,

Ryan

On Dec 2, 2019, at 02:36, daniele visaggio  wrote:


Hi,

Try to check if your trunk has access to moh resources.

This means your trunk needs to be placed inside a device pool whose mrgl 
contains at least one mrg containing at least one moh server.

Sip trunk > device pool > mrgl > mrg > moh server.

Additional test: call some external party through your sip trunk and try to put 
called party on hold. Does the called party hear moh?

Regards

On Mon, Dec 2, 2019, 07:54 Dana Tong 
mailto:dana.t...@yellit.com.au>> wrote:
Sorry forgot to mention that Supplementary services such as call forward all, 
transfer, hold/retrieve, and conferencing work fine.



From: Dana Tong
Sent: Monday, 2 December 2019 4:37 PM
To: cisco-voip@puck.nether.net
Subject: Native Call Queuing on UCM 12.5 with SIP Trunks

Hi all,

This being back on the tools is doing my head in. I think I’ve been away too 
long.

So I have configured native call queuing on UCM 12.5.x
Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.

External calls hear the initial announcement.
There is no MOH after the announcement and the external user has ring-back tone.
There is no periodic announcement.

Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?

Cheers
Dana

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Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

I may look at post-sales deployments a little differently than some...

I figure, if you’re coming to me to install, upgrade and/or repair or adjust 
this UC solution for you, it’s more than likely because you don’t have the 
capacity or depth to do it for yourself, or with your internal team.

To that end, I approach the deployment as if I’m likely to be the last person 
to touch the deployment, with specific depth-of-knowledge. So I want to leave 
the deployment as resilient and fault tolerant (to organic failure as well as 
human failure) as possible in every way possible.

I assume that once I have the UC system up and running to agreed-upon 
specifications that no one else is really going to touch the solution outside 
of administrative MACD. At which point, I become less worried about the 
resilience of the solution itself (I have that covered all day long) and more 
worried about the other actors in the environment who can impact the solution 
with their actions.

Now, if it’s a managed services scenario, that can be a horse of a different 
color because I can expect to be responsible for the entire solution’s 
operation on a day-to-day basis. I may recommend NIC teaming in that case, 
because I know I (or my team) are likely to be the only ones in the environment 
making those sorts of network changes that would be impactful to load balancing 
mechanisms on VMware hypervisors.

It all comes down to risk and understanding, which I gauge through customer 
interaction. If the customer understands the caveats to teaming and accepts 
that, great. If they are more of the “it guy that just wants the phones to 
ring”, perhaps not.

-Ryan

On Nov 28, 2019, at 14:49, Ryan Huff  wrote:

 The issues I’ve experienced with it, admittedly, have nothing to do 
(generally) with VMware or Cisco UC specifically.

The issues I’ve had to deal with (not many mind you) over the years have either 
been from an over enthusiastic network engineer who just wanted to standardize 
all LAG groups in the enterprise to LACP (and not do their due diligence 
beforehand) or a network engineer who thought they should move a channel group 
member to another switch for physical redundancy and not respect LAG rule #1 
(all bundled ports in one switch or switch stack, but not multiple switches 
trunked together).

So my preference in this regard, is really just a hedge against someone else 
doing something stupid rather than just because I think it works better one way 
over the other. Like driving a car, sometimes a good offense is a good defense 
;).

I simply do it as a matter of making my deployment more tolerant to changes 
within the network, unless the customer or business goal has a requirement for 
nic teaming, then of course, by all means.

- Ryan

On Nov 28, 2019, at 14:17, Anthony Holloway  
wrote:


Interesting comment/experience.  I have not had any issues attributed to 
loading balancing based on IP hash, and have been doing that on about 4-5 
installs a year for the last 6 years.  Not too mention the environments I'm in, 
where I was not the deployment Engineer, but support the environment 
nonetheless.  Either I'm just not seeing the issues with it, or the issues are 
not directly related to the setting.

On Thu, Nov 28, 2019 at 1:02 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
I have experienced unpleasantness in the past with IP Hash... it is not enough 
traffic to justify active/active on the trunks to risk the load balancing 
oddities that occur on the vSphere standard switch...

I am going to suggest they change it back to originating port ID and break the 
channel group.


Jonathan

On Wed, Nov 27, 2019 at 7:37 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Honestly, and this is just my preference based on my years of experience in 
post-sales engineering and my desire to not be on support calls at 
stupid-thirty AM...

For a typical Cisco UC on UCS "business edition" hypervisor setup, I would 
change the hypervisor's vSwitch load balancing mechanism to "Route based on 
originating port ID" and put the vNIC failover to active/standby (assuming just 
the two typical vmnic0/1), then on the switch, unbundle the ports from the 
channel group and make the ports individual trunk / access ports (would depend 
on how you are handling 802.1Q tags).

Active/Standby is usually a sufficient NIC failover strategy for most 
customers, in most scenarios. Unless teamed NICs on the chassis are a material 
requirement in your scenario for some reason, I'd consider un-teaming the NICs 
and just let them be active/standby.

I've not experienced where the convergence time for failover between the NICs 
is so significant that it disrupts UC communications in a meaningful way, that 
can't also be tolerated and assessed to a brief "blip". Could it cause "in 
progress" calls to fail? Probably. Could it cause calls terminated on CUCM 
(MTP) to fail? Possibly. Could it caus

Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

The issues I’ve experienced with it, admittedly, have nothing to do (generally) 
with VMware or Cisco UC specifically.

The issues I’ve had to deal with (not many mind you) over the years have either 
been from an over enthusiastic network engineer who just wanted to standardize 
all LAG groups in the enterprise to LACP (and not do their due diligence 
beforehand) or a network engineer who thought they should move a channel group 
member to another switch for physical redundancy and not respect LAG rule #1 
(all bundled ports in one switch or switch stack, but not multiple switches 
trunked together).

So my preference in this regard, is really just a hedge against someone else 
doing something stupid rather than just because I think it works better one way 
over the other. Like driving a car, sometimes a good offense is a good defense 
;).

I simply do it as a matter of making my deployment more tolerant to changes 
within the network, unless the customer or business goal has a requirement for 
nic teaming, then of course, by all means.

- Ryan

On Nov 28, 2019, at 14:17, Anthony Holloway  
wrote:


Interesting comment/experience.  I have not had any issues attributed to 
loading balancing based on IP hash, and have been doing that on about 4-5 
installs a year for the last 6 years.  Not too mention the environments I'm in, 
where I was not the deployment Engineer, but support the environment 
nonetheless.  Either I'm just not seeing the issues with it, or the issues are 
not directly related to the setting.

On Thu, Nov 28, 2019 at 1:02 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
I have experienced unpleasantness in the past with IP Hash... it is not enough 
traffic to justify active/active on the trunks to risk the load balancing 
oddities that occur on the vSphere standard switch...

I am going to suggest they change it back to originating port ID and break the 
channel group.


Jonathan

On Wed, Nov 27, 2019 at 7:37 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Honestly, and this is just my preference based on my years of experience in 
post-sales engineering and my desire to not be on support calls at 
stupid-thirty AM...

For a typical Cisco UC on UCS "business edition" hypervisor setup, I would 
change the hypervisor's vSwitch load balancing mechanism to "Route based on 
originating port ID" and put the vNIC failover to active/standby (assuming just 
the two typical vmnic0/1), then on the switch, unbundle the ports from the 
channel group and make the ports individual trunk / access ports (would depend 
on how you are handling 802.1Q tags).

Active/Standby is usually a sufficient NIC failover strategy for most 
customers, in most scenarios. Unless teamed NICs on the chassis are a material 
requirement in your scenario for some reason, I'd consider un-teaming the NICs 
and just let them be active/standby.

I've not experienced where the convergence time for failover between the NICs 
is so significant that it disrupts UC communications in a meaningful way, that 
can't also be tolerated and assessed to a brief "blip". Could it cause "in 
progress" calls to fail? Probably. Could it cause calls terminated on CUCM 
(MTP) to fail? Possibly. Could it cause disruptions to Finesse agents (if UCCX 
is in play)? Possibly. However, the convergence is very quick and is usually 
tolerated in the same way that a "brief moment of packet loss" is tolerated.

Again, evaluate whether nic teaming is a material requirement in your 
environment, but if it is not, I'd consider un-teaming and just going to 
active/standby.

Thanks,

Ryan


From: Jonathan Charles mailto:jonv...@gmail.com>>
Sent: Wednesday, November 27, 2019 7:48 PM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Preservation Mode, long time for call setups...

Would you recommend changing it to Originating Port ID?


Jonathan

On Wed, Nov 27, 2019 at 6:25 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I would expect the same behavior from PAgP with ESXi.

-Ryan

On Nov 27, 2019, at 19:19, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


They are channel group ON... (so, no LACP) on the switch...


Jonathan

On Wed, Nov 27, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
AFAIK, VMware has always required a distributed vSwitch for LACP, but the 
earliest reference I can find tonight is 5.1, though I believe it’s referenced 
the same way in the documentation of every version since then.

https://kb.vmware.com/s/article/2034277<https://nam02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F2034277=02%7C01%7C%7Ccd286c10b00648d6ebaf08d774379e12%7C84df9e7fe9f640afb435%7C1%7C0%7C637105654525924715=9QpVu56%2FrhWYyaOUC

Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

Honestly, and this is just my preference based on my years of experience in 
post-sales engineering and my desire to not be on support calls at 
stupid-thirty AM...

For a typical Cisco UC on UCS "business edition" hypervisor setup, I would 
change the hypervisor's vSwitch load balancing mechanism to "Route based on 
originating port ID" and put the vNIC failover to active/standby (assuming just 
the two typical vmnic0/1), then on the switch, unbundle the ports from the 
channel group and make the ports individual trunk / access ports (would depend 
on how you are handling 802.1Q tags).

Active/Standby is usually a sufficient NIC failover strategy for most 
customers, in most scenarios. Unless teamed NICs on the chassis are a material 
requirement in your scenario for some reason, I'd consider un-teaming the NICs 
and just let them be active/standby.

I've not experienced where the convergence time for failover between the NICs 
is so significant that it disrupts UC communications in a meaningful way, that 
can't also be tolerated and assessed to a brief "blip". Could it cause "in 
progress" calls to fail? Probably. Could it cause calls terminated on CUCM 
(MTP) to fail? Possibly. Could it cause disruptions to Finesse agents (if UCCX 
is in play)? Possibly. However, the convergence is very quick and is usually 
tolerated in the same way that a "brief moment of packet loss" is tolerated.

Again, evaluate whether nic teaming is a material requirement in your 
environment, but if it is not, I'd consider un-teaming and just going to 
active/standby.

Thanks,

Ryan


From: Jonathan Charles 
Sent: Wednesday, November 27, 2019 7:48 PM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Preservation Mode, long time for call setups...

Would you recommend changing it to Originating Port ID?


Jonathan

On Wed, Nov 27, 2019 at 6:25 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I would expect the same behavior from PAgP with ESXi.

-Ryan

On Nov 27, 2019, at 19:19, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


They are channel group ON... (so, no LACP) on the switch...


Jonathan

On Wed, Nov 27, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
AFAIK, VMware has always required a distributed vSwitch for LACP, but the 
earliest reference I can find tonight is 5.1, though I believe it’s referenced 
the same way in the documentation of every version since then.

https://kb.vmware.com/s/article/2034277<https://nam02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F2034277=02%7C01%7C%7Ca802058aed07451c4a6e08d7739cbbe1%7C84df9e7fe9f640afb435%7C1%7C0%7C637104989304335769=s%2FLBSv5M9CRT8ofriXEX8%2FguNjuLklCQ1NVFA9qwcuI%3D=0>

Sent from my iPhone

On Nov 27, 2019, at 19:11, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Route based on IP hash should be fine for 802.3ad, but technically, VMWare 
only supports it with a distributed vSwitch (would need an EA or Enterprise 
license for the hypervisor, not the “free” license) and not a standard vSwitch.
I’ve seen it work with a standard vSwitch, for long periods of time even, and 
then the CAM table on a switch gets rebuilt (switch reload, power loss ...etc), 
then all hell breaks loose and you can’t get teaming to work again.

If those c220s are business editions and/or have the “free” license (non 
enterprise), then that’s likely a problem. You’d likely see evidence of this in 
the switch syslog (Mac flaps, possibly err-disable... etc).

What is the reason for suspecting you need to change the NIC teaming to 
active/passive?

Phones going into SRST mode (may be displayed as preservation mode on phones) 
is an indication the phone’s IP lost network connectivity to all the call 
control servers listed in the phone’s configuration (xml) file.

The delayed call setup could be due to the call traversing an 
unexpected/unoptimized network path, due to disruption in it’s connection to 
its preferred call control server.

Thanks,

Ryan

On Nov 27, 2019, at 18:17, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


Customer has a two C220-M4S's with CUCM 11.5... both C-series are connected to 
the same 4-stack 3850 (port channel, mode on)

Customer is reporting Preservation Mode kicking in on the LAN and some calls 
taking a long time to setup.

Currently, VMware is set to Route based on IP Hash with PAgP channel groups.

I think we need to change it to  Route Based on Originating Virtual Port 
instead, but I cannot prove it before hand...

What could be causing the Preservation Mode on the LAN?


Jonathan


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Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

I would expect the same behavior from PAgP with ESXi.

-Ryan

On Nov 27, 2019, at 19:19, Jonathan Charles  wrote:


They are channel group ON... (so, no LACP) on the switch...


Jonathan

On Wed, Nov 27, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
AFAIK, VMware has always required a distributed vSwitch for LACP, but the 
earliest reference I can find tonight is 5.1, though I believe it’s referenced 
the same way in the documentation of every version since then.

https://kb.vmware.com/s/article/2034277<https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkb.vmware.com%2Fs%2Farticle%2F2034277=02%7C01%7C%7C68933632a9034cb6982808d77398b185%7C84df9e7fe9f640afb435%7C1%7C0%7C637104971952651559=hpirlhOhpUibcfreuD26bRnR3D2w5NtTGqjxz3f0uIU%3D=0>

Sent from my iPhone

On Nov 27, 2019, at 19:11, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Route based on IP hash should be fine for 802.3ad, but technically, VMWare 
only supports it with a distributed vSwitch (would need an EA or Enterprise 
license for the hypervisor, not the “free” license) and not a standard vSwitch.
I’ve seen it work with a standard vSwitch, for long periods of time even, and 
then the CAM table on a switch gets rebuilt (switch reload, power loss ...etc), 
then all hell breaks loose and you can’t get teaming to work again.

If those c220s are business editions and/or have the “free” license (non 
enterprise), then that’s likely a problem. You’d likely see evidence of this in 
the switch syslog (Mac flaps, possibly err-disable... etc).

What is the reason for suspecting you need to change the NIC teaming to 
active/passive?

Phones going into SRST mode (may be displayed as preservation mode on phones) 
is an indication the phone’s IP lost network connectivity to all the call 
control servers listed in the phone’s configuration (xml) file.

The delayed call setup could be due to the call traversing an 
unexpected/unoptimized network path, due to disruption in it’s connection to 
its preferred call control server.

Thanks,

Ryan

On Nov 27, 2019, at 18:17, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


Customer has a two C220-M4S's with CUCM 11.5... both C-series are connected to 
the same 4-stack 3850 (port channel, mode on)

Customer is reporting Preservation Mode kicking in on the LAN and some calls 
taking a long time to setup.

Currently, VMware is set to Route based on IP Hash with PAgP channel groups.

I think we need to change it to  Route Based on Originating Virtual Port 
instead, but I cannot prove it before hand...

What could be causing the Preservation Mode on the LAN?


Jonathan


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Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

Well, at the very least, I would say that’s an unsupported configuration (using 
LACP on ESXi with a standard switch); that alone should be enough justification 
to change that. As for “proof”, you’d see it in the switch’s syslog.

However, the matter at hand (preservation mode), is due to a loss of 
connectivity, which could absolutely be due to the unsupported vSwitch 
configuration.

-Ryan

On Nov 27, 2019, at 19:18, Jonathan Charles  wrote:


I remembered another UCS set to IP Hash and having all kinds of connectivity 
issues... this is a standard switch, not a distributed and just the basic 
license.

On Wed, Nov 27, 2019 at 6:11 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Route based on IP hash should be fine for 802.3ad, but technically, VMWare only 
supports it with a distributed vSwitch (would need an EA or Enterprise license 
for the hypervisor, not the “free” license) and not a standard vSwitch.
I’ve seen it work with a standard vSwitch, for long periods of time even, and 
then the CAM table on a switch gets rebuilt (switch reload, power loss ...etc), 
then all hell breaks loose and you can’t get teaming to work again.

If those c220s are business editions and/or have the “free” license (non 
enterprise), then that’s likely a problem. You’d likely see evidence of this in 
the switch syslog (Mac flaps, possibly err-disable... etc).

What is the reason for suspecting you need to change the NIC teaming to 
active/passive?

Phones going into SRST mode (may be displayed as preservation mode on phones) 
is an indication the phone’s IP lost network connectivity to all the call 
control servers listed in the phone’s configuration (xml) file.

The delayed call setup could be due to the call traversing an 
unexpected/unoptimized network path, due to disruption in it’s connection to 
its preferred call control server.

Thanks,

Ryan

> On Nov 27, 2019, at 18:17, Jonathan Charles 
> mailto:jonv...@gmail.com>> wrote:
>
> 
> Customer has a two C220-M4S's with CUCM 11.5... both C-series are connected 
> to the same 4-stack 3850 (port channel, mode on)
>
> Customer is reporting Preservation Mode kicking in on the LAN and some calls 
> taking a long time to setup.
>
> Currently, VMware is set to Route based on IP Hash with PAgP channel groups.
>
> I think we need to change it to  Route Based on Originating Virtual Port 
> instead, but I cannot prove it before hand...
>
> What could be causing the Preservation Mode on the LAN?
>
>
> Jonathan
>
>
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Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

AFAIK, VMware has always required a distributed vSwitch for LACP, but the 
earliest reference I can find tonight is 5.1, though I believe it’s referenced 
the same way in the documentation of every version since then.

https://kb.vmware.com/s/article/2034277

Sent from my iPhone

On Nov 27, 2019, at 19:11, Ryan Huff  wrote:

Route based on IP hash should be fine for 802.3ad, but technically, VMWare 
only supports it with a distributed vSwitch (would need an EA or Enterprise 
license for the hypervisor, not the “free” license) and not a standard vSwitch.
I’ve seen it work with a standard vSwitch, for long periods of time even, and 
then the CAM table on a switch gets rebuilt (switch reload, power loss ...etc), 
then all hell breaks loose and you can’t get teaming to work again.

If those c220s are business editions and/or have the “free” license (non 
enterprise), then that’s likely a problem. You’d likely see evidence of this in 
the switch syslog (Mac flaps, possibly err-disable... etc).

What is the reason for suspecting you need to change the NIC teaming to 
active/passive?

Phones going into SRST mode (may be displayed as preservation mode on phones) 
is an indication the phone’s IP lost network connectivity to all the call 
control servers listed in the phone’s configuration (xml) file.

The delayed call setup could be due to the call traversing an 
unexpected/unoptimized network path, due to disruption in it’s connection to 
its preferred call control server.

Thanks,

Ryan

On Nov 27, 2019, at 18:17, Jonathan Charles  wrote:


Customer has a two C220-M4S's with CUCM 11.5... both C-series are connected to 
the same 4-stack 3850 (port channel, mode on)

Customer is reporting Preservation Mode kicking in on the LAN and some calls 
taking a long time to setup.

Currently, VMware is set to Route based on IP Hash with PAgP channel groups.

I think we need to change it to  Route Based on Originating Virtual Port 
instead, but I cannot prove it before hand...

What could be causing the Preservation Mode on the LAN?


Jonathan


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Re: [cisco-voip] Preservation Mode, long time for call setups...

[Ryan Huff]

Route based on IP hash should be fine for 802.3ad, but technically, VMWare only 
supports it with a distributed vSwitch (would need an EA or Enterprise license 
for the hypervisor, not the “free” license) and not a standard vSwitch.
I’ve seen it work with a standard vSwitch, for long periods of time even, and 
then the CAM table on a switch gets rebuilt (switch reload, power loss ...etc), 
then all hell breaks loose and you can’t get teaming to work again.

If those c220s are business editions and/or have the “free” license (non 
enterprise), then that’s likely a problem. You’d likely see evidence of this in 
the switch syslog (Mac flaps, possibly err-disable... etc).

What is the reason for suspecting you need to change the NIC teaming to 
active/passive?

Phones going into SRST mode (may be displayed as preservation mode on phones) 
is an indication the phone’s IP lost network connectivity to all the call 
control servers listed in the phone’s configuration (xml) file.

The delayed call setup could be due to the call traversing an 
unexpected/unoptimized network path, due to disruption in it’s connection to 
its preferred call control server.

Thanks,

Ryan

> On Nov 27, 2019, at 18:17, Jonathan Charles  wrote:
> 
> 
> Customer has a two C220-M4S's with CUCM 11.5... both C-series are connected 
> to the same 4-stack 3850 (port channel, mode on)
> 
> Customer is reporting Preservation Mode kicking in on the LAN and some calls 
> taking a long time to setup.
> 
> Currently, VMware is set to Route based on IP Hash with PAgP channel groups.
> 
> I think we need to change it to  Route Based on Originating Virtual Port 
> instead, but I cannot prove it before hand...
> 
> What could be causing the Preservation Mode on the LAN?
> 
> 
> Jonathan
> 
> 
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Re: [cisco-voip] unityconnection db replication issue

[Ryan Huff]

I would do a few things things;

- utils NTP status (replace any NTP server higher than stratum 3). Cisco says 
stratum 3 or lower is where you need to be; while stratum 5 isn’t incredibly 
high, it may be just a little too high.

I’m not a fan of using fqdns for NTP servers, try to use IP addresses and while 
I’d prefer you use an internal NTP server (to your network) that references 
external NTP sources, directly specifying external NTP servers on a UC server 
is better than having an unreliably high stratum. Use “utils NTP server”... 
list, add and delete to manipulate.

- Next, on both CUC servers, after you get NTP straightened out, from the CLI 
(primary then HA), issue “utils service restart Cisco Tomcat”. Wait about 20 
after the primary, then do the HA.

After about 20 minutes from when you restarted the tomcat service on the HA, do 
“utils dbreplication runtimestate” from the CLI of the primary. If you still 
don’t have “(2) setup complete” for both nodes, you’ll probably need to 
manually reset cluster replication.

To do that, perform the following in order specified (all from the respective 
CLIs;


1.) “utils dbreplication stop all”  (only on the primary server)

2.) “utils dbreplication dropadmindb” (first on the HA server, then the primary 
server)

3.) “utils dbreplication reset all” (only on the primary server)


This process overall could take an hour or more, so be patient.


Afterwards, try “utils dbreplication runtimestate” from the CLI of the primary 
server and see what you get, you should see “(2) Setup Complete” on both nodes.


If you do, then run “show CUC cluster status” on the CLI of the primary server, 
and verify the primary server is the Unity Connection DB primary. 
Alternatively, go to “Tools > Cluster” under Unity Connection Servicability in 
the Web Admin GUI and make sure the primary server, is the Unity Connection DB 
primary.


If you still don’t have “(2) Setup Complete” for both nodes, I’d call Cisco TAC 
and get a case going.

Thanks,

Ryan

On Nov 25, 2019, at 20:43, Jinto Alakkal Kunjumon  wrote:


Hi Naray,

In addition you may also verify below


1. “Cisco DB, A Cisco DB Replicator, Cisco  Database layer Monitor” these 
services should be running.

2. Informix database also rely heavily on DNS, if you have DNS configured both 
forward and reverse DNS lookup should work.


Thanks!
Jinto.



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From: cisco-voip  on behalf of naresh 
rathore 
Sent: Monday, November 25, 2019 8:29:56 PM
To: Ryan Huff 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] unityconnection db replication issue

hi Ryan



following is the output


admin:utils diagnose test

Log file: platform/log/diag1.log

Starting diagnostic test(s)
===
test - disk_space  : Passed (available: 17668 MB, used: 10920 MB)
skip - disk_files  : This module must be run directly and off hours
test - service_manager : Passed
test - tomcat  : Passed
test - tomcat_deadlocks: Passed
test - tomcat_keystore : Passed
test - tomcat_connectors   : Passed
test - tomcat_threads  : Passed
test - tomcat_memory   : Passed
test - tomcat_sessions : Failed - The following web applications have an 
unusually large number of active sessions: vmrest.  Please collect all of the 
Tomcat logs for root cause analysis: file get activelog tomcat/logs/*
skip - tomcat_heapdump : This module must be run directly and off hours
test - validate_network: Passed
test - raid: Passed
test - system_info : Passed (Collected system information in diagnostic 
log)
test - ntp_reachability: Passed
test - ntp_clock_drift : Passed
test - ntp_stratum : Failed
The reference NTP server is a stratum 5 clock.
NTP servers with stratum 5 or worse clocks are deemed unreliable.
Please consider using an NTP server with better stratum level.

Please use OS Admin GUI to add/delete NTP servers.

skip - sdl_fragmentation   : This module must be run directly and off hours
skip - sdi_fragmentation   : This module must be run directly and off hours

Diagnostics Completed


 The final output will be in Log file: platform/log/diag1.log


 Please use 'file view activelog platform/log/diag1.log' command to see the 
output

admin:


Regards


Naray

____
From: Ryan Huff 
Sent: Tuesday, November 26, 2019 6:27 AM
To: naresh rathore 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] unityconnection db replication issue

Run “ utils diagnose test “ from the primary CLI; everything pass?

Sent from my iPhone

On Nov 25, 2019, at 20:11, naresh rathore  wrote:


hi,


unityconnection version is Version 11.5.1

Re: [cisco-voip] unityconnection db replication issue

[Ryan Huff]

Run “ utils diagnose test “ from the primary CLI; everything pass?

Sent from my iPhone

On Nov 25, 2019, at 20:11, naresh rathore  wrote:


hi,


unityconnection version is Version 11.5.1.14900-11 . I am currently facing db 
replication issue. following is the output of utils dbreplication status and 
runtimestate.

admin:utils dbreplication status

connect to rapcucon01_ccm11_5_1_14900_11 failed
Enterprise Replication not active  (62)
command failed -- unable to connect to server specified  (5)

Replication status check is now running in background.
Use command 'utils dbreplication runtimestate' to check its progress

The final output will be in file 
cm/trace/dbl/sdi/ReplicationStatus.2019_11_26_11_31_03.out

Please use "file view activelog 
cm/trace/dbl/sdi/ReplicationStatus.2019_11_26_11_31_03.out " command to see the 
output
admin:utils dbreplication runtimestate

Server Time: Tue Nov 26 11:36:38 AEDT 2019

Cluster Replication State: dbmonpreflightcheck file is found locally, please 
check again in few minutes...

DB Version: ccm11_5_1_14900_11

Repltimeout set to: 300s
PROCESS option set to: 1


Cluster Detailed View from RAPCUCON01 (2 Servers):

  PING  DB/RPC/   REPL.Replication  
  REPLICATION SETUP
SERVER-NAME IP ADDRESS(msec)DbMon?QUEUEGroup ID 
  (RTMT) & Details
--- -------   ----  
  --
RAPCUCON02  10.163.72.111 0.133 Y/Y/Y --   (-)  
  (-) Replication Not Setup
RAPCUCON01  10.163.72.110 0.021 Y/Y/Y --   (-)  
  (-) Replication Not Setup



on RTMT, i can see following message
On Tue Nov 26 11:41:19 AEDT 2019; alert DBReplicationFailure has occured. 
Counter Replicate_State of Number of Replicates Created and State of 
Replication(ReplicateCount) on node RAPCUCON01 has state value of 4. 
ReasonCode: Replication Setup did not succeed.



I am able to ping between the two, both unity connection are on the same subnet 
and there is no firewall in between.


I also used the following link to restart dbreplication


https://www.cisco.com/c/en/us/support/docs/unified-communications/unity-connection/116942-technote-uc-00.html


Regards


Naray


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Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

[Ryan Huff]

Have you tried adding the IMP identity cert into the Expressway trust? It 
shouldn’t have to work that way, but if it does, might point to an issue with 
how the CA chain is being recognized in the trust.

Also, make sure to do a full reboot of the Expressway node after adding certs 
into the truststore (again, you shouldn’t have to do that but I’ve seen this 
work before).

Sent from my iPhone

On Nov 17, 2019, at 18:58, Jonathan Charles  wrote:


When I try to refresh the IMP nodes, I get Failed: Unable to communicate with 
[[IMPNODE] CryptoError: Decryption failure.

On Sun, Nov 17, 2019 at 5:54 PM Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
I re-uploaded the root and intermediate CA certificate... still get the same 
error...

I also tried adding a new AXL user... same error...


Jonathan

On Sun, Nov 17, 2019 at 5:48 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Likely certificate / trust issues..

Sent from my iPhone

On Nov 17, 2019, at 18:36, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:


Yep, we are running into clustering issues...

Getting Inactive: (Remote host is reachable but connection is not established. 
Either refresh this page, or check the credentials.)

For IMP connection, so MRA is down...

Still looking for a fix...


Jonathan

On Fri, Nov 15, 2019 at 7:17 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
I’ve done 2 8.11.x to 12.5.5 fine (clustered setup, 4). There is a bug with 
clustering to watch out for but I did not encounter it. The 12.5 Cisco download 
page has a note and link about this.

Currently working on jabberd process high memory consumption issue on one node 
that has been present since 8.11.x which 12.5 had memory leak fix for but still 
an issue. Slow memory increase over time just on one of the edge nodes.

Going to look over 12.5.6 release notes now

Erick



On Fri, Nov 15, 2019 at 3:28 PM Matt Jacobson 
mailto:m4ttjacob...@gmail.com>> wrote:
If that is the case, then I would double check that it is supported. In the 
release notes there is a chart for supported platforms based on serial numbers. 
If it is a legacy Tandberg box, then I suspect 12.x may not work out for you.

On Fri, Nov 15, 2019 at 14:30 Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
This is a legacy Tandberg VCS for video only... no MRA, no remote phones... 
just inbound and outbound sip video...


Jonathan

On Fri, Nov 15, 2019 at 12:44 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
We’re at 12.5.3 and probably moving to 12.5.5/12.5.6 somewhere in the Holiday 
timeframe when everything quiets down a bit.

There hasn’t been really any significant issue upgrading from 8 -> 12, but 
there have been a couple of bugs that largely are all resolved by deleting and 
rebuilding whatever the thing is that is misbehaving.

The requirement for the _cup_login and _cisco-uds SRVs went away though it 
still endlessly logs a warning about not finding them, but it will work.

You do also gain the ability to play with the openssl cipher strings but in my 
limited experience trying to change those to bump them up a notch, it ends up 
breaking XMPP or something.

Adam

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Friday, November 15, 2019 11:59 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Thanks, the latest is 12.5.6, released last week, I am avoiding it like the 
plague...and the bug fix doesn't apply to us.

I am going with 12.5.5 (released in August).

I already have release keys (Cisco AM sent them over)...

Hybrid services are on a separate VCS-C that is already 12.5.

My plan is to get new certs if we have any issues


Thanks!


Jonathan

On Fri, Nov 15, 2019 at 10:46 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
A couple of thoughts for you...


  *   Get the software release key for 12.x now (you'll be asked to enter it 
during the upgrade in the GUI). You'll need to work with TAC > GLO for this if 
(and I assume this would be your case) the existing 8.7 serial is active in 
Cisco's licensing system. The caveat to trying to do this with Cisco's 
self-service license re-host tool is that while the 8.7 serial is active, it 
won't allow you to assign the new 12.x software release PAK to the serial 
because the serial is already assigned to another software release key.

 *   Take a backup first, your only roll back option is to re-install 8.7 
and restore the backup.

  *   Your VMware Hypervisor needs to be 6.0/5/7.

  *   If you have Hybrid Services configured, make sure the management 
connector is up to date first.

  *   SSL Certificate validation changed a bit in 8.8+

 *   Verify proper forward / reverse DNS for all the relevant touch points
 *   Make sure the Expressway certificate trus

Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

[Ryan Huff]

Likely certificate / trust issues..

Sent from my iPhone

On Nov 17, 2019, at 18:36, Jonathan Charles  wrote:


Yep, we are running into clustering issues...

Getting Inactive: (Remote host is reachable but connection is not established. 
Either refresh this page, or check the credentials.)

For IMP connection, so MRA is down...

Still looking for a fix...


Jonathan

On Fri, Nov 15, 2019 at 7:17 PM Erick Bergquist 
mailto:erick...@gmail.com>> wrote:
I’ve done 2 8.11.x to 12.5.5 fine (clustered setup, 4). There is a bug with 
clustering to watch out for but I did not encounter it. The 12.5 Cisco download 
page has a note and link about this.

Currently working on jabberd process high memory consumption issue on one node 
that has been present since 8.11.x which 12.5 had memory leak fix for but still 
an issue. Slow memory increase over time just on one of the edge nodes.

Going to look over 12.5.6 release notes now

Erick



On Fri, Nov 15, 2019 at 3:28 PM Matt Jacobson 
mailto:m4ttjacob...@gmail.com>> wrote:
If that is the case, then I would double check that it is supported. In the 
release notes there is a chart for supported platforms based on serial numbers. 
If it is a legacy Tandberg box, then I suspect 12.x may not work out for you.

On Fri, Nov 15, 2019 at 14:30 Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:
This is a legacy Tandberg VCS for video only... no MRA, no remote phones... 
just inbound and outbound sip video...


Jonathan

On Fri, Nov 15, 2019 at 12:44 PM Pawlowski, Adam 
mailto:aj...@buffalo.edu>> wrote:
We’re at 12.5.3 and probably moving to 12.5.5/12.5.6 somewhere in the Holiday 
timeframe when everything quiets down a bit.

There hasn’t been really any significant issue upgrading from 8 -> 12, but 
there have been a couple of bugs that largely are all resolved by deleting and 
rebuilding whatever the thing is that is misbehaving.

The requirement for the _cup_login and _cisco-uds SRVs went away though it 
still endlessly logs a warning about not finding them, but it will work.

You do also gain the ability to play with the openssl cipher strings but in my 
limited experience trying to change those to bump them up a notch, it ends up 
breaking XMPP or something.

Adam

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Jonathan Charles
Sent: Friday, November 15, 2019 11:59 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Thanks, the latest is 12.5.6, released last week, I am avoiding it like the 
plague...and the bug fix doesn't apply to us.

I am going with 12.5.5 (released in August).

I already have release keys (Cisco AM sent them over)...

Hybrid services are on a separate VCS-C that is already 12.5.

My plan is to get new certs if we have any issues


Thanks!


Jonathan

On Fri, Nov 15, 2019 at 10:46 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
A couple of thoughts for you...


  *   Get the software release key for 12.x now (you'll be asked to enter it 
during the upgrade in the GUI). You'll need to work with TAC > GLO for this if 
(and I assume this would be your case) the existing 8.7 serial is active in 
Cisco's licensing system. The caveat to trying to do this with Cisco's 
self-service license re-host tool is that while the 8.7 serial is active, it 
won't allow you to assign the new 12.x software release PAK to the serial 
because the serial is already assigned to another software release key.

 *   Take a backup first, your only roll back option is to re-install 8.7 
and restore the backup.

  *   Your VMware Hypervisor needs to be 6.0/5/7.

  *   If you have Hybrid Services configured, make sure the management 
connector is up to date first.

  *   SSL Certificate validation changed a bit in 8.8+

 *   Verify proper forward / reverse DNS for all the relevant touch points
 *   Make sure the Expressway certificate trust is up-to-date with all the 
current CUCM,CUC,IMP identity certificates (self-signed) or CA certificates 
(public CA signed certificates).
 *   no duplicate certificates in the Expressway trusts
Beyond that, just pay attention to the caveats list in the upgrade doc for your 
version of 12.5.x (12.5.4 is the latest I think).

Thanks,

Ryan


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Sent: Friday, November 15, 2019 10:57 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Can we just upgrade directly or do we need to go to an intermediary version 
first?

Also, any gotchas besides new certificates?


Jonathan
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Re: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

[Ryan Huff]

A couple of thoughts for you...


  *   Get the software release key for 12.x now (you'll be asked to enter it 
during the upgrade in the GUI). You'll need to work with TAC > GLO for this if 
(and I assume this would be your case) the existing 8.7 serial is active in 
Cisco's licensing system. The caveat to trying to do this with Cisco's 
self-service license re-host tool is that while the 8.7 serial is active, it 
won't allow you to assign the new 12.x software release PAK to the serial 
because the serial is already assigned to another software release key.

 *   Take a backup first, your only roll back option is to re-install 8.7 
and restore the backup.

  *   Your VMware Hypervisor needs to be 6.0/5/7.

  *   If you have Hybrid Services configured, make sure the management 
connector is up to date first.

  *   SSL Certificate validation changed a bit in 8.8+
 *   Verify proper forward / reverse DNS for all the relevant touch points
 *   Make sure the Expressway certificate trust is up-to-date with all the 
current CUCM,CUC,IMP identity certificates (self-signed) or CA certificates 
(public CA signed certificates).
 *   no duplicate certificates in the Expressway trusts

Beyond that, just pay attention to the caveats list in the upgrade doc for your 
version of 12.5.x (12.5.4 is the latest I think).

Thanks,

Ryan


From: cisco-voip  on behalf of Jonathan 
Charles 
Sent: Friday, November 15, 2019 10:57 AM
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] VCS Expressway upgrade, 8.7 to 12.5

Can we just upgrade directly or do we need to go to an intermediary version 
first?

Also, any gotchas besides new certificates?


Jonathan
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Re: [cisco-voip] DNS and LDAP Domain name change - current process node is IP

[Ryan Huff]

From a server perspective, just make sure the forward A record and reverse PTR 
record for the new FQDN exist BEFORE using the CLI command to run the sanity 
check scripts to change the domain. You can change the DNS records shortly 
before running the CLI command, but not for long as it would eventually cause 
cluster replication issues. Make sure the reverse PTR for the old FQDN is 
removed/changed to point at the NEW FQDN.

Regarding the processNode names.. no real impact to leave them as IP references 
(changing them to FQDN can offer some advantages and conveniences when dealing 
with MRA, Expressway, IM & Presence).

If you do decide to change CUCM’s server references to FQDN at some point, make 
sure all server nodes have a forward and reverse DNS record and make sure all 
phones/devices have access to DNS servers that can resolve the CUCM server’s 
FQDN (this step is really important). Also, make sure to adjust/verify the 
Enterprise Parameter URLs for authentication and directories (though they can 
usually be left to use IP references without issue).

Certs are regenerated; so with public CA certs that means a new CSR and certs 
after the change. With self-signed certs, you just get new certs that’ll need 
to be re-trusted by tour browser or imported into your device’s truststore.

Sent from my iPhone

On Nov 11, 2019, at 18:47, Nick Britt  wrote:


Sorry the ask it so change from the DNS suffix of 
customername.us.com<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcustomername.us.com=02%7C01%7C%7C2609025ca7364935d96f08d767019242%7C84df9e7fe9f640afb435%7C1%7C0%7C637091128749175619=PbdJpfrUqvvG7cS7uBJq0ll6fv47o0R5mj1EWkTkb4c%3D=0>
 to 
customer.com<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcustomer.com=02%7C01%7C%7C2609025ca7364935d96f08d767019242%7C84df9e7fe9f640afb435%7C1%7C0%7C637091128749175619=eIpn94qwo9a%2BYBGJs1XEJl1JAQVy5Qyb5KyzxNitE%2Fk%3D=0>.

Also the users from 
customername.us.com<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcustomername.us.com=02%7C01%7C%7C2609025ca7364935d96f08d767019242%7C84df9e7fe9f640afb435%7C1%7C0%7C637091128749185630=UdD3B4rmbmUnK03CRqmlmBToeKW9Fk3DZywPFf79Ygg%3D=0>
 have been moved into a  
customername.com<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fcustomername.com=02%7C01%7C%7C2609025ca7364935d96f08d767019242%7C84df9e7fe9f640afb435%7C1%7C0%7C637091128749195642=kJeOUvsQs1AZ5xYP7IipTdqAga%2FXGa13gpJRc0QFK8A%3D=0>
 OU on different LDAP servers with the same usernames.

The servers are configured with a DNS domain and DNS servers but they process 
node ID is the IP address of the servers (without the suffix)

Does that make sense?

On Mon, Nov 11, 2019 at 3:26 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
what are you trying to do? Do you need to add a domain name to UC servers that 
currently do not have a domain name?

Sent from my iPhone

On Nov 11, 2019, at 18:21, Nick Britt 
mailto:nickolasjbr...@gmail.com>> wrote:


A customer has had a domain name, this includes the DNS and the active 
directory integration. I am trying to pull together the necessary steps for 
each application.

Below is what I have deduced from the documentation so far

Change Domain name CUCM, Pub and Sub

The CUCM processNode name is the IP address (System - > Server) changing the 
domain name will have no effect on the CTL/ITL files as phones only reference 
the IP currently.
Remove each server from PLM and add back in post-change
Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.
Current Active directory authentication and LDAP authentication will be moved 
from one server to another. The usernames are the same between the the Active 
Directory domains so the device associations should remain when the LDAP 
integration is change between one AD domain to another.

Change Domain name IMP, Pub and Sub

Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.

CUC
Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.
LDAP is used to manually "import" user name/extension then the users are added 
manually
SMTP is used for voicemail to mail integration instead of unified messaging so 
no changes needed as the mail server details remains the same.

I've also seen reports of mgcp sccp gw's unregistering if relying on DNS but 
the IP is used for each MGCP registration.

I would appreciate a heads up if you have encountered any issues with similar 
changes.

--
- Nick
_

Re: [cisco-voip] DNS and LDAP Domain name change - current process node is IP

[Ryan Huff]

what are you trying to do? Do you need to add a domain name to UC servers that 
currently do not have a domain name?

Sent from my iPhone

On Nov 11, 2019, at 18:21, Nick Britt  wrote:


A customer has had a domain name, this includes the DNS and the active 
directory integration. I am trying to pull together the necessary steps for 
each application.

Below is what I have deduced from the documentation so far

Change Domain name CUCM, Pub and Sub

The CUCM processNode name is the IP address (System - > Server) changing the 
domain name will have no effect on the CTL/ITL files as phones only reference 
the IP currently.
Remove each server from PLM and add back in post-change
Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.
Current Active directory authentication and LDAP authentication will be moved 
from one server to another. The usernames are the same between the the Active 
Directory domains so the device associations should remain when the LDAP 
integration is change between one AD domain to another.

Change Domain name IMP, Pub and Sub

Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.

CUC
Security certs will need to be re-signed by the root CA
Each domain name and DNS change will need to be completed independently and db 
replication status to be checked before moving onto subscriber.
LDAP is used to manually "import" user name/extension then the users are added 
manually
SMTP is used for voicemail to mail integration instead of unified messaging so 
no changes needed as the mail server details remains the same.

I've also seen reports of mgcp sccp gw's unregistering if relying on DNS but 
the IP is used for each MGCP registration.

I would appreciate a heads up if you have encountered any issues with similar 
changes.

--
- Nick
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Re: [cisco-voip] Expressway cluster certificates.

[Ryan Huff]

So having more certs than need in the Truststore generally wont cause issues, 
it’s just one more certificate that can potentially be trusted.

As long as the new certificates are signed by the same internal CA as the one 
that is currently in the truststore for CUCM (all nodes), then you shouldn’t 
need to have the identity certificates in the truststore.

One reason that may have been done is because the original person wasn’t able 
to get CUCM to properly recognize the internal CA and trust certificates signed 
by it.

This could happen if the CA chain was uploaded incorrectly. The root should be 
uploaded first, then any intermediates.

Sent from my iPhone

On Oct 14, 2019, at 17:40, ROZA, Ariel  wrote:


Hi Ryan,

Both Expressway servers are signed by the internal CA. I have uploaded the root 
and intermediate certificates, too.
But I am renewing the certificates on an existing cluster, and whoever 
instelled it, they manually added the ExpC certs into tomcat-trust.

So, I understand that it would be safe to remove the ExpC certs from 
tomcat-trust and everything would be working fine?
What about the use the cluster name/don´t use the cluster name contradiction?

Thanks,

Ariel.

De: Ryan Huff 
Enviado el: lunes, 14 de octubre de 2019 18:14
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] Expressway cluster certificates.

Are the expressway-C server using self-signed certificates (I doubt it because 
you said they are multi-san)?

Generally, CUCM doesn’t need to trust the identity certificate (unless it is 
self signed). In all other cases, CUCM needs to trust the certificate authority 
the signed the expressway-c certificates.

If for example, GoDaddy signed the SSL certificates for the Expressway-C, CUCM 
just needs to trust the GoDaddy certificate authority chain.
Sent from my iPhone


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Re: [cisco-voip] Expressway cluster certificates.

[Ryan Huff]

Are the expressway-C server using self-signed certificates (I doubt it because 
you said they are multi-san)?

Generally, CUCM doesn’t need to trust the identity certificate (unless it is 
self signed). In all other cases, CUCM needs to trust the certificate authority 
the signed the expressway-c certificates.

If for example, GoDaddy signed the SSL certificates for the Expressway-C, CUCM 
just needs to trust the GoDaddy certificate authority chain.

Sent from my iPhone

On Oct 14, 2019, at 17:07, ROZA, Ariel  wrote:


Hi, Guys

I am renewing the certificates in an Expressway X8.10.1 cluster. But I am 
running into a conflict between the official documentation and how CUCM works.

I have set both Expressway-C certificates to use the Cluster name for the 
Common Name and each server´s name as a SAN, as the oficial guide states.
But when I load both signed certificates into CUCM trust stores, it shows only 
one of the certificates, instead of both, as CUCM uses the CN tu build its 
listo f certs, and both ExpC´s CN is the same (Although they are two diferente 
certificates)

So, I started to re-read all related documents I could find and I found some 
contradictions that I do not now how to solve.

On one hand, I have the official “Certificate  Creation and Deployment Guide” 
that states:

“A certificate identifies the Expressway. It contains names by which it is 
known and to which traffic is routed. If the Expressway is known by multiple 
names for these purposes, such as if it is part of a cluster, this must be 
represented in the X.509 subject data, according to the guidance of RFC5922. 
The certificate must contain the FQDN of both the Expressway itself and of the 
cluster. The following lists show what must be included in the X.509 subject, 
depending on the deployment model chosen.
If the Expressway is not clustered:
■ Subject Common Name = FQDN of Expressway
■ Subject Alternate Names = leave blank*
If the Expressway is clustered, with individual certificates per Expressway:
■ Subject Common Name = FQDN of cluster
■ Subject Alternate Name = FQDN of Expressway peer, FQDN of cluster*

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-10/Cisco-Expressway-Certificate-Creation-and-Use-Deployment-Guide-X8-10.pdf

On the other hand I have the “Configure and Troubleshoot Collaboration Edge 
(MRA) Certificates“ that says:

Cluster Certificates

It is strongly recommended that if you have a cluster of Expressway-C or 
Expressway-E servers for redundancy that you generate a separate CSR for each 
server and have it signed by a CA.  Most deployments will use the server name 
for the subject and list all peers and the cluster ID as SANs.  It is possible 
for you to use the cluster-id as the subject to use the same certificate for 
all nodes in the cluster, therefore avoiding the cost of multiple certs signed 
by a public CA.  If absolutely necessary, this can be done with the following 
process or by using OpenSSL to generate both the private key and CSR manually:

Step 1.  Generate a CSR on the master of the cluster and configure it to list 
the cluster-alias as the subject.  Add all peers in the cluster as alternative 
names, along with all other required SANs.

Step 2.  Sign this CSR and upload to the master peer.

Step 3.  Log into the master as root and download the private key located in 
/tandberg/persistent/certs.

Step 4.  Upload both the signed certificate and matching private key to each 
other peer in the cluster.

Note: This is not recommended for the following reasons:
1. It is a security risk because all peers are using the same private key.  If 
one is somehow compromised an attacker can decrypt traffic from any of the 
servers.
2.  If a change needs to be made to the certificate, this entire process must 
be followed again rather than a simple CSR generation and signing.

https://www.cisco.com/c/en/us/support/docs/unified-communications/expressway/213872-configure-and-troubleshoot-collaboration.html#anc17


So, one says to use the cluster name, the other says the opposite. And I have 
the CUCM showing me only one cert intead of two.


What should I do? Re-sign both certificates with the peer name as CN 

Re: [cisco-voip] UCCX not working after change to SIP

[Ryan Huff]

Could be that the UCCX trigger’s partition isn’t in the SIP trunk’s inbound 
CSS. I assume the trigger’s css can reach the partition of the CTI ports as 
that doesn’t seem like something that would have been changed in a scenario 
like this.

Sent from my iPhone

On Oct 13, 2019, at 22:20, Kent Roberts  wrote:

 Ryan, that’s what I was thinking, but when Dana said to a station works fine, 
I started thinking a bit like some of the fun and weird messages that show up 
with codecs, etc.   (basically codec mismatch and a bounce off to another 
number/pattern and that failed it….   Might need a few logs to see what its 
doing with uccx in the mix.



On Oct 13, 2019, at 8:15 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Sort of sounds like you may have a Class of Service (CSS/Partition) issue with 
the SIP trunk between CUBE and CUCM, since that is most likely the most 
significant configuration change/add since moving to SIP, that could generate 
404 responses.

Could also be an issue with the device pool on the SIP trunk, depending on how 
you have the dialplan structured.

CUCM Generally won’t issue a station not found (404) message, unless CUCM 
ingress element (the SIP trunk) doesn’t find the station; either because the 
station doesn’t truly exist, or doesn’t exist in the configuration context it 
has access to.

Regional problems would issue a 404 generally.

Sent from my iPhone

On Oct 13, 2019, at 21:58, Dana Tong 
mailto:dana.t...@yellit.com.au>> wrote:


Hi all,

Customer recently changed from ISDN gateways (H323 registered) to centralised 
SIP and now the UCCX IVR isn’t working.

The SIP trunk is set to use the 64K region and I did also try setting the CODEC 
to both G.711alaw and ulaw on the dial-peers on the SIP CUBE router.
It’s weird.

The debug CCSIP messages are showing 404 not found, and the voip dialpeer inout 
were showing a match on the correct dial-peers.
But then the logs say no match as per below.

Anyone seen this kind of error after changing from ISDN to SIP?

Somewhat sanitised below. Trying to get RTMT access.



Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.52.3.164:5060;branch=z9hG4bKritgbh304ou76jse28v0.1
From: ;tag=39906919-1571014929892-
To: "777 777";tag=EDD3AA89-2562
Date: Mon, 14 Oct 2019 11:02:09 GMT
Call-ID: 
BW120209892141019-860012483@10.10.10.10<mailto:BW120209892141019-860012483@10.10.10.10>
CSeq: 602191603 INVITE
Allow-Events: telephone-event
Warning: 399 10.10.253.10 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Reason: Q.850;cause=1
Content-Length: 0
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Re: [cisco-voip] UCCX not working after change to SIP

[Ryan Huff]

*** Regional issues would NOT cause a 404 response generally.

Sent from my iPhone

On Oct 13, 2019, at 22:15, Ryan Huff  wrote:

 Sort of sounds like you may have a Class of Service (CSS/Partition) issue 
with the SIP trunk between CUBE and CUCM, since that is most likely the most 
significant configuration change/add since moving to SIP, that could generate 
404 responses.

Could also be an issue with the device pool on the SIP trunk, depending on how 
you have the dialplan structured.

CUCM Generally won’t issue a station not found (404) message, unless CUCM 
ingress element (the SIP trunk) doesn’t find the station; either because the 
station doesn’t truly exist, or doesn’t exist in the configuration context it 
has access to.

Regional problems would issue a 404 generally.

Sent from my iPhone

On Oct 13, 2019, at 21:58, Dana Tong  wrote:


Hi all,

Customer recently changed from ISDN gateways (H323 registered) to centralised 
SIP and now the UCCX IVR isn’t working.

The SIP trunk is set to use the 64K region and I did also try setting the CODEC 
to both G.711alaw and ulaw on the dial-peers on the SIP CUBE router.
It’s weird.

The debug CCSIP messages are showing 404 not found, and the voip dialpeer inout 
were showing a match on the correct dial-peers.
But then the logs say no match as per below.

Anyone seen this kind of error after changing from ISDN to SIP?

Somewhat sanitised below. Trying to get RTMT access.



Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.52.3.164:5060;branch=z9hG4bKritgbh304ou76jse28v0.1
From: ;tag=39906919-1571014929892-
To: "777 777";tag=EDD3AA89-2562
Date: Mon, 14 Oct 2019 11:02:09 GMT
Call-ID: BW120209892141019-860012483@10.10.10.10
CSeq: 602191603 INVITE
Allow-Events: telephone-event
Warning: 399 10.10.253.10 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Reason: Q.850;cause=1
Content-Length: 0
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Re: [cisco-voip] UCCX not working after change to SIP

[Ryan Huff]

Sort of sounds like you may have a Class of Service (CSS/Partition) issue with 
the SIP trunk between CUBE and CUCM, since that is most likely the most 
significant configuration change/add since moving to SIP, that could generate 
404 responses.

Could also be an issue with the device pool on the SIP trunk, depending on how 
you have the dialplan structured.

CUCM Generally won’t issue a station not found (404) message, unless CUCM 
ingress element (the SIP trunk) doesn’t find the station; either because the 
station doesn’t truly exist, or doesn’t exist in the configuration context it 
has access to.

Regional problems would issue a 404 generally.

Sent from my iPhone

On Oct 13, 2019, at 21:58, Dana Tong  wrote:


Hi all,

Customer recently changed from ISDN gateways (H323 registered) to centralised 
SIP and now the UCCX IVR isn’t working.

The SIP trunk is set to use the 64K region and I did also try setting the CODEC 
to both G.711alaw and ulaw on the dial-peers on the SIP CUBE router.
It’s weird.

The debug CCSIP messages are showing 404 not found, and the voip dialpeer inout 
were showing a match on the correct dial-peers.
But then the logs say no match as per below.

Anyone seen this kind of error after changing from ISDN to SIP?

Somewhat sanitised below. Trying to get RTMT access.



Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.52.3.164:5060;branch=z9hG4bKritgbh304ou76jse28v0.1
From: ;tag=39906919-1571014929892-
To: "777 777";tag=EDD3AA89-2562
Date: Mon, 14 Oct 2019 11:02:09 GMT
Call-ID: BW120209892141019-860012483@10.10.10.10
CSeq: 602191603 INVITE
Allow-Events: telephone-event
Warning: 399 10.10.253.10 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Reason: Q.850;cause=1
Content-Length: 0
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Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

Webex Hybrid Calling definitely sounds like a good fit for you then; it’ll also 
give your cloud registered devices a way to dial the on-prem extensions.

Basically, when the cloud registered device is setup, you select Hybrid calling 
as the PSTN service (assuming Hybrid calling has already been setup) and then 
it sends signaling to CUCM via Expressway-C > CUCM. The effective media path is 
device <> cloud.

If you have Expressway B2B, you can also leverage that to allow your cloud 
devices to make B2B SIP calls via Cloud > Expressway-C > CUCM > Expressway-C > 
Expressway-E > Internet. The idea was to make Hybrid Calling for cloud devices 
“transparent” to the user over cloud calling in terms of PSTN capabilities, 
with the added feature of interacting with on-prem extensions as if the device 
was registered on-prem.

There are more than a few scenarios where Webex Hybrid calling will trombone 
the call, and it’s by design and due to the nature of the scenario. Under the 
hood, the call legs and SIP messages can get hairy from a troubleshooting 
perspective (TranslatorX is a beast for this), and Cisco has had more than a 
few complaints about it, but it is what it is.

Sent from my iPhone

On Oct 4, 2019, at 23:40, Lelio Fulgenzi  wrote:



I’m trying to do a bit of everything, really.

In our case, I’d like to have a few cloud registered WebEx room devices still 
be able to call our extensions. It’s the one thing we loose vs. on-prem reg 
WebEx room devices.

I still have to get it working (I’m guessing there are IP address ranges I have 
to permit) but a cloud registered device can call @myphone.acme.com

If I can create a macro on cloud registered devices like you can on CE devices, 
then it gives me that functionality.

We don’t have Webex Teams deployed. We don’t have Webex pstn / calling enabled.

So it’s either a hybrid call setup or a macro.

I’ll have to investigate further.

-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<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Cb16503f6614e48ab04dc08d74945d472%7C84df9e7fe9f640afb435%7C1%7C0%7C637058436587224475=K8Hkz7W%2F3ijLyQTh0Nc6ZgnMbiXTurUDovMLe%2FjqT6k%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 11:25 PM, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

Wait, I thought this was for other businesses to call you.  Are you saying that 
to call within your own cloud you have to dial that giant URI?  Is there no 
directory, or extension dialing?  Clearly, I have not done a single Webex 
calling deployment yet.

On Fri, Oct 4, 2019 at 10:06 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Ok. Looking at it from the other way around, could I create a macro(?) on the 
cloud registered devices that ask for a 5 digit extension and then add the 
appropriate SIP domain to the extension to place the call?

-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<https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Cb16503f6614e48ab04dc08d74945d472%7C84df9e7fe9f640afb435%7C1%7C0%7C637058436587234480=hP1iUi1T8p7kIR8cnYIG6z3JKod80gGGYp47f1seNCA%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 10:32 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

It’s a little bit of an ugly trombone, but it does work..

Sent from my iPhone

On Oct 4, 2019, at 22:09, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:



Darn. Double darn.

Let’s hope webex offers up custom domain registration for devices soon.

‘Cause room...@acme.rooms.webex.com<mailto:room...@acme.rooms.webex.com> is a 
bit much.

-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutritio

Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

Right... but the codec is registered in the Webex cloud (in this scenario), so 
you need to share the uri with the hybrid remote destination to trombone the 
call back out to the cloud if you’re letting the call come all the way into 
CUCM, and do it in a “least amount of MACD way”.

Sent from my iPhone

On Oct 4, 2019, at 22:57, NateCCIE  wrote:


Doesn’t cucm have the ability to look at the user portion of the URI only?  For 
like when you’re routing to a DN?  Or I think you can add the short domain to 
the list of the CUCM “owned” domains in enterprise parameters.

From: Ryan Huff 
Sent: Friday, October 4, 2019 8:51 PM
To: NateCCIE 
Cc: Lelio Fulgenzi ; cisco-voip voyp list 

Subject: Re: [cisco-voip] SIP Domain substitution

Hey Nate ... the original ask I think, was to do it all with DNS only and no 
intervention at layer 4, which to my knowledge, DNS alone couldn’t do.

Expressway search rule, CUCM LUA script... etc could all do it in reality.

However, the actual goal appears to be dialing a Webex cloud registered codec, 
using a non cloud uri (...@rooms.webex.com<mailto:...@rooms.webex.com>), and 
for that Webex Hybrid calling with Expressway B2B would get you there, and also 
checks the “no additional transformation needed” box.

Sent from my iPhone


On Oct 4, 2019, at 22:41, NateCCIE 
mailto:natec...@gmail.com>> wrote:
 I am not thinking right?  Can’t a dns srv get the call routed to a specific 
host? Then a quick expressway transform to change the domain, and you’re done.

Think of it as a different internal domain va external domain.

f...@company.com<mailto:f...@company.com> does goes to 
expressway.companyinfrastructuredomain.com which does a quick trans to 
foo@internal.local<mailto:foo@internal.local>


Sent from my iPhone


On Oct 4, 2019, at 8:36 PM, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Interesting. I’ll have to look into that.  Thx.
-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<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Ca4b60cc4d36b4fc2019808d7493fb875%7C84df9e7fe9f640afb435%7C1%7C0%7C637058410328173751=twi7VQkh2ncJnN1kEK%2F05epIfJ9YypR1V1WRZ2ys5oM%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 10:32 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

It’s a little bit of an ugly trombone, but it does work..

Sent from my iPhone


On Oct 4, 2019, at 22:09, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Darn. Double darn.

Let’s hope webex offers up custom domain registration for devices soon.

‘Cause room...@acme.rooms.webex.com<mailto:room...@acme.rooms.webex.com> is a 
bit much.
-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<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Ca4b60cc4d36b4fc2019808d7493fb875%7C84df9e7fe9f640afb435%7C1%7C0%7C637058410328183761=X%2BFjx4BuROhc4qzL%2ByLbLOImNEqfUQDU30uEy6CDFvY%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 9:05 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
What it sounds like you are trying to do to me, is allow the call to ultimately 
setup with a URI different than the URI that was dialed, without the calling 
party being the wiser.

DNS won’t be able to do anything with regards to that I don’t think, because it 
really sounds like you’re trying to manipulate/transform the called URI, and 
you’ll need something to interact with the SIP message stack for that I’d think.

You can create a round robin A record, that resolves to multiple IP addresses, 
so when the client looks up the DNS SRV, it receives multiple targets to try 
before considering the SRV target “unreachable” (SRV weights and priorities 
determine the ordering of the target addresses resolved for the client). 
However, this won

Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

Hey Nate ... the original ask I think, was to do it all with DNS only and no 
intervention at layer 4, which to my knowledge, DNS alone couldn’t do.

Expressway search rule, CUCM LUA script... etc could all do it in reality.

However, the actual goal appears to be dialing a Webex cloud registered codec, 
using a non cloud uri (...@rooms.webex.com), and for that Webex Hybrid calling 
with Expressway B2B would get you there, and also checks the “no additional 
transformation needed” box.

Sent from my iPhone

On Oct 4, 2019, at 22:41, NateCCIE  wrote:

 I am not thinking right?  Can’t a dns srv get the call routed to a specific 
host? Then a quick expressway transform to change the domain, and you’re done.

Think of it as a different internal domain va external domain.

f...@company.com does goes to expressway.companyinfrastructuredomain.com which 
does a quick trans to foo@internal.local


Sent from my iPhone

On Oct 4, 2019, at 8:36 PM, Lelio Fulgenzi  wrote:



Interesting. I’ll have to look into that.  Thx.

-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<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C9b720a4af69b41bebb5008d7493d7691%7C84df9e7fe9f640afb435%7C1%7C0%7C637058400631419029=Bm%2FOh%2Fp3TDYkOYis27I47D3rrDAJDEp4doaBw5lr9XA%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 10:32 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

It’s a little bit of an ugly trombone, but it does work..

Sent from my iPhone

On Oct 4, 2019, at 22:09, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:



Darn. Double darn.

Let’s hope webex offers up custom domain registration for devices soon.

‘Cause room...@acme.rooms.webex.com<mailto:room...@acme.rooms.webex.com> is a 
bit much.

-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<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C9b720a4af69b41bebb5008d7493d7691%7C84df9e7fe9f640afb435%7C1%7C0%7C637058400631419029=Bm%2FOh%2Fp3TDYkOYis27I47D3rrDAJDEp4doaBw5lr9XA%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 9:05 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

What it sounds like you are trying to do to me, is allow the call to ultimately 
setup with a URI different than the URI that was dialed, without the calling 
party being the wiser.

DNS won’t be able to do anything with regards to that I don’t think, because it 
really sounds like you’re trying to manipulate/transform the called URI, and 
you’ll need something to interact with the SIP message stack for that I’d think.

You can create a round robin A record, that resolves to multiple IP addresses, 
so when the client looks up the DNS SRV, it receives multiple targets to try 
before considering the SRV target “unreachable” (SRV weights and priorities 
determine the ordering of the target addresses resolved for the client). 
However, this won’t have the ability to change the called URI, which is 
ultimately what I think you’re attempting in the scenario (DNS and SIP messages 
are on different networking layers).

As Dave mentioned below, Expressway or a LUA script (sip normalization) in CUCM 
seems to be uniquely qualified for what you’re wanting to do.

Sent from my iPhone

On Oct 4, 2019, at 20:40, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:



I’ve seen some references to Cisco SIP proxy server.

Would that help?

-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<https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7

Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

Webex Hybrid Calling (with Expressway B2B), could in theory, help accomplish 
this. The codec is still cloud registered, though Hybrid calling would allow 
for an on-prem URI to be associated with the Webex remote destination of the 
codec.

The call would come into the on-prem URI via B2B like normal, and assuming the 
Hybrid integration was setup correctly, ring the Webex remote destination which 
rings the cloud registered codec.

It’s a little bit of an ugly trombone, but it does work..

Sent from my iPhone

On Oct 4, 2019, at 22:09, Lelio Fulgenzi  wrote:



Darn. Double darn.

Let’s hope webex offers up custom domain registration for devices soon.

‘Cause room...@acme.rooms.webex.com<mailto:room...@acme.rooms.webex.com> is a 
bit much.

-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<https://eur04.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C91eb00070c1b45238c9d08d749390724%7C84df9e7fe9f640afb435%7C1%7C0%7C637058381594062845=NZ1r%2BfF%2FekCgt%2FARyjJ3rxBVHLHQXVdDERhMIM4sISE%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 9:05 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

What it sounds like you are trying to do to me, is allow the call to ultimately 
setup with a URI different than the URI that was dialed, without the calling 
party being the wiser.

DNS won’t be able to do anything with regards to that I don’t think, because it 
really sounds like you’re trying to manipulate/transform the called URI, and 
you’ll need something to interact with the SIP message stack for that I’d think.

You can create a round robin A record, that resolves to multiple IP addresses, 
so when the client looks up the DNS SRV, it receives multiple targets to try 
before considering the SRV target “unreachable” (SRV weights and priorities 
determine the ordering of the target addresses resolved for the client). 
However, this won’t have the ability to change the called URI, which is 
ultimately what I think you’re attempting in the scenario (DNS and SIP messages 
are on different networking layers).

As Dave mentioned below, Expressway or a LUA script (sip normalization) in CUCM 
seems to be uniquely qualified for what you’re wanting to do.

Sent from my iPhone

On Oct 4, 2019, at 20:40, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:



I’ve seen some references to Cisco SIP proxy server.

Would that help?

-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<https://eur04.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C91eb00070c1b45238c9d08d749390724%7C84df9e7fe9f640afb435%7C1%7C0%7C637058381594072853=LEFEOBUPvVtGUiBc8mGoRoSp1EOwAw%2FzWr1y4QRrkj0%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 7:46 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

According to RFC 2782 
(https://www.ietf.org/rfc/rfc2782.txt<https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.ietf.org%2Frfc%2Frfc2782.txt=02%7C01%7C%7C91eb00070c1b45238c9d08d749390724%7C84df9e7fe9f640afb435%7C1%7C0%7C637058381594072853=%2FzjXq6A407ULW1tL6YQHpLhkShhif1%2FacmD9VUNKLNE%3D=0>),
 it does not, under the “Target Definition”; “there must be one or more address 
records for this name, the name must not be an alias”.

However, I can tell you that I have used a CNAME in the SRV target field 
before, and it appeared to work at the time. Still, depending on the 
application, doing so could potentially cause some weird issue with regards to 
PTR or something.

Sent from my iPhone

On Oct 4, 2019, at 19:10, Brian Meade mailto:bmead...@vt.edu>> 
wrote:


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

On Fri, Oct 4, 2019 at 12:26 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Yeah – I’d want this to happen all within DNS. But of course, in a supported 
fashion. I’m not interested in spending time modifying infrastructure at this 
time.

I’ve done some searching, and there’s talk of RR records, but we haven’t found 
much documentation.


---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON 

Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

What it sounds like you are trying to do to me, is allow the call to ultimately 
setup with a URI different than the URI that was dialed, without the calling 
party being the wiser.

DNS won’t be able to do anything with regards to that I don’t think, because it 
really sounds like you’re trying to manipulate/transform the called URI, and 
you’ll need something to interact with the SIP message stack for that I’d think.

You can create a round robin A record, that resolves to multiple IP addresses, 
so when the client looks up the DNS SRV, it receives multiple targets to try 
before considering the SRV target “unreachable” (SRV weights and priorities 
determine the ordering of the target addresses resolved for the client). 
However, this won’t have the ability to change the called URI, which is 
ultimately what I think you’re attempting in the scenario (DNS and SIP messages 
are on different networking layers).

As Dave mentioned below, Expressway or a LUA script (sip normalization) in CUCM 
seems to be uniquely qualified for what you’re wanting to do.

Sent from my iPhone

On Oct 4, 2019, at 20:40, Lelio Fulgenzi  wrote:



I’ve seen some references to Cisco SIP proxy server.

Would that help?

-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
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 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Oct 4, 2019, at 7:46 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

According to RFC 2782 
(https://www.ietf.org/rfc/rfc2782.txt<https://nam04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.ietf.org%2Frfc%2Frfc2782.txt=02%7C01%7C%7C8d67c4a26a104ee8d3e808d7492c8ff2%7C84df9e7fe9f640afb435%7C1%7C0%7C637058328048087715=IuzbX7iIL%2B701WKSoqOVA8jUKD2xWXx5ACgHIlyxvsI%3D=0>),
 it does not, under the “Target Definition”; “there must be one or more address 
records for this name, the name must not be an alias”.

However, I can tell you that I have used a CNAME in the SRV target field 
before, and it appeared to work at the time. Still, depending on the 
application, doing so could potentially cause some weird issue with regards to 
PTR or something.

Sent from my iPhone

On Oct 4, 2019, at 19:10, Brian Meade mailto:bmead...@vt.edu>> 
wrote:


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

On Fri, Oct 4, 2019 at 12:26 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Yeah – I’d want this to happen all within DNS. But of course, in a supported 
fashion. I’m not interested in spending time modifying infrastructure at this 
time.

I’ve done some searching, and there’s talk of RR records, but we haven’t found 
much documentation.


---
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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://nam04.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7C8d67c4a26a104ee8d3e808d7492c8ff2%7C84df9e7fe9f640afb435%7C1%7C0%7C637058328048097726=nq9QNHJ9%2F0%2Fx6MWj0warAw6sAdZy120eZOYLfOmLtYw%3D=0>
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From: Dave Goodwin mailto:dave.good...@december.net>>
Sent: Friday, October 4, 2019 12:09 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: cisco-voip voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] SIP Domain substitution

Are you wanting this to all happen within DNS instead of happening within a SIP 
UA? As far as I understand, if DNS redirected somewhere (SRV or CNAME record 
for example) it would not change the destination URI the originator is trying 
to reach. The SIP protocol has redirection codes (such as 301 or 302) but 
whether or how you might be able to use them depends on the SIP UAs being used.

You might also be able to use something like a SIP normalization script (CUCM), 
SIP profiles (CUBE), or maybe search pattern replacements (Expressway) to just 
translate the domain as calls flow in/out. I'm guessing what might be feasible 
without knowing more of the picture.

On Fri, Oct 4, 2019 at 11:10 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Does SIP allow for domain name substitution?

By this I mean, instead of advertising or dialing 
coy...@phones.

Re: [cisco-voip] SIP Domain substitution

[Ryan Huff]

According to RFC 2782 (https://www.ietf.org/rfc/rfc2782.txt), it does not, 
under the “Target Definition”; “there must be one or more address records for 
this name, the name must not be an alias”.

However, I can tell you that I have used a CNAME in the SRV target field 
before, and it appeared to work at the time. Still, depending on the 
application, doing so could potentially cause some weird issue with regards to 
PTR or something.

Sent from my iPhone

On Oct 4, 2019, at 19:10, Brian Meade  wrote:


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

On Fri, Oct 4, 2019 at 12:26 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Yeah – I’d want this to happen all within DNS. But of course, in a supported 
fashion. I’m not interested in spending time modifying infrastructure at this 
time.

I’ve done some searching, and there’s talk of RR records, but we haven’t found 
much documentation.


---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: Dave Goodwin mailto:dave.good...@december.net>>
Sent: Friday, October 4, 2019 12:09 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: cisco-voip voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] SIP Domain substitution

Are you wanting this to all happen within DNS instead of happening within a SIP 
UA? As far as I understand, if DNS redirected somewhere (SRV or CNAME record 
for example) it would not change the destination URI the originator is trying 
to reach. The SIP protocol has redirection codes (such as 301 or 302) but 
whether or how you might be able to use them depends on the SIP UAs being used.

You might also be able to use something like a SIP normalization script (CUCM), 
SIP profiles (CUBE), or maybe search pattern replacements (Expressway) to just 
translate the domain as calls flow in/out. I'm guessing what might be feasible 
without knowing more of the picture.

On Fri, Oct 4, 2019 at 11:10 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Does SIP allow for domain name substitution?

By this I mean, instead of advertising or dialing 
coy...@phones.america.acmemanufacturing.com
 I want to use coy...@zing.com

But I don’t want to have to reorganize and reprogram anything.

I just want the DNS to say, “hey, use this domain instead and try again.”
-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | 
le...@uoguelph.ca

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

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Re: [cisco-voip] IP Phone 8811 backgrounds

[Ryan Huff]

I find that on-prem CUCM is still used fairly substantially.. :).

The thumbnail needs to be PNG and 139x109 (not 159x109). I’d also double check 
the spelling and casing of the directory in TFTP; Desktops/800x400

Your images must be in PNG format and the dimensions of the full sized image 
must be within 800 pixels by 480 pixels. Thumbnail images are 139 pixels 
(width) by 109 pixels (height).

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8800-series/english/adminguide/P881_BK_C136782F_00_cisco-ip-phone-8800_series/P881_BK_C136782F_00_cisco-ip-phone-8811-8841_chapter_01010.html#concept_325325E245AAC3BF76989A0D4E66A47F

Sent from my iPhone

On Sep 23, 2019, at 20:43, Dana Tong  wrote:


Hi all,

Not sure if anyone is still using on-prem CUCM much these days but am trying to 
update some 8811 background and am getting a “Wallpaper cannot be set. 
Incompatible image size”.

My image and thumbnail are:
800 x 480 x 24 bit
Thumbnails are: 159 x 109 x 24 which match the existing backgrounds.

Although there is some documentation that says they need to be 139 x 109 for 
the Thumbnails.

I would have thought the original included images would be okay? But they are 
not displaying either.

Anyone done this and know for certain what they should be?

Cheers
Dana



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Re: [cisco-voip] TAC Support for Unsupported Configurations

[Ryan Huff]

Wow, this one hits home..

I’ve found (and this is just my experience, which is entirely subjective) that, 
if in a dire moment of need, on a Saturday morning at 12:12 AM EST, if you show 
a humanness to yourself, TAC will show moments of kindness ... supporting the 
statement that yes indeed, it really is about whether the TAC agent wants to 
help or not.

Personally, I’ve never experienced a TAC agent that hit me with the, “call me 
back when you fix XYZ” scenario. I’ve heard it discussed in the community 
before but have always considered it as lore.

I suppose it’s all about the individual scenario; if I open a case on a c220 
M3S/Xeon E5-26xx with 14 physical cores allocated ... I’d expect to have the 
phone hung up on me, but perhaps not (speaking from real life experience). Some 
times you find some truly exceptional TAC agents..

-Ryan

On Sep 10, 2019, at 17:30, Anthony Holloway  
wrote:


Why do I get the feeling that you're speaking from experience, with that crazy 
setup?

That's kind of been my experience too, just to confirm.  I have had mixed 
results with TAC in the past, on how far they were willing to go for me.  It's 
definitely a human experience.

On Tue, Sep 10, 2019 at 3:35 PM Charles Goldsmith 
mailto:w...@woka.us>> wrote:
I think he's conflicted :)

I've always been told that it's best effort, and as you stated, probably how 
much effort the TAC engineer wants to put into it.  It is likely handled on a 
case by case basis as well as their queue/work load.  If you come in there 
trying to run CUCM 8.0 on VMware fusion on an AMD based MACOS that is connected 
to a CUBE running in GNS3, they may not be in a hurry to help you :)

On Tue, Sep 10, 2019 at 2:05 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
Or maybe it's just up to the assigned TAC Engineer to decide whether they want 
to work the case or not?



Source: 
https://community.cisco.com/t5/contact-center/agent-login-failure-uccx-finesse-ver-10-6-1/m-p/2870582/highlight/true#M92340



Source: 
https://community.cisco.com/t5/contact-center/uccx-11-5-1-finesse-jabber-em-7841/m-p/3091142/highlight/true#M100912


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Re: [cisco-voip] CIMC MIC self-signed cert and new browsers

[Ryan Huff]

May want to look at the browser’s address bar and see if there is an applet / 
plugin looking for permission to run.

- R

On Aug 29, 2019, at 01:56, Dana Tong  wrote:


Hi all,

New server install. Trying to login to the CIMC to configure. Chrome, Firefox, 
IE all have a hissy fit. I did manage to get in once on one of the servers 
using firefox and dropping the max TLS version to 3. But after setting the 
password and change IP and hostname, it won’t let me back in.

Chrome just keeps going round and round if I choose to proceed.

Any tricks on getting browsers to work with this?

Cheers




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Re: [cisco-voip] CUCM SU release cycle

[Ryan Huff]

Let’s not get ahead of ourselves there ;). Just like war, usually (but not 
always) the people who want 100% cloud calling or think it’s a great idea are 
the people who’ve never experienced it.. lol

Sent from my iPhone

On Aug 15, 2019, at 13:22, Lelio Fulgenzi  wrote:


You forgot how everyone will be migrating to Webex Calling before then. And 
your upgrade cycle will be out of control. Just like how Webex Teams has that 
green restart symbol every two weeks.

---
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>

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 | @UofGCCS on Instagram, Twitter and Facebook



From: Anthony Holloway 
Sent: Thursday, August 15, 2019 12:05 PM
To: Lelio Fulgenzi 
Cc: Charles Goldsmith ; cisco-voip voyp list 

Subject: Re: [cisco-voip] CUCM SU release cycle

So if I do the math...

No more minor versions
...punches some keys...

And 2 month SU cycles
...punches more keys

With an upper limit of 3 SUs
...punches even more keys...

That's a new major version every 6 months!

That means we'll see
..key punching intensifies...

CUCM 69 by mid-2046.  Just in time for me to retire!

On Thu, Aug 15, 2019 at 10:57 AM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
Pretty sure I remember them saying there likely wouldn’t be that many SU’s 
either, three at most?

*sigh*

---
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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>

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 | @UofGCCS on Instagram, Twitter and Facebook



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Thursday, August 15, 2019 11:50 AM
To: Charles Goldsmith mailto:w...@woka.us>>
Cc: cisco-voip voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CUCM SU release cycle

Make it SU5 in memory of the .5 releases.

On Thu, Aug 15, 2019 at 9:18 AM Charles Goldsmith 
mailto:w...@woka.us>> wrote:
I didn't see an announcement, was just told about the change, Cisco doesn't 
like us waiting for the .5 release to push out to customers.  We all know that 
the .0 releases have historically been more challenging.

So now, I plan to wait until at least su2 before upgrading :)


On Thu, Aug 15, 2019 at 9:15 AM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
Why not just all Major versions all the time?  Google Chrome is on version 76.

But seriously though, anyone got a reference to this announcement?  I didn't 
see it in the cisco live preso linked earlier.

If not, what's the reported reason for dropping minor release numbers?

On Thu, Aug 15, 2019 at 5:40 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
After 12.5, no more “.5” releases, it’ll just be major versions (and the SUs in 
between). After 12.5 we skip 13 and go right to 14 (then presumably, 15 after 
that).
Sent from my iPhone

On Aug 15, 2019, at 02:05, Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:

What's going on with .5 releases?  I don't think I heard about that.

On Wed, Aug 14, 2019 at 11:16 PM Charles Goldsmith 
mailto:w...@woka.us>> wrote:
Yeah, I think with the move away from the .5 releases, we'll be getting more 
SU's and less major releases.

On Wed, Aug 14, 2019 at 10:58 PM Ki Wi 
mailto:kiwi.vo...@gmail.com>> wrote:
Hi Group,
in the past , the SU release is every 6 months (usually longer than that, 
approximately twice a year maximum) but now Cisco is changing to every 2 months?

Reference : Page 20 of the link
https://www.ciscolive.com/c/dam/r/ciscolive/us/docs/2019/pdf/PSOCOL-1000.pdf<https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.ciscolive.com%2Fc%2Fdam%2Fr%2Fciscolive%2Fus%2Fdocs%2F2019%2Fpdf%2FPSOCOL-1000.pdf=02%7C01%7C%7C22113f37d40644de57c608d721a523d2%7C84df9e7fe9f640afb435%7C1%7C0%7C637014865456439678=yeh%2BPcDLDmSl8oD0w3mTIZBpZXMDrDLZWh5YVl9aV4U%3D=0>


--
Regards,
Ki Wi
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Re: [cisco-voip] CUCM SU release cycle

[Ryan Huff]

Anthony,

Slide 20 of the deck from this session 
(https://www.ciscolive.com/global/on-demand-library.html?search=cucm%20version=ciscoliveus2019#/session/1540336593046001DbNj)
 makes a loose reference to it in the timeline graphic.

There is a deck out there that gives a deeper explanation (and I'll send it to 
you if I have some time to look for it); but it was essentially because the BU 
felt features/enhancements were delayed to the field because everyone had a 
fear of " .0 " (not a wholly unfounded fear, IMO) and also some marketing jazz 
about being more inline with software revisionist trends .. etc.

-Ryan


From: Lelio Fulgenzi 
Sent: Thursday, August 15, 2019 10:31 AM
To: Matthew Loraditch ; Anthony Holloway 
; Ryan Huff 
Cc: Charles Goldsmith ; cisco-voip voyp list 

Subject: RE: [cisco-voip] CUCM SU release cycle




Yeah, this was the jist. No one liked “dot oh” releases.



It just means people are now going to make up their own upgrade cycle in their 
head based on what we hear. Waiting until SU2 maybe? Who knows.



---

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Computing and Communications Services | University of Guelph

Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1

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[University of Guelph Cornerstone with Improve Life tagline]



From: cisco-voip  On Behalf Of Matthew 
Loraditch
Sent: Thursday, August 15, 2019 10:20 AM
To: Anthony Holloway ; Ryan Huff 

Cc: Charles Goldsmith ; cisco-voip voyp list 

Subject: Re: [cisco-voip] CUCM SU release cycle



I heard this in a preso. Customers were not moving to .0 releases because they 
were perceived to be bad and waiting for .5 or .1 type releases. The defect 
rates though are not much if any different these days and they won’t want to be 
making releases that nobody uses.





Matthew Loraditch​

Sr. Network Engineer

p: 443.541.1518

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From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Anthony Holloway
Sent: Thursday, August 15, 2019 10:16 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: Charles Goldsmith mailto:w...@woka.us>>; cisco-voip voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CUCM SU release cycle



Why not just all Major versions all the time?  Google Chrome is on version 76.



But seriously though, anyone got a reference to this announcement?  I didn't 
see it in the cisco live preso linked earlier.



If not, what's the reported reason for dropping minor release numbers?



On Thu, Aug 15, 2019 at 5:40 AM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

After 12.5, no more “.5” releases, it’ll just be major versions (and the SUs in 
between). After 12.5 we skip 13 and go right to 14 (then presumably, 15 after 
that).

Sent from my iPhone



On Aug 15, 2019, at 02:05, Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:



What's going on with .5 releases?  I don't think I heard about that.



On Wed, Aug 14, 2019 at 11:16 PM Charles Goldsmith 
mailto:w...@woka.us>

Re: [cisco-voip] CUCM SU release cycle

[Ryan Huff]

After 12.5, no more “.5” releases, it’ll just be major versions (and the SUs in 
between). After 12.5 we skip 13 and go right to 14 (then presumably, 15 after 
that).

Sent from my iPhone

On Aug 15, 2019, at 02:05, Anthony Holloway  
wrote:


What's going on with .5 releases?  I don't think I heard about that.

On Wed, Aug 14, 2019 at 11:16 PM Charles Goldsmith 
mailto:w...@woka.us>> wrote:
Yeah, I think with the move away from the .5 releases, we'll be getting more 
SU's and less major releases.

On Wed, Aug 14, 2019 at 10:58 PM Ki Wi 
mailto:kiwi.vo...@gmail.com>> wrote:
Hi Group,
in the past , the SU release is every 6 months (usually longer than that, 
approximately twice a year maximum) but now Cisco is changing to every 2 months?

Reference : Page 20 of the link
https://www.ciscolive.com/c/dam/r/ciscolive/us/docs/2019/pdf/PSOCOL-1000.pdf


--
Regards,
Ki Wi
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Re: [cisco-voip] Unity DRS components

[Ryan Huff]

In Unity, the Primary and HA DRS components are separated and operate 
independently. Though it’s generally not necessary to backup the HA node.

- In a DR recovery scenario of both Unity Servers, you can restore the Primary 
and install a new HA.

- In a DR recovery scenario of the HA only, just install a new HA (because if 
you restored a HA DRS, it would take even longer because you have to reset 
cluster replication).

- In a DR scenario of the Primary only (and the HA is still running); install a 
new Primary and renegotiate the Primary to the HA. This is far quicker than 
reinstalling a new Primary and then a new HA.

Sent from my iPhone

On Aug 13, 2019, at 09:14, Lelio Fulgenzi  wrote:


Is this one of those “back up the publisher only, restore the publisher, 
reinstall the subscriber and let the publisher re-image the subscriber” sort of 
things?

We’ve seen that in this case, platform information is never restored and who 
the heck likes to remember that sort of stuff?

---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: cisco-voip  On Behalf Of Anthony 
Holloway
Sent: Monday, August 12, 2019 11:24 AM
To: Charles Goldsmith 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Unity DRS components




Source: Install, Upgrade, and Maintenance Guide for Cisco Unity Connection 
Release 
11.x


On Mon, Aug 12, 2019 at 9:22 AM Charles Goldsmith 
mailto:w...@woka.us>> wrote:
Unity Connection has always been the oddball, the pub only backs it self up, 
you have to schedule the sub to do it's own backup.

On Mon, Aug 12, 2019 at 8:30 AM Myron Young 
mailto:mdavid_yo...@hotmail.com>> wrote:
Morning,

Is it just me or shouldn’t both the Unity Pub and Sub servers be shown as 
available “components registered with Disaster Recovery System” when running 
either a manual or scheduled backup?

 I see all nodes in the cluster for UCM but not seeing it on the Unity cluster; 
and confirmed the DRS local and master services are running on both servers.
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Re: [cisco-voip] Do you need TMS to get OBTP

[Ryan Huff]

To do OBTP on prem; TMS/TMS-XE. If you want to go the cloud registration route 
for the endpoint, Expressway-C/management connector and Hybrid calendar 
connector (on-prem Exchange) or cloud (Webex) calendar (O365 or Google)

Sent from my iPhone

On Aug 7, 2019, at 12:39, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

OK, do we need TMS XE as well, or can the Calendar Connector handle it? (they 
are O365)


Thanks!


On Wed, Aug 7, 2019 at 11:36 AM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
You need TMS. Something has to publish the schedule to the endpoints, on prem 
that’s TMS. In the cloud that’s Webex.


Matthew Loraditch​
Sr. Network Engineer
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 Jonathan Charles
Sent: Wednesday, August 7, 2019 12:34 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Do you need TMS to get OBTP

Have a customer with CUCM 10.5 and they want to register video endpoints 
locally to CCM.

They do not have TMS, but they want OBTP... can we just install an Expressway 
Connector and use the Webex Hybrid cloud to use @webex and @meet to get OBTP 
without TMS?



Jonathan
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Re: [cisco-voip] Uplinx Report Tool?

[Ryan Huff]

 and, you can maximize the application window.. *key selling point* lol. 
Seriously though, both are really good products.

Sent from my iPhone

On Aug 3, 2019, at 06:51, Stephen Welsh 
mailto:stephen.we...@unifiedfx.com>> wrote:

Hi Nick, hope you are well.

Yes I do follow this email group (among many other communities) for an 
excellent insight to what’s happening in the world of Cisco UC, however modesty 
is not my strong point ;)

I’m partial to a bit of guerrilla marketing, sometimes I go a bit far. Once I 
did get banned from the Cisco Community site, so I apologise if this self 
promotion offends.

[Guerrilla Marketing On]

PhoneView 7.0 (http://download.unifiedfx.com/PhoneView) introduces more unique 
features:
* Virtual Cisco Endpoints (register multiple Jabber devices on a single PC)
* Soft phone and MRA phone support (see call activity & control calls)
* UCCX Integration (see and set real-time agent state)

PhoneView 7.1 (beta due Sept)
* AutomationFX/PhoneFX policy feature
* AutomationFX Community Edition (Free):
Exposes Cisco UCM CTI, AXL & RISPort via REST API
Automate, Develop and Test easily with Cisco UC
Create custom Cisco UC applications
Python SDK 
(https://github.com/unifiedfx/automationfx-python
 )

[Guerrilla Marketing Off]

Kind Regards

Stephen Welsh

Sent from my iPad

On 3 Aug 2019, at 07:35, Nick Britt 
mailto:nickolasjbr...@gmail.com>> wrote:


uplinx is brilliant, I am sure Stephen Welsh is skulking around here practicing 
his modesty. I will continue to push customers to buy it.

On Sat, Aug 3, 2019 at 12:46 AM Fares Alsaafani 
mailto:fares@gmail.com>> wrote:
Hi Matthew, I have used remote control for Cisco phone software was great saved 
my day on remote site upgrade cutover.

On Fri, Aug 2, 2019 at 6:16 AM Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:
https://www.uplinx.com/reporttool-usd/

Anyone heard of/used these folks?



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








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Best Regards

FARES ALSAAFANI
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Re: [cisco-voip] Uplinx Report Tool?

[Ryan Huff]

I use them all the time. As long as you invest the time to creating good CSS 
and templates, it’s great. The default templates are generic and just create 
“data dumps” in Word format... IMO.

However, and this is my one diva moment here, the GUI wind will not maximize, 
lol... they jail it for a smaller screen resolution... it’s fine and completely 
readable... but it just p*sses me off that I can’t maximize the dang window ... 
lol

Sent from my iPhone

On Aug 2, 2019, at 09:16, Matthew Loraditch 
mailto:mloradi...@heliontechnologies.com>> 
wrote:

https://www.uplinx.com/reporttool-usd/

Anyone heard of/used these folks?



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




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Re: [cisco-voip] CPU Reservations

[Ryan Huff]

People view dial tone as if it were in the US Bill of Rights. Let it not be 
there when expected and see what happens. It’s a fascinating social experience.

Sent from my iPhone

On Jul 10, 2019, at 10:47, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

My answer back to them is this most of the time:

Me: “Please pick up your phone.”
Them: (they pick up their phone)
Me: “What did you hear?”
Them: “Dial Tone”
Me: _That’s_ why we need resource reservation.

And then I go on to ask them how they’d feel if that dial tone or other 
features were delayed by a ½ second? 1 second? 2 seconds?

They’re still not happy, but they begin to get the picture.

Of course, my example may not be the best example, but it gets the message 
across.




---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: Palmer, Brian mailto:brian.pal...@bcbsfl.com>>
Sent: Wednesday, July 10, 2019 9:35 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; Wes Sisk 
(wsisk) mailto:ws...@cisco.com>>
Cc: cisco-voip@puck.nether.net; Pawlowski, 
Adam mailto:aj...@buffalo.edu>>
Subject: RE: [cisco-voip] CPU Reservations

The VM team here when I told them we had to have resource reservation and no 
oversubscription always complain.  “Your servers don’t consume anywhere near 
that capacity I can show you the performance stats”  “Every vendor asks for 
resource reservation that comes in”

That is what I always hear from them.

Brian Palmer| VoiceOps | DC6 3 355
904-905-8263  |  Internal Ext: 58263

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Monday, July 8, 2019 10:42 PM
To: Wes Sisk (wsisk) mailto:ws...@cisco.com>>
Cc: cisco-voip@puck.nether.net; Pawlowski, 
Adam mailto:aj...@buffalo.edu>>
Subject: Re: [cisco-voip] CPU Reservations



“ccm.bin at its core is a distributed real time state machine.”

^^^ this ^^^

This is what I try to tell people when they tell me that they can run 50 
machines on the same boxes that I run 10.

And they _still_ don’t get it.


-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
2W1
519-824-4120 Ext. 56354 | 
le...@uoguelph.ca

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 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

On Jul 8, 2019, at 7:38 PM, Wes Sisk (wsisk) via cisco-voip 
mailto:cisco-voip@puck.nether.net>> wrote:
ccm.bin at its core is a distributed real time state machine.

We comply with applicable Federal civil rights laws and do not discriminate.

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the 

Re: [cisco-voip] CPU Reservations

[Ryan Huff]

Adam,

You'd probably be less likely to have an "issue" if the host's aggregate 
compute resources are at least 30%-35% below subscribed capacity (but no 
guarantees). Real time traffic on the software such as mtp, moh,.. etc (I think 
UCOS even has some files that internally communicate via real time or near real 
time) can be a diva at times. There are two things that come to mind that I 
have always seen give issue to real time traffic without fail;


  *   latency and jitter, albeit spatial or mechanical such as distance or 
clock cycles
  *   unreliable or latent synchronization (I'd humbly suggest this is anything 
over 5 hops from a cesium clock, but Cisco says 3 so we'll go with that)

I've found that if you violate either of these, you can be in for a wild ride. 
The symptoms are often obscure, inconsistent and seemingly unrelated to any 
other known issues; a small delay in voice that randomly pops up here or there 
for one or two phones, a presence HA pair that randomly fails over for no 
apparent reason, some phones not showing call history, SIP trunks going out of 
service.. etc. These issues can be very difficult to troubleshoot because there 
won't "appear" to be anything wrong.

Its "hit or miss" at best with the CPU reservations (on a host that is not 
already over subscribed) I'd say; if the UC VM has to wait (delay) on cycle 
time, even for a fraction of a second, it may or may not cause you an issue... 
just depends on what the server was trying to do at the time, and if it 
involved real time traffic. If you've got UC VMs in the mix with HA / 
clustering, then the VMs will be even less tolerant of asking for cycle time 
rather than it just being available.

The safest path is to guarantee (reserve) the require resources to the UC VM, 
even though it may not ever (or nearly never) use the full capacity (because 
having cycle time readily available and having to ask the scheduler for cycle 
time that is available is not the same thing).

Think of it like insurance, you're not paying for it because you don't need it 
(actual waste), you're paying because of that one time you do need it and don't 
know it.

Thanks,

Ryan

From: cisco-voip  on behalf of Pawlowski, 
Adam 
Sent: Monday, July 8, 2019 11:57 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] CPU Reservations


Hi all,



It’s been a bit since I’ve asked this question, if I have here before.



Do we all run our UC appliances in VMWare with the full CPU MHz and core 
reservations prescribed by Cisco, in production? Or, if you have information on 
hand regarding the actual resource usage, have any of you taken on resizing the 
VM reservations?



The various documents are very much so clear that oversubscription isn’t 
supported, but, it also talks about vCPU to cores which I’m told doesn’t really 
play out in VMWare as it’s a MHz reservation that can be scheduled in to 
available hardware.



There are various statements peppered in about running your own VM environment 
with best practices – but also the 1:1 pcore:vcore comments.



Is anyone turning these knobs? Has anyone stepped over that pcore:vcore line 
when it appears there are enough resources?



I’m looking for thoughts or unforeseen consequences that we can use to back 
somewhat of the case as to why we need to continue to fund hardware at scale 
which is largely idle.



Adam




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Re: [cisco-voip] How difficult is hybrid call (connector) setup ?

[Ryan Huff]

It all comes down to what you are using your Expressways for. In a Webex Hybrid 
call connect scenario you could in theory, have calls with source uri’s using 
your domain, which in a pure B2B scenario is usually something you’d deny with 
a call policy.

As I understand it, the dedicated Expressway recommendation (outside of pure 
capacity reasons) is to make it easier to write call policies that don’t 
interfere with other use cases since Expressway doesn’t really have a 
partitioning mechanism (outside of what you can do with search rules).

You really have to go to the Ninja master level with your regular expressions 
in your search rules and call polices to get multiple use cases setup and using 
call policies to reduce toll fraud... and have everything work.

... and this is where Cisco should, in my opinion, step up to the plate a 
little. The best answer really shouldn’t be, “just deploy and use another 
Expressway”.

Yes, that is easier than refining CPL and search rules, but many customers run 
tight compute/storage budgets (Ex. be6k) and cant always spin up more 
Expressways.

-R

On Jul 2, 2019, at 22:06, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Ok. Thanks.

I think we’re on a low version of expressway. I’ll have to confirm.

My memory is failing, and I can’t find it in my notes, but in one session they 
talked about using a separate pair of C’s and E’s or have a high risk of toll 
fraud.

I believe it was in my Sunday techtorial. I’ll have to reach out to them.

-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
2W1
519-824-4120 Ext. 56354 | 
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On Jul 2, 2019, at 9:49 PM, Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

It’s not that bad (maybe 1 - 2 hours to get a functional test going).

The most unexpected thing I think you may run into is the use/need for MTLS 
(TCP:5062) between the Edge and Control Hub. Also, your Expressway version 
should be 12.5.x (I think 8.11.4 may still work but you’ll get an alarm telling 
you to upgrade if it does work).

Outside of that, it’s a splash of nerd knob turning in the control hub, some 
search rule / traversal & dns zone magic in Expressway C/E and setting up the 
management/call connector in Expressway, you can even re-use a MRA (unified 
communications) traversal client/server (or create a dedicated traversal if so 
inclined).

Here is the guide you’d want to follow for it and it’s pretty complete and well 
written: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html<https://nam04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2FcloudCollaboration%2Fspark%2Fhybridservices%2Fcallservices%2Fcmgt_b_ciscospark-hybrid-call-service-config-guide.html=02%7C01%7C%7Cc007f1e5951c4fe8148208d6ff5afe84%7C84df9e7fe9f640afb435%7C1%7C0%7C636977163613993708=Ryhmmh%2F1SxqFB8uOU6S%2FTWioJ5IUVgDkjxCEUk0SdvE%3D=0>

Thanks,

Ryan

On Jul 2, 2019, at 21:30, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


On a scale from 1 to 10, how difficult and/or time consuming is it to setup 
call hybrid connector with WebEx?

I’m just about convinced that I’d rather register our room devices and SX20s to 
the cloud. But people would like to be able to dial extensions and PSTN 
numbers. Being able to use our audio bridges would also be of benefit (ya, I 
understand the irony of that).


-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<https://nam04.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs=02%7C01%7C%7Cc007f1e5951c4fe8148208d6ff5afe84%7C84df9e7fe9f640afb435%7C1%7C0%7C636977163613993708=OIht9e0rIYKhqiZKeK1AY7AmCK6CCCdNKhMyDVC82us%3D=0>
 | @UofGCCS on Instagram, Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]
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Re: [cisco-voip] How difficult is hybrid call (connector) setup ?

[Ryan Huff]

It’s not that bad (maybe 1 - 2 hours to get a functional test going).

The most unexpected thing I think you may run into is the use/need for MTLS 
(TCP:5062) between the Edge and Control Hub. Also, your Expressway version 
should be 12.5.x (I think 8.11.4 may still work but you’ll get an alarm telling 
you to upgrade if it does work).

Outside of that, it’s a splash of nerd knob turning in the control hub, some 
search rule / traversal & dns zone magic in Expressway C/E and setting up the 
management/call connector in Expressway, you can even re-use a MRA (unified 
communications) traversal client/server (or create a dedicated traversal if so 
inclined).

Here is the guide you’d want to follow for it and it’s pretty complete and well 
written: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html

Thanks,

Ryan

On Jul 2, 2019, at 21:30, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


On a scale from 1 to 10, how difficult and/or time consuming is it to setup 
call hybrid connector with WebEx?

I’m just about convinced that I’d rather register our room devices and SX20s to 
the cloud. But people would like to be able to dial extensions and PSTN 
numbers. Being able to use our audio bridges would also be of benefit (ya, I 
understand the irony of that).


-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

[University of Guelph Cornerstone with Improve Life tagline]
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Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

[Ryan Huff]

Actually, I prefer to disable IMP HA before installing in the inactive PT. This 
way, if you have to flip back to the other partition for some reason, IMP is 
already in an “HA disabled” state, which tends to make IMP recover a little 
better in my experience. Then just enable HA once everything is stable.

Thanks,

Ryan

On May 28, 2019, at 14:03, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Personally, I hate the fact that IM can’t be restarted without first 
disabling HA. So basically, if you have an unattended restart, you have to go 
in and make configuration changes.

Yuck.

---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: NateCCIE mailto:natec...@gmail.com>>
Sent: Tuesday, May 28, 2019 12:46 PM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>; 'Bill Talley' 
mailto:btal...@gmail.com>>
Cc: 'voyp list, cisco-voip' 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] IM Upgrade Steps during CUCM upgrade

Yeah, I HATE this bug.  Why in the world can’t the docwiki or what ever it’s 
called be updated quicker than a bug be filed/made universally known, and who 
came up with these recommendations TAC or the BU.

But I have seem IMP just not start services, and adding resources magic fix it.

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Lelio Fulgenzi
Sent: Tuesday, May 28, 2019 9:46 AM
To: Bill Talley mailto:btal...@gmail.com>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade


In the words of the immortal Chris Farley….

holy schnikes

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

They want 4 vCPU if possible? They think these things grow on trees? 

Right now, our two IMP servers are at 2 vCPU and 4GB of RAM. (5000 user OVA).

I’ll have to see about coordinating this change as well. We don’t have a lot of 
capacity/activity on these servers, so I think we should be OK for now.

Funny thing – Bug updated May 20, 2019, but virtualization docs still show old 
OVA information.

---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: Bill Talley mailto:btal...@gmail.com>>
Sent: Tuesday, May 28, 2019 11:08 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

That’s the process I typically follow without issues.

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

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

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

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

Conditions:
Performance Issues

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

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




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

On May 28, 2019, at 8:47 AM, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

I'm reading the upgrade guide, specifically, the time 

Re: [cisco-voip] IM Upgrade Steps during CUCM upgrade

[Ryan Huff]

This is what I do for an in-place upgrade with no other changes (hostname, IP 
address.. etc).

I do the cucm pub first (obviously), then imp pub (which is more like a cucm 
sub for upgrade purposes), then all the cucm and imp subs; everything into the 
inactive pt.

I do the pubs individually, then I’ll do a couple subs at a time .. etc.

Next I Switch version on the pubs; cucm, imp then the subs.

On the switch version, I wait till one node is fully up (tomcat started) before 
switching another.

May not be as efficient as it could be, but has kept me out of trouble thus 
far; plan your dive, dive your plan.

On May 28, 2019, at 09:48, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


I’m reading the upgrade guide, specifically, the time sequencing:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/upgrade/11_5_1/cucm_b_upgrade-guide-cucm-115/cucm_b_upgrade-guide-cucm-115_chapter_010010.html

and it mentions upgrading the IM publisher (to inactive partition) at the 
same time as upgrading the subscribers. Then doing a switch version on the IMP 
pub at the same time as the CUCM subs.

Anyone do this parallel type upgrade before?

Sure would save a lot of time.

Lelio


---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



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

[Ryan Huff]

I do

Sent from my iPhone

On May 23, 2019, at 19:09, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Can anyone confirm they have a control hub org with multiple control hub / 
common identity managed Webex sites enabled?

I’m not talking linked sites. These are WebEx sites that were created from 
control hub as control hub managed from the get go.

There is some scuttlebutt that says this is not possible and it has me 
concerned.

Lelio


-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 
2W1
519-824-4120 Ext. 56354 | 
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] Cisco Live CLUS anyone??

[Ryan Huff]

rh...@byteworks.com<mailto:rh...@byteworks.com>, if any hard chargers are in 
the process of setting up a space...

Sent from my iPhone

On May 23, 2019, at 14:13, Peter Slow 
mailto:peter.s...@gmail.com>> wrote:

I want some Cheetos! I’ll be there!

On Thu, May 23, 2019 at 11:12 Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
I’m out here eating Cheetos with you. Let’s do a team’s space...

Sent from my iPhone

On May 23, 2019, at 14:01, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

We're a little over two weeks away from CLUS.  Ryan were you a fan of the Webex 
Teams space for us?  Or is this one of those Rounders moments, where everyone 
else is in the space right now, and I'm on the outside wondering if it even 
exists?  :(

On Tue, Apr 30, 2019 at 11:15 AM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
By "green body paint," I hope you meant Hulk, because I already called dibs on 
cosplaying as Gamora.

On Tue, Apr 30, 2019 at 10:53 AM Jason Aarons (Americas) 
mailto:jason.aar...@dimensiondata.com>> wrote:
So which Avenger character are you going to be?  We fully expect you to dress 
up in character.  How do you look wearing green body paint?


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Stephen Welsh
Sent: Tuesday, April 30, 2019 2:35 AM
To: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Cisco Live CLUS anyone??


I’ll be there too.

The WebEx room was fun and helpful, another would be great.

We will have a stand again

Stephen Welsh
Founder & CTO
UnifiedFX
Sent from my iPhone

On 29 Apr 2019, at 20:39, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
I'll be there!

Last year, I started a webex teams room for those of us who were in attendance 
(and wanted to be in the room).  A few of us used it to coordinate meetups.  
There was also some good conversation in the room during the week as well.  
Mostly, heads-ups on events, humor and commentary on what was being learned 
that week.

I might even been in the Engineering Deathmatch again, as a defending champ.  I 
hope to god they don't put me up against you Ryan.  Fingers crossed.

On Mon, Apr 29, 2019 at 2:19 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Just curious to know whom (if anyone) from the list will be at CLUS this year 
in San Diego? I will be and would love to meetup with those from the list whom 
I’ve only exchanged emails with thus far.
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Re: [cisco-voip] Cisco Live CLUS anyone??

[Ryan Huff]

I’m out here eating Cheetos with you. Let’s do a team’s space...

Sent from my iPhone

On May 23, 2019, at 14:01, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

We're a little over two weeks away from CLUS.  Ryan were you a fan of the Webex 
Teams space for us?  Or is this one of those Rounders moments, where everyone 
else is in the space right now, and I'm on the outside wondering if it even 
exists?  :(

On Tue, Apr 30, 2019 at 11:15 AM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
By "green body paint," I hope you meant Hulk, because I already called dibs on 
cosplaying as Gamora.

On Tue, Apr 30, 2019 at 10:53 AM Jason Aarons (Americas) 
mailto:jason.aar...@dimensiondata.com>> wrote:
So which Avenger character are you going to be?  We fully expect you to dress 
up in character.  How do you look wearing green body paint?


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Stephen Welsh
Sent: Tuesday, April 30, 2019 2:35 AM
To: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Cisco Live CLUS anyone??


I’ll be there too.

The WebEx room was fun and helpful, another would be great.

We will have a stand again

Stephen Welsh
Founder & CTO
UnifiedFX
Sent from my iPhone

On 29 Apr 2019, at 20:39, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:
I'll be there!

Last year, I started a webex teams room for those of us who were in attendance 
(and wanted to be in the room).  A few of us used it to coordinate meetups.  
There was also some good conversation in the room during the week as well.  
Mostly, heads-ups on events, humor and commentary on what was being learned 
that week.

I might even been in the Engineering Deathmatch again, as a defending champ.  I 
hope to god they don't put me up against you Ryan.  Fingers crossed.

On Mon, Apr 29, 2019 at 2:19 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Just curious to know whom (if anyone) from the list will be at CLUS this year 
in San Diego? I will be and would love to meetup with those from the list whom 
I’ve only exchanged emails with thus far.
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Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via local PSTN

[Ryan Huff]

Apologies, didn’t mean to cause the need to meditate 

Sent from my iPhone

On May 16, 2019, at 11:02, Anthony Holloway 
mailto:avholloway+cisco-v...@gmail.com>> wrote:

"... this stuff is soo fluid right now, it’s a contradictory mess..."

This bothers me to an unhealthy degree.  I really need to meditate or something.

On Wed, May 15, 2019 at 9:39 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
Yup, this stuff is soo fluid right now, it’s a contradictory mess . The 
on-prem registration, IMHO, is rrr from prime time (as you even noted, 
“preview mode”). I honestly think it’s a bit further out than what the 
marketing department would have us believe too ;).

Hopefully though, one day, we’ll all have this unified unicorn they’re 
promising... Jeams?... Jams? ... who knows 

Tops to you mate!

On May 15, 2019, at 22:31, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:

Hi Ryan,

Yeah sorry, I realise you guys were talking WebEx Hybrid Call (formally Spark 
Hybrid).

Just pointing out there is some new stuff on horizon. Calling from WebEx Teams 
via CUCM feature.
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_chapter_011.html<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2FcloudCollaboration%2Fwbxt%2Fucmcalling%2Funified-cm-wbx-teams-deployment-guide%2Funified-cm-wbx-teams-deployment-guide_chapter_011.html=02%7C01%7C%7C0c06fdfb1500415242ba08d6da0f89d8%7C84df9e7fe9f640afb435%7C1%7C0%7C636936157605890504=70z9uOBPVxB6F%2BpIk9YGMHr1OxsfRoc1%2BJGi61SJ8CU%3D=0>

It looks like this will replace the calling from Teams part of the old Hybrid 
Call (although not on mobile clients yet) – in fact it seems you have to remove 
their old Hybrid config to make them work.

Looks like you’d still need to retain Hybrid Call for cloud registered devices.

There was one specific annoyance with the Hybrid Call from the apps, and I 
can’t find it in my Teams search ☹

Cheers,

Tim


From: Ryan Huff mailto:ryanh...@outlook.com>>
Date: Thursday, 16 May 2019 at 12:14 pm
To: Tim Smith mailto:tim.sm...@enject.com.au>>
Cc: Jonathan Charles mailto:jonv...@gmail.com>>, 
"cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via 
local PSTN

This is Webex Hybrid Calling (which was formerly Spark Hybrid calling). Whether 
you configure for cloud registered codec devices, or Webex Teams clients, both 
use cases use the same configuration path / scenario to enable PSTN call via 
CUCM.
Sent from my iPhone

On May 15, 2019, at 22:08, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:
Hey guys,

I think this one has changed a little.
We did “Spark hybrid calling” for one customer with the Spark RD devices in CUCM
Honestly, the experience was a little confusing.

I think the new direction is going to be the WebEx Calling via CUCM (it’s in 
preview mode still)
https://help.webex.com/en-us/n15ylys/Explore-Calling-in-Cisco-Webex-Teams-Unified-CM<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fhelp.webex.com%2Fen-us%2Fn15ylys%2FExplore-Calling-in-Cisco-Webex-Teams-Unified-CM=02%7C01%7C%7C0c06fdfb1500415242ba08d6da0f89d8%7C84df9e7fe9f640afb435%7C1%7C0%7C636936157605920531=QiAUsNC6BtKjjvJQWQEYxQ3FT7qc8oLU6UredrBrSnU%3D=0>

It’s not parity with the Hybrid Calling yet. (i.e. I think it’s only desktop)
Either way, I’d check out all the details first.

If you are not already on there, make sure you are on the Fabian bot in WebEx 
teams. (Not sure if it’s partners only)
These hybrid features are really starting to rock and roll now.

Cheers,

Tim

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Date: Thursday, 16 May 2019 at 9:30 am
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: "cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via 
local PSTN

Thanks!... looks like I have some more reading to do... so how does it prevent 
anyone from sending a pstn number to my expressway? How does it authenticate 
the Webex devices to pass calls to CUCM for?

Customer has enterprise licensing, so they should be able to do whatever they 
want...


Jonathan



On Wed, May 15, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
You’ll need a specific Webex DNS zone and the traversal trunk really just needs 
to support pre-loaded route headers and SIP parameter preservation (those are 
the most significant differences over the traversal / neighbor zone you might 
h

Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via local PSTN

[Ryan Huff]

Yup, this stuff is soo fluid right now, it’s a contradictory mess . The 
on-prem registration, IMHO, is rrr from prime time (as you even noted, 
“preview mode”). I honestly think it’s a bit further out than what the 
marketing department would have us believe too ;).

Hopefully though, one day, we’ll all have this unified unicorn they’re 
promising... Jeams?... Jams? ... who knows 

Tops to you mate!

On May 15, 2019, at 22:31, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:

Hi Ryan,

Yeah sorry, I realise you guys were talking WebEx Hybrid Call (formally Spark 
Hybrid).

Just pointing out there is some new stuff on horizon. Calling from WebEx Teams 
via CUCM feature.
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_chapter_011.html<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2FcloudCollaboration%2Fwbxt%2Fucmcalling%2Funified-cm-wbx-teams-deployment-guide%2Funified-cm-wbx-teams-deployment-guide_chapter_011.html=02%7C01%7C%7C9379fa8f35284f2e6bfc08d6d9a6a4be%7C84df9e7fe9f640afb435%7C1%7C0%7C636935707083032550=b4UfPgVLInjyqap7dxRoF7T3PXcBna%2FOW0%2F7GY7maxA%3D=0>

It looks like this will replace the calling from Teams part of the old Hybrid 
Call (although not on mobile clients yet) – in fact it seems you have to remove 
their old Hybrid config to make them work.

Looks like you’d still need to retain Hybrid Call for cloud registered devices.

There was one specific annoyance with the Hybrid Call from the apps, and I 
can’t find it in my Teams search ☹

Cheers,

Tim


From: Ryan Huff mailto:ryanh...@outlook.com>>
Date: Thursday, 16 May 2019 at 12:14 pm
To: Tim Smith mailto:tim.sm...@enject.com.au>>
Cc: Jonathan Charles mailto:jonv...@gmail.com>>, 
"cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via 
local PSTN

This is Webex Hybrid Calling (which was formerly Spark Hybrid calling). Whether 
you configure for cloud registered codec devices, or Webex Teams clients, both 
use cases use the same configuration path / scenario to enable PSTN call via 
CUCM.
Sent from my iPhone

On May 15, 2019, at 22:08, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:
Hey guys,

I think this one has changed a little.
We did “Spark hybrid calling” for one customer with the Spark RD devices in CUCM
Honestly, the experience was a little confusing.

I think the new direction is going to be the WebEx Calling via CUCM (it’s in 
preview mode still)
https://help.webex.com/en-us/n15ylys/Explore-Calling-in-Cisco-Webex-Teams-Unified-CM<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fhelp.webex.com%2Fen-us%2Fn15ylys%2FExplore-Calling-in-Cisco-Webex-Teams-Unified-CM=02%7C01%7C%7C9379fa8f35284f2e6bfc08d6d9a6a4be%7C84df9e7fe9f640afb435%7C1%7C0%7C636935707083042561=zbKn4%2Bo6qOMVpZuBvCVfY2L1QZF3s80v9w%2FeBPvmAKw%3D=0>

It’s not parity with the Hybrid Calling yet. (i.e. I think it’s only desktop)
Either way, I’d check out all the details first.

If you are not already on there, make sure you are on the Fabian bot in WebEx 
teams. (Not sure if it’s partners only)
These hybrid features are really starting to rock and roll now.

Cheers,

Tim

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Date: Thursday, 16 May 2019 at 9:30 am
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: "cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via 
local PSTN

Thanks!... looks like I have some more reading to do... so how does it prevent 
anyone from sending a pstn number to my expressway? How does it authenticate 
the Webex devices to pass calls to CUCM for?

Customer has enterprise licensing, so they should be able to do whatever they 
want...


Jonathan



On Wed, May 15, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
You’ll need a specific Webex DNS zone and the traversal trunk really just needs 
to support pre-loaded route headers and SIP parameter preservation (those are 
the most significant differences over the traversal / neighbor zone you might 
have setup for B2B).

It’s a simple enough configuration, but there are a few more moving parts than 
what the marketing may lead one to believe. Here is the configuration 
documentation: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww

Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via local PSTN

[Ryan Huff]

This is Webex Hybrid Calling (which was formerly Spark Hybrid calling). Whether 
you configure for cloud registered codec devices, or Webex Teams clients, both 
use cases use the same configuration path / scenario to enable PSTN call via 
CUCM.

Sent from my iPhone

On May 15, 2019, at 22:08, Tim Smith 
mailto:tim.sm...@enject.com.au>> wrote:

Hey guys,

I think this one has changed a little.
We did “Spark hybrid calling” for one customer with the Spark RD devices in CUCM
Honestly, the experience was a little confusing.

I think the new direction is going to be the WebEx Calling via CUCM (it’s in 
preview mode still)
https://help.webex.com/en-us/n15ylys/Explore-Calling-in-Cisco-Webex-Teams-Unified-CM<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fhelp.webex.com%2Fen-us%2Fn15ylys%2FExplore-Calling-in-Cisco-Webex-Teams-Unified-CM=02%7C01%7C%7C3415229c927c4735729d08d6d9a36719%7C84df9e7fe9f640afb435%7C1%7C0%7C636935693163773518=mKiI9NZtHIy0acs0DrBHnhydCAT10YX9AQVHXWqGH%2BA%3D=0>

It’s not parity with the Hybrid Calling yet. (i.e. I think it’s only desktop)
Either way, I’d check out all the details first.

If you are not already on there, make sure you are on the Fabian bot in WebEx 
teams. (Not sure if it’s partners only)
These hybrid features are really starting to rock and roll now.

Cheers,

Tim

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Jonathan Charles mailto:jonv...@gmail.com>>
Date: Thursday, 16 May 2019 at 9:30 am
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: "cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via 
local PSTN

Thanks!... looks like I have some more reading to do... so how does it prevent 
anyone from sending a pstn number to my expressway? How does it authenticate 
the Webex devices to pass calls to CUCM for?

Customer has enterprise licensing, so they should be able to do whatever they 
want...


Jonathan



On Wed, May 15, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
You’ll need a specific Webex DNS zone and the traversal trunk really just needs 
to support pre-loaded route headers and SIP parameter preservation (those are 
the most significant differences over the traversal / neighbor zone you might 
have setup for B2B).

It’s a simple enough configuration, but there are a few more moving parts than 
what the marketing may lead one to believe. Here is the configuration 
documentation: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2FcloudCollaboration%2Fspark%2Fhybridservices%2Fcallservices%2Fcmgt_b_ciscospark-hybrid-call-service-config-guide.html=02%7C01%7C%7C3415229c927c4735729d08d6d9a36719%7C84df9e7fe9f640afb435%7C1%7C0%7C636935693163783529=rhoOV7yB%2FqTM7xjIPec0dXyzQ29ijSvsGME7RUenYxE%3D=0>

Oh and don’t forget to enable MTLS on the edge and also be aware the ControlHub 
now requires CCM 11.5.1SU3 or better (it detects CCM version via call connector 
on Exp-C). It wouldn’t allow you to enable hybrid calling on cloud registered 
devices otherwise.

You can technically still get away using Expressway 8.11.4, but that’ll soon be 
a deprecated version for hybrid calling (you’ll get an alarm about it), so 
might as well go to 12.5.2 and be done with it.

BTW, if you try to upgrade an 8.x Expressway to 12.5.x, you will interact with 
GLO for the 12.x release key (can’t do it from the self service portal because 
the existing 8.x virtual license is already associated to a PAK and GLO has to 
invalidate that relationship first, then hash your new keys to 12.5.x).

Good Luck!

- Ryan

On May 15, 2019, at 18:42, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Very good question. From what I understand, there’s a special traversal link 
built and it’s all “built-in” and uses the CSS of the remote destination or 
something like that.

I’ve read absolutely zero docs about this. This is all based on a quick convo I 
had. I had the same worries and if I recall correctly, my worries were somewhat 
alleviated.

However, that being said, there is only one template in control hub, so if your 
user needs a different setup on their remote destination (or something like 
that) you need to go make a manual change.

It’s sorta like how there’s only one licensing template in control hub for new 
users. We’re gonna struggle with that. We might have to engage (professional) 
services which make uses of APIs to assign different services for different 
users in webex. But I digress.

-sent from mobile device-


Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Service

Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via local PSTN

[Ryan Huff]

Inbound Flow (PSTN user calls user’s DID):

GW > CCM > (per userWebex Hybrid CTI Device) > Exp-C > Exp-E > ControlHub > 
(rings your cloud registered device)

Outbound Flow (cloud registered device calls PSTN via CCM or other on-prem 
device):

Cloud registered device (ControlHub) > Exp-E > Exp-C > CCM (interacts with the 
dialplan for onnet/offnet calling).

Basically, it’s a B2B call that gets 15 pieces of flair added to it so it can 
utilize your on-prem gateway for PSTN access. When you go through the 
configuration, One of the steps will lead you to creating CTI devices in 
communications manager which share the users DID (Essentially, and on premise 
representation of the cloud registered device). The inbound flow is a little 
unique as it essentially capitalizes on a bastardized form of SNR to ring the 
cloud device.

As far as security goes, you are mostly at the mercy of traditional call policy 
rules (or more specifically writing search rules for your zones).

I have found the following to be two good "reject" policies that tend not to 
interfere with most deployments (though they could if the internal URIs match 
the policy). Most organizations have Directory URIs that ultimately have been 
inherited from the user's email address or other corporate standardizations 
which these policies tend to avoid and also tend to deny routing to a 
surprising amount of obvious junk (typically, I apply CPL at the edge):


  *   ^[0-9,a-z,A-Z]{0,6}@.*
 *The first 0-6 characters are made up of alphanumerics 0-9 and/or 
upper/lower case letters, “@“ anything
*   Example: 123...@domain.com<mailto:123...@domain.com>
*   Example: noa...@domian.com<mailto:noa...@domian.com>
*   Example: 9...@domain.com<mailto:9...@domain.com>

  *   ^[0-9,a-z,A-Z]{0,6}$
 *   The first 0-6 characters are made up of alphanumerics 0-9 and/or 
upper/lower case letter and do not exceed the 6th character
 *   Example: 1000
 *   Example: 
 *   Example: NoAuth
 *

Good Luck!

-Ryan

On May 15, 2019, at 19:30, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

Thanks!... looks like I have some more reading to do... so how does it prevent 
anyone from sending a pstn number to my expressway? How does it authenticate 
the Webex devices to pass calls to CUCM for?

Customer has enterprise licensing, so they should be able to do whatever they 
want...


Jonathan



On Wed, May 15, 2019 at 6:16 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:
You’ll need a specific Webex DNS zone and the traversal trunk really just needs 
to support pre-loaded route headers and SIP parameter preservation (those are 
the most significant differences over the traversal / neighbor zone you might 
have setup for B2B).

It’s a simple enough configuration, but there are a few more moving parts than 
what the marketing may lead one to believe. Here is the configuration 
documentation: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html<https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.cisco.com%2Fc%2Fen%2Fus%2Ftd%2Fdocs%2Fvoice_ip_comm%2FcloudCollaboration%2Fspark%2Fhybridservices%2Fcallservices%2Fcmgt_b_ciscospark-hybrid-call-service-config-guide.html=02%7C01%7C%7C7ee4a253802142b2a30f08d6d98d40fe%7C84df9e7fe9f640afb435%7C1%7C0%7C636935598030629961=1n%2BPAGaE0G2GZrt3Jq7l2uo41jWG0NYoZh5cUcHFAaw%3D=0>

Oh and don’t forget to enable MTLS on the edge and also be aware the ControlHub 
now requires CCM 11.5.1SU3 or better (it detects CCM version via call connector 
on Exp-C). It wouldn’t allow you to enable hybrid calling on cloud registered 
devices otherwise.

You can technically still get away using Expressway 8.11.4, but that’ll soon be 
a deprecated version for hybrid calling (you’ll get an alarm about it), so 
might as well go to 12.5.2 and be done with it.

BTW, if you try to upgrade an 8.x Expressway to 12.5.x, you will interact with 
GLO for the 12.x release key (can’t do it from the self service portal because 
the existing 8.x virtual license is already associated to a PAK and GLO has to 
invalidate that relationship first, then hash your new keys to 12.5.x).

Good Luck!

- Ryan

On May 15, 2019, at 18:42, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Very good question. From what I understand, there’s a special traversal link 
built and it’s all “built-in” and uses the CSS of the remote destination or 
something like that.

I’ve read absolutely zero docs about this. This is all based on a quick convo I 
had. I had the same worries and if I recall correctly, my worries were somewhat 
alleviated.

However, that being said, there is only one template in control hub, so if your 
user needs a different setup on their remote destination (or something like 
that) you need to go make a man

Re: [cisco-voip] Call flow for device registered to Hybrid Cloud via local PSTN

[Ryan Huff]

You’ll need a specific Webex DNS zone and the traversal trunk really just needs 
to support pre-loaded route headers and SIP parameter preservation (those are 
the most significant differences over the traversal / neighbor zone you might 
have setup for B2B).

It’s a simple enough configuration, but there are a few more moving parts than 
what the marketing may lead one to believe. Here is the configuration 
documentation: 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark/hybridservices/callservices/cmgt_b_ciscospark-hybrid-call-service-config-guide.html

Oh and don’t forget to enable MTLS on the edge and also be aware the ControlHub 
now requires CCM 11.5.1SU3 or better (it detects CCM version via call connector 
on Exp-C). It wouldn’t allow you to enable hybrid calling on cloud registered 
devices otherwise.

You can technically still get away using Expressway 8.11.4, but that’ll soon be 
a deprecated version for hybrid calling (you’ll get an alarm about it), so 
might as well go to 12.5.2 and be done with it.

BTW, if you try to upgrade an 8.x Expressway to 12.5.x, you will interact with 
GLO for the 12.x release key (can’t do it from the self service portal because 
the existing 8.x virtual license is already associated to a PAK and GLO has to 
invalidate that relationship first, then hash your new keys to 12.5.x).

Good Luck!

- Ryan

On May 15, 2019, at 18:42, Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:


Very good question. From what I understand, there’s a special traversal link 
built and it’s all “built-in” and uses the CSS of the remote destination or 
something like that.

I’ve read absolutely zero docs about this. This is all based on a quick convo I 
had. I had the same worries and if I recall correctly, my worries were somewhat 
alleviated.

However, that being said, there is only one template in control hub, so if your 
user needs a different setup on their remote destination (or something like 
that) you need to go make a manual change.

It’s sorta like how there’s only one licensing template in control hub for new 
users. We’re gonna struggle with that. We might have to engage (professional) 
services which make uses of APIs to assign different services for different 
users in webex. But I digress.

-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

[University of Guelph Cornerstone with Improve Life tagline]

On May 15, 2019, at 5:20 PM, Jonathan Charles 
mailto:jonv...@gmail.com>> wrote:

Enabling Cisco hybrid call and routing calls to the PSTN using local gateway 
(via Expressway C/E pair).

What search rules do we need on the E and C?

How do we prevent toll fraud if we have E.164 patterns inbound on our 
Expressways?

Am I being paranoid?


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

[Ryan Huff]

The checksum is written to the OS. I’ve heard tell from the old country of a 
TAC agent using it as a way out of an otherwise entitled case.

If you inject a boot table from a RedHat image into a Redhat image, there isn’t 
any functional difference.

I would not inject a boot table from a RedHat into a CentOS based image though  
;).

-Ryan

On May 15, 2019, at 17:26, Charles Goldsmith 
mailto:wo...@justfamily.org>> wrote:

I don't know how true this is, but years ago I was told that when you do an 
install, that the md5sum of the iso is written out in the install, so that TAC 
can tell if it's a legit image that is used to do the installation.

However, at least once on a TAC supplied iso that I've gotten a failure on the 
"check installation media" portion of the install.  When I asked TAC about it, 
they told me to ignore and proceed with the install.  My guess is that the 
particular ISO I had didn't have the correct md5 on it.

Others have installed just fine that I've received from them.

Take that for what it's worth.

Btw, you don't need to use Ultra ISO to make an iso bootable, linux tools can 
do the same thing.  Doesn't cost an Ultra ISO license and you don't have to 
download the ISO to your desktop and then upload it.  Not always feasible when 
doing things remotely.  Not that I've ever made an ISO for a customer, just 
saying :)


On Wed, May 15, 2019 at 3:01 PM Evgeny Izetov 
mailto:eize...@gmail.com>> wrote:
I wonder if TAC also gave up - UltraISO'd it themselves and forgot to add 
Bootable_ :-)

On Wed, May 15, 2019 at 3:46 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:
I remember when it used to as simple as “format /s”

-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | 
le...@uoguelph.ca

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

[University of Guelph Cornerstone with Improve Life tagline]

On May 15, 2019, at 3:22 PM, Charles Goldsmith 
mailto:wo...@justfamily.org>> wrote:

It's not.  And just in case they changed things, I went and downloaded the 
latest 12.0 and 12.5 of both CUCM and CUC and none of them have the bootable 
part of the ISO.

Simply renaming a file doesn't make it bootable :)


On Wed, May 15, 2019 at 1:36 PM Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>> 
wrote:
That.  Can't.  Be.  True.  Right?  If so, Brian Meade has been wasting his time 
with UltraISO.

On Wed, May 15, 2019 at 1:26 PM Evgeny Izetov 
mailto:eize...@gmail.com>> wrote:
That's good to know. Was it 12.x or 11.x?

On Wed, May 15, 2019 at 2:19 PM Haas, Neal 
mailto:nh...@fresnocountyca.gov>> wrote:
I had a TAC Call last week, they told me to add BOOTABLE to the name (in front) 
and that was it. They said all ISO’s are now bootable with the name change…..







From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Evgeny Izetov
Sent: Wednesday, May 15, 2019 11:17 AM
To: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Cc: voyp list, cisco-voip 
(cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] PUT Tool Bootables - what version?

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

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

On Wed, May 15, 2019 at 12:18 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Same with CUPS if I’m not mistaken.

---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

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



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Charles Goldsmith
Sent: Wednesday, May 15, 2019 12:09 PM
To: Evgeny Izetov mailto:eize...@gmail.com>>
Cc: voyp list, cisco-voip