Re: [cisco-voip] moving jabber client between clusters for support?

[Nick Barnett]

Thanks, let me try and clarify a bit.

 

We have a support team that needs to log their Jabber device into multiple 
clusters. Using the "home cluster" setting is too cumbersome and requires CUCM 
access. Our testers do not have CUCM access, but they can modify their own 
local config files.

 

Ultimately, I'm looking for a bootstrap hack, or some other way, to MANUALLY 
and / or STATICALLY define which cluster the Jabber client registers with.

 

This is purely for our support group with CUCM access and our testing team 
without CUCM access. No real end users need to use this feature... but security 
is taking away our CIPC and I need to have another way for them to jump between 
prod clusters...

 

Additionally, I personally have to jump between prod and non-prod, so i want 
this fix for me as well.

 

Ideally, there is something local that a tech can do to their jabber config 
file, bootstrap, or registry that will make their jabber register to a specific 
cluster instead of using UPN discovery.  Granted, some of the installation 
switches seem to fundamentally change how jabber is installed (imagine 
that!)... so another requirement would be NOT HAVING TO uninstall and reinstall 
with different switches. (If in the end, the support team MUST be configured to 
never use UPN, that's fine, i just want to know the best way).

 

Does that make it any clearer? I'm just looking for an easy way to hop between 
clusters without discovery to replace CIPC.

 

Thanks!


On Wed, Apr 28, 2021, at 9:26 AM, Lelio Fulgenzi wrote:
> I may not understand exactly what you’re trying to do, but, I think that will 
> come out during discussion.

>  

> I have a production cluster and a development cluster.

>  

> To switch between the clusters (and to switch users) I have had to do two 
> things:

>  

>  * Create a new service discovery domain with the appropriate servers listed 
> (including development expressway cluster)
>  * Ensure I use the UDS disabled switch when installing Jabber (this allows 
> for different userIDs to be used)
>  

> I know there are a few parameters out there that allow you to push the domain 
> out, but I’m not 100% sure how that works to be honest. Especially because it 
> sounded like they were meant to be done at install time. And still required a 
> domain to be set.

>  

> Let the games begin!

>  

>  

>  

>  


> *From:* cisco-voip  *On Behalf Of *Nick 
> Barnett
> *Sent:* Wednesday, April 28, 2021 10:16 AM
> *To:* cisco-voip 
> *Subject:* [cisco-voip] moving jabber client between clusters for support?

>  

> *CAUTION:* This email originated from outside of the University of Guelph. Do 
> not click links or open attachments unless you recognize the sender and know 
> the content is safe. If in doubt, forward suspicious emails to 
> ith...@uoguelph.ca

>  

> Up until now, I've just been using CIPC as it does everything I needed it to 
> do... for the most part. Starting this year, I have to jump through way too 
> many hoops to keep using CIPC as it is EOL and our security team doesn't like 
> that.

>  

> I'm trying to figure out how to configure my Jabber client to register to a 
> specific non-prod cluster, but I'm not having much luck.

>  

> Our prod has 2 clusters with ILS. the internal SRV record for discovery 
> contains all nodes for both clusters. This works just fine in production and 
> relies on the HOME CLUSTER checkbox for the user to be logged into the 
> correct cluster.

>  

> My problem comes because I work on non-prod systems as well as prod. The only 
> way I've been able to figure out how to get my jabber to jump to non-prod 
> system is to create a NEW srv record that only contains the nodes of the 
> cluster I want to work with and then change the entry in the bootstrap file.

>  

> Is there some way to override discovery and hard code a TFTP or subscriber so 
> that Jabber goes to the intended cluster?  Or, maybe there's a way to do this 
> without a custom SRV record for each non-prod cluster using registry like we 
> could in Cucilync? Some other trick?  I'm hopeful there is some combination 
> of bootstrap settings that I need to figure out, but all of those fields can 
> be a nightmare to get correct without unintentionally breaking something else.

>  

> Thanks,

> Nick

>  



Thanks,
Nick
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[cisco-voip] moving jabber client between clusters for support?

[Nick Barnett]

Up until now, I've just been using CIPC as it does everything I needed it to 
do... for the most part. Starting this year, I have to jump through way too 
many hoops to keep using CIPC as it is EOL and our security team doesn't like 
that.

I'm trying to figure out how to configure my Jabber client to register to a 
specific non-prod cluster, but I'm not having much luck.

Our prod has 2 clusters with ILS. the internal SRV record for discovery 
contains all nodes for both clusters. This works just fine in production and 
relies on the HOME CLUSTER checkbox for the user to be logged into the correct 
cluster.

My problem comes because I work on non-prod systems as well as prod. The only 
way I've been able to figure out how to get my jabber to jump to non-prod 
system is to create a NEW srv record that only contains the nodes of the 
cluster I want to work with and then change the entry in the bootstrap file.

Is there some way to override discovery and hard code a TFTP or subscriber so 
that Jabber goes to the intended cluster?  Or, maybe there's a way to do this 
without a custom SRV record for each non-prod cluster using registry like we 
could in Cucilync? Some other trick?  I'm hopeful there is some combination of 
bootstrap settings that I need to figure out, but all of those fields can be a 
nightmare to get correct without unintentionally breaking something else.

Thanks,
Nick
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Re: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.

[Nick Barnett]

One final follow-up on this... turns out that YES there was a PSTN switching 
issue... but also we could have increased a timer on survivability.tcl on the 
ingress leg. Just leaving this here in case anyone ever does a search.

application
service survivability
param setup-timeout 15


On Mon, Apr 19, 2021, at 8:24 AM, Nick Barnett wrote:
> The PSTN issue was resolved by the time I could get more traces. i still have 
> no idea what timer was popping, TAC couldn't figure it out either... so I 
> guess hope you don't ever have a mystery 7 second timer that cuts off your 
> calls.
> 
> On Fri, Apr 16, 2021, at 11:25 AM, Nick Barnett wrote:
>> Unfortunately, I only had ccsip and ccapi inout debugs running... I'll add 
>> those 2 other sip debugs the next time. This is what I had handy from last 
>> night. I'm sure new traces would be very helpful, but can't get to it right 
>> now and wanted to at least respond.
>> 
>> Here is a sanitized copy of the cancel.  The time between the trying and the 
>> cancel is always between 6.8 and 6.9 seconds.  
>> 
>> CANCEL sip:1764...@voip.centurylink.com:5100 SIP/2.0
>> Via: SIP/2.0/UDP voip.centurylink.com:5060;branch=z9hG4bK2804DC216B
>> From: sip:165222@10.10.10.10;tag=7FF3BC40-14BB
>> To: 
>> Date: Thu, 15 Apr 2021 18:39:42 GMT
>> Call-ID: c6ab9407-9d5011eb-88bf8f36-hi-...@voip.centurylink.com 
>> <mailto:c6ab9407-9d5011eb-88bf8f36-4c717...@voip.centurylink.com>
>> CSeq: 102 CANCEL
>> Max-Forwards: 70
>> Timestamp: 1618511989
>> Reason: Q.850;cause=0
>> Content-Length: 0
>> 
>> That pesky 0 for the cause code is making life difficult.
>> 
>> On Fri, Apr 16, 2021, at 10:29 AM, Sreekanth Narayanan (sreenara) wrote:
>>> Nick,
>>> 
>>> What's the disconnect cause from the CUBE? 102?
>>> 
>>> Do you have logs for this call? Would be clear which timers are expiring, 
>>> causing the problem.
>>> debug ccsip message
>>> debug ccsip error
>>> debug ccsip info
>>> 
>>> -sreekanth
>>> 
>>> 
>>> *From:* cisco-voip  on behalf of Nick 
>>> Barnett 
>>> *Sent:* Friday, April 16, 2021 6:53 PM
>>> *To:* cisco-voip 
>>> *Subject:* [cisco-voip] Outbound SIP connection failing in CUBE due to some 
>>> timer... maybe.
>>>  
>>> Yes, very vague subject. Sorry about that. Some calls to certain wireless 
>>> carriers on our ITSP connections have started failing. 
>>> 
>>> Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP
>>> 
>>> The call goes out Lumen, the 401 auth and challenge response are fine, the 
>>> INVITE is then sent with SDP. We get a TRYING response which we immediately 
>>> ACK. Up until this point, the entire call flow is NORMAL. 
>>> 
>>> If we don't receive a 18X response within  7 Seconds, the, the CUBE sends a 
>>> cancel. Yes, the CUBE.
>>> 
>>> It appears that the far end is taking too long to send the 18X message. we 
>>> involved our carrier and they can see the 18X come back a split second 
>>> later (sometimes), but our side has already closed the connection.
>>> 
>>> I looked at all of the sip-ua timers and retry settings. nothing adds up to 
>>> 7 seconds. Most timers are set to 500 msec. I'm not sure where to look? 
>>> It's not on the sip profile. i tried bumping up the connect, update, info 
>>> and trying timers (one at a time), but it didn't make any difference. Maybe 
>>> I was supposed to do something to make sip-ua changes "kick in" like bounce 
>>> the sip service which I didn't do... not sure on that part.
>>> 
>>> Please tell me there is something simple I'm missing. Pointers?
>>> 
>>> Thanks,
>>> Nick
>>> 
>>> 
>>> p.s. : some possibly relevant config  and the timers and retries from my 
>>> sip-ua
>>> 
>>> retry invite 2
>>> retry response 6
>>> retry bye 10
>>> retry cancel 10
>>> retry prack 10
>>> retry update 6
>>> retry rel1xx 6
>>> retry notify 10
>>> retry refer 10
>>> retry info 6
>>> retry register 6
>>> retry subscribe 6
>>> retry keepalive 6
>>> retry options 6
>>> timers trying 500
>>> timers expires 18
>>> timers connect 500
>>> timers connection aging 5
>>> timers disconnect 500
>>> timers prack 500
>>> timers update 500
>>> timers rel1xx 500
>>> timers notify 750
>>> timers refer 500
>>> timers hold 2880
>>> timers info 500
>>> timers register 500
>>> timers buffer-invite 0
>>> timers keepalive down 30
>>> timers keepalive active 120
>>> timers dns registrar-cache 3600
>>> timers options 500
>> 
>> 
>> Thanks,
>> Nick
>> 
>> ___
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>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
> 
> 
> Thanks,
> Nick
> 
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> 


Thanks,
Nick
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Re: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.

[Nick Barnett]

The PSTN issue was resolved by the time I could get more traces. i still have 
no idea what timer was popping, TAC couldn't figure it out either... so I guess 
hope you don't ever have a mystery 7 second timer that cuts off your calls.

On Fri, Apr 16, 2021, at 11:25 AM, Nick Barnett wrote:
> Unfortunately, I only had ccsip and ccapi inout debugs running... I'll add 
> those 2 other sip debugs the next time. This is what I had handy from last 
> night. I'm sure new traces would be very helpful, but can't get to it right 
> now and wanted to at least respond.
> 
> Here is a sanitized copy of the cancel.  The time between the trying and the 
> cancel is always between 6.8 and 6.9 seconds.  
> 
> CANCEL sip:1764...@voip.centurylink.com:5100 SIP/2.0
> Via: SIP/2.0/UDP voip.centurylink.com:5060;branch=z9hG4bK2804DC216B
> From: sip:165222@10.10.10.10;tag=7FF3BC40-14BB
> To: 
> Date: Thu, 15 Apr 2021 18:39:42 GMT
> Call-ID: c6ab9407-9d5011eb-88bf8f36-hi-...@voip.centurylink.com 
> <mailto:c6ab9407-9d5011eb-88bf8f36-4c717...@voip.centurylink.com>
> CSeq: 102 CANCEL
> Max-Forwards: 70
> Timestamp: 1618511989
> Reason: Q.850;cause=0
> Content-Length: 0
> 
> That pesky 0 for the cause code is making life difficult.
> 
> On Fri, Apr 16, 2021, at 10:29 AM, Sreekanth Narayanan (sreenara) wrote:
>> Nick,
>> 
>> What's the disconnect cause from the CUBE? 102?
>> 
>> Do you have logs for this call? Would be clear which timers are expiring, 
>> causing the problem.
>> debug ccsip message
>> debug ccsip error
>> debug ccsip info
>> 
>> -sreekanth
>> 
>> 
>> *From:* cisco-voip  on behalf of Nick 
>> Barnett 
>> *Sent:* Friday, April 16, 2021 6:53 PM
>> *To:* cisco-voip 
>> *Subject:* [cisco-voip] Outbound SIP connection failing in CUBE due to some 
>> timer... maybe.
>>  
>> Yes, very vague subject. Sorry about that. Some calls to certain wireless 
>> carriers on our ITSP connections have started failing. 
>> 
>> Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP
>> 
>> The call goes out Lumen, the 401 auth and challenge response are fine, the 
>> INVITE is then sent with SDP. We get a TRYING response which we immediately 
>> ACK. Up until this point, the entire call flow is NORMAL. 
>> 
>> If we don't receive a 18X response within  7 Seconds, the, the CUBE sends a 
>> cancel. Yes, the CUBE.
>> 
>> It appears that the far end is taking too long to send the 18X message. we 
>> involved our carrier and they can see the 18X come back a split second later 
>> (sometimes), but our side has already closed the connection.
>> 
>> I looked at all of the sip-ua timers and retry settings. nothing adds up to 
>> 7 seconds. Most timers are set to 500 msec. I'm not sure where to look? It's 
>> not on the sip profile. i tried bumping up the connect, update, info and 
>> trying timers (one at a time), but it didn't make any difference. Maybe I 
>> was supposed to do something to make sip-ua changes "kick in" like bounce 
>> the sip service which I didn't do... not sure on that part.
>> 
>> Please tell me there is something simple I'm missing. Pointers?
>> 
>> Thanks,
>> Nick
>> 
>> 
>> p.s. : some possibly relevant config  and the timers and retries from my 
>> sip-ua
>> 
>> retry invite 2
>> retry response 6
>> retry bye 10
>> retry cancel 10
>> retry prack 10
>> retry update 6
>> retry rel1xx 6
>> retry notify 10
>> retry refer 10
>> retry info 6
>> retry register 6
>> retry subscribe 6
>> retry keepalive 6
>> retry options 6
>> timers trying 500
>> timers expires 18
>> timers connect 500
>> timers connection aging 5
>> timers disconnect 500
>> timers prack 500
>> timers update 500
>> timers rel1xx 500
>> timers notify 750
>> timers refer 500
>> timers hold 2880
>> timers info 500
>> timers register 500
>> timers buffer-invite 0
>> timers keepalive down 30
>> timers keepalive active 120
>> timers dns registrar-cache 3600
>> timers options 500
> 
> 
> Thanks,
> Nick
> 
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> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


Thanks,
Nick
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Re: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.

[Nick Barnett]

Unfortunately, I only had ccsip and ccapi inout debugs running... I'll add 
those 2 other sip debugs the next time. This is what I had handy from last 
night. I'm sure new traces would be very helpful, but can't get to it right now 
and wanted to at least respond.

Here is a sanitized copy of the cancel.  The time between the trying and the 
cancel is always between 6.8 and 6.9 seconds.  

CANCEL sip:1764...@voip.centurylink.com:5100 SIP/2.0
Via: SIP/2.0/UDP voip.centurylink.com:5060;branch=z9hG4bK2804DC216B
From: sip:165222@10.10.10.10;tag=7FF3BC40-14BB
To: 
Date: Thu, 15 Apr 2021 18:39:42 GMT
Call-ID: c6ab9407-9d5011eb-88bf8f36-hi-...@voip.centurylink.com 
<mailto:c6ab9407-9d5011eb-88bf8f36-4c717...@voip.centurylink.com>
CSeq: 102 CANCEL
Max-Forwards: 70
Timestamp: 1618511989
Reason: Q.850;cause=0
Content-Length: 0

That pesky 0 for the cause code is making life difficult.

On Fri, Apr 16, 2021, at 10:29 AM, Sreekanth Narayanan (sreenara) wrote:
> Nick,
> 
> What's the disconnect cause from the CUBE? 102?
> 
> Do you have logs for this call? Would be clear which timers are expiring, 
> causing the problem.
> debug ccsip message
> debug ccsip error
> debug ccsip info
> 
> -sreekanth
> 
> 
> *From:* cisco-voip  on behalf of Nick 
> Barnett 
> *Sent:* Friday, April 16, 2021 6:53 PM
> *To:* cisco-voip 
> *Subject:* [cisco-voip] Outbound SIP connection failing in CUBE due to some 
> timer... maybe. 
>  
> Yes, very vague subject. Sorry about that. Some calls to certain wireless 
> carriers on our ITSP connections have started failing. 
> 
> Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP
> 
> The call goes out Lumen, the 401 auth and challenge response are fine, the 
> INVITE is then sent with SDP. We get a TRYING response which we immediately 
> ACK. Up until this point, the entire call flow is NORMAL. 
> 
> If we don't receive a 18X response within  7 Seconds, the, the CUBE sends a 
> cancel. Yes, the CUBE.
> 
> It appears that the far end is taking too long to send the 18X message. we 
> involved our carrier and they can see the 18X come back a split second later 
> (sometimes), but our side has already closed the connection.
> 
> I looked at all of the sip-ua timers and retry settings. nothing adds up to 7 
> seconds. Most timers are set to 500 msec. I'm not sure where to look? It's 
> not on the sip profile. i tried bumping up the connect, update, info and 
> trying timers (one at a time), but it didn't make any difference. Maybe I was 
> supposed to do something to make sip-ua changes "kick in" like bounce the sip 
> service which I didn't do... not sure on that part.
> 
> Please tell me there is something simple I'm missing. Pointers?
> 
> Thanks,
> Nick
> 
> 
> p.s. : some possibly relevant config  and the timers and retries from my 
> sip-ua
> 
> retry invite 2
> retry response 6
> retry bye 10
> retry cancel 10
> retry prack 10
> retry update 6
> retry rel1xx 6
> retry notify 10
> retry refer 10
> retry info 6
> retry register 6
> retry subscribe 6
> retry keepalive 6
> retry options 6
> timers trying 500
> timers expires 18
> timers connect 500
> timers connection aging 5
> timers disconnect 500
> timers prack 500
> timers update 500
> timers rel1xx 500
> timers notify 750
> timers refer 500
> timers hold 2880
> timers info 500
> timers register 500
> timers buffer-invite 0
> timers keepalive down 30
> timers keepalive active 120
> timers dns registrar-cache 3600
> timers options 500


Thanks,
Nick
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Re: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.

[Nick Barnett]

This was working. It broke yesterday at like 11am.  Lumen says ATT has an 
interconnect issue. ATT is clueless about it though.  It's all over the place 
though. Some calls that are routing through ATT are failing. Some that end on 
Tmobile are failing.  I will try to refresh the auth password on it... that's a 
weird one.

On Fri, Apr 16, 2021, at 11:04 AM, Carlo Calabrese wrote:
> 
> Was this working before?
> I had problems before on a 4331 CUBE. I had to redo the password on the CUBE 
> to Lumen.
> it was the same password that was already there. and a reboot didn't fix it.
> 
> 
> 
> On Friday, April 16, 2021, 08:32:22 AM PDT, Sreekanth Narayanan (sreenara) 
> via cisco-voip  wrote:
> 
> 
> Nick,
> 
> What's the disconnect cause from the CUBE? 102?
> 
> Do you have logs for this call? Would be clear which timers are expiring, 
> causing the problem.
> debug ccsip message
> debug ccsip error
> debug ccsip info
> 
> -sreekanth
> 
> 
> *From:* cisco-voip  on behalf of Nick 
> Barnett 
> *Sent:* Friday, April 16, 2021 6:53 PM
> *To:* cisco-voip 
> *Subject:* [cisco-voip] Outbound SIP connection failing in CUBE due to some 
> timer... maybe. 
>  
> Yes, very vague subject. Sorry about that. Some calls to certain wireless 
> carriers on our ITSP connections have started failing. 
> 
> Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP
> 
> The call goes out Lumen, the 401 auth and challenge response are fine, the 
> INVITE is then sent with SDP. We get a TRYING response which we immediately 
> ACK. Up until this point, the entire call flow is NORMAL. 
> 
> If we don't receive a 18X response within  7 Seconds, the, the CUBE sends a 
> cancel. Yes, the CUBE.
> 
> It appears that the far end is taking too long to send the 18X message. we 
> involved our carrier and they can see the 18X come back a split second later 
> (sometimes), but our side has already closed the connection.
> 
> I looked at all of the sip-ua timers and retry settings. nothing adds up to 7 
> seconds. Most timers are set to 500 msec. I'm not sure where to look? It's 
> not on the sip profile. i tried bumping up the connect, update, info and 
> trying timers (one at a time), but it didn't make any difference. Maybe I was 
> supposed to do something to make sip-ua changes "kick in" like bounce the sip 
> service which I didn't do... not sure on that part.
> 
> Please tell me there is something simple I'm missing. Pointers?
> 
> Thanks,
> Nick
> 
> 
> p.s. : some possibly relevant config  and the timers and retries from my 
> sip-ua
> 
> retry invite 2
> retry response 6
> retry bye 10
> retry cancel 10
> retry prack 10
> retry update 6
> retry rel1xx 6
> retry notify 10
> retry refer 10
> retry info 6
> retry register 6
> retry subscribe 6
> retry keepalive 6
> retry options 6
> timers trying 500
> timers expires 18
> timers connect 500
> timers connection aging 5
> timers disconnect 500
> timers prack 500
> timers update 500
> timers rel1xx 500
> timers notify 750
> timers refer 500
> timers hold 2880
> timers info 500
> timers register 500
> timers buffer-invite 0
> timers keepalive down 30
> timers keepalive active 120
> timers dns registrar-cache 3600
> timers options 500
> ___
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> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


Thanks,
Nick
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[cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.

[Nick Barnett]

Yes, very vague subject. Sorry about that. Some calls to certain wireless 
carriers on our ITSP connections have started failing. 

Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP

The call goes out Lumen, the 401 auth and challenge response are fine, the 
INVITE is then sent with SDP. We get a TRYING response which we immediately 
ACK. Up until this point, the entire call flow is NORMAL. 

If we don't receive a 18X response within  7 Seconds, the, the CUBE sends a 
cancel. Yes, the CUBE.

It appears that the far end is taking too long to send the 18X message. we 
involved our carrier and they can see the 18X come back a split second later 
(sometimes), but our side has already closed the connection.

I looked at all of the sip-ua timers and retry settings. nothing adds up to 7 
seconds. Most timers are set to 500 msec. I'm not sure where to look? It's not 
on the sip profile. i tried bumping up the connect, update, info and trying 
timers (one at a time), but it didn't make any difference. Maybe I was supposed 
to do something to make sip-ua changes "kick in" like bounce the sip service 
which I didn't do... not sure on that part.

Please tell me there is something simple I'm missing. Pointers?

Thanks,
Nick


p.s. : some possibly relevant config  and the timers and retries from my sip-ua

retry invite 2
retry response 6
retry bye 10
retry cancel 10
retry prack 10
retry update 6
retry rel1xx 6
retry notify 10
retry refer 10
retry info 6
retry register 6
retry subscribe 6
retry keepalive 6
retry options 6
timers trying 500
timers expires 18
timers connect 500
timers connection aging 5
timers disconnect 500
timers prack 500
timers update 500
timers rel1xx 500
timers notify 750
timers refer 500
timers hold 2880
timers info 500
timers register 500
timers buffer-invite 0
timers keepalive down 30
timers keepalive active 120
timers dns registrar-cache 3600
timers options 500___
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[cisco-voip] Report or query to find phones with web server enabled?

[Nick Barnett]

Does anyone have a way to do this? This setting is stored in the device 
specific settings for the phone, so retrieving the entry in the device table 
isn't sufficient. There is also the advanceddeviceconfigparams table, but I 
can't figure out how to interract with it properly. 

Usually I start out with something simple like:
select * from device where name='SEPDEADBEEF1234'

and then I can drill down, my next attempt was:
select * from advanceddeviceconfigparams where fkdevice='PKID-OF-MY-PHONE'

which returned nothing

then I went to my lab cluster and did:
select * from advanceddeviceconfigparams

and nothing came back.

I was expecting a list of UUIDs and settings to come back that I would then 
continue to step through until I got the data I needed. So I guess I don't know 
how device table interracts with advanceddeviceconfigparams. What am i missing 
here?

Any help is appreciated.

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

[Nick Barnett]

That PLAR idea is fun :)

2 audio files at once is not ideal...

I haven't tested this yet, but this is where I'm going. Looking at possibly 
turning this into a container to post audio files to and get back a muxed Left 
/ Right "stereo" conversation.
https://securitronlinux.com/debian-testing/merge-left-and-right-audio-tracks-into-a-stereo-file-with-sox-on-linux/

Now I just need to figure out how long a recorded call can be... I guess I 
hadn't thought of that part as either a requirement or request...


On Mon, Jan 11, 2021, at 9:52 PM, Lelio Fulgenzi wrote:
> 
> I wonder if a PLAR on a DN that is listed as an alternate extension would 
> help.  
> 
> So, 
> 
> Tap PLAR
> Then merge calls
> 
> You could even move the merge Soft key earlier?
> 
> 
> Sent from my iPhone
> 
>> On Jan 11, 2021, at 9:53 PM, Kent Roberts  wrote:
>> 
>>  
>> 

>> *CAUTION:* This email originated from outside of the University of Guelph. 
>> Do not click links or open attachments unless you recognize the sender and 
>> know the content is safe. If in doubt, forward suspicious emails to 
>> ith...@uoguelph.ca
>> 
>> 

>> My guess is it won’t get fixed as it would help us all out! LOL. 
>> 
>> 
>>> On Jan 11, 2021, at 2:18 PM, NateCCIE  wrote:
>>> 
>>> There is a record softkey that does what you’re asking Leilo.  The problem 
>>> with Unity Live record is the CUCM BIB calls the number twice, one for TX 
>>> and one for RX, and you get two-one sided voicemails in the inbox.  If that 
>>> gets fixed, it’s an amazing option.
>>>  
>>> *From:* cisco-voip  *On Behalf Of 
>>> *Lelio Fulgenzi
>>> *Sent:* Monday, January 11, 2021 2:00 PM
>>> *To:* Nick Barnett ; Brian Meade 
>>> *Cc:* cisco-voip 
>>> *Subject:* Re: [cisco-voip] Alternatives for MediaSense simple recording?
>>>  
>>> It’s too bad that CUCM doesn’t have a LiveRecord softkey macro that does 
>>> the conferencing and dialing of the live record extension.
>>>  
>>> To ask people to press conference and then dial live record and the 
>>> conference again, is just way to much to ask. I think.
>>>  
>>> Has Live Record support from CUCM side improved at all?
>>>  
>>>  
>>> *From:* Nick Barnett  
>>> *Sent:* Monday, January 11, 2021 3:50 PM
>>> *To:* Brian Meade ; Lelio Fulgenzi 
>>> *Cc:* cisco-voip 
>>> *Subject:* Re: [cisco-voip] Alternatives for MediaSense simple recording?
>>>  
>>> *CAUTION:* This email originated from outside of the University of Guelph. 
>>> Do not click links or open attachments unless you recognize the sender and 
>>> know the content is safe. If in doubt, forward suspicious emails to 
>>> ith...@uoguelph.ca
>>>  
>>> Unity live record, that's one I haven't thought of yet. Thanks!
>>>  
>>> I'm pretty sure, mediasense is totally dead. We just upgraded to CUCM 12.5 
>>> SU3 in October. MediaSense 11.5 su2 said it was compatible with CUCM 12.x, 
>>> but in this case, it only meant 12.x THRU 12.5 SU2.  The BU's solution was 
>>> to downgrade to SU2. We kind of pushed them and they came back with a fix. 
>>> Apparently between CUCM 12.5 SU2 and 12.5 SU3, CUCM forced HTTPs for AXL 
>>> connections.  to fix it, TAC had to root into my nodes and make a change to 
>>> the haproxy.conf file to stop forcing HTTPS.
>>>  
>>> This whole mess took me right up to the last day of support and I think 
>>> everyone at cisco hated me, but they were clear there would be no more 
>>> support for this monster. meh
>>>  
>>> Thanks,
>>> Nick
>>>  
>>> On Mon, Jan 11, 2021, at 2:24 PM, Brian Meade wrote:
>>>> Unity Connection Live Record may be an option you could try and have it 
>>>> conference in that number.
>>>>  
>>>> I think MediaSense is still around for video call handlers/video 
>>>> voicemail/video on hold if I remember correctly.  I think they only killed 
>>>> it for recording calls.
>>>>  
>>>> On Mon, Jan 11, 2021 at 1:56 PM Lelio Fulgenzi  wrote:
>>>>> I sure was sad when they EOL’ed Media Sense. I really wanted to do video 
>>>>> call handlers and video voicemail and greetings. 

>>>>>  

>>>>> Take a look at https://www.mns.vc/ they might have what you’re looking 
>>>>> for.

>>>>>  

>>>>> Lelio

>>>>>  

>>>>>  

>>>>>

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

[Nick Barnett]

Unity live record, that's one I haven't thought of yet. Thanks!

I'm pretty sure, mediasense is totally dead. We just upgraded to CUCM 12.5 SU3 
in October. MediaSense 11.5 su2 said it was compatible with CUCM 12.x, but in 
this case, it only meant 12.x THRU 12.5 SU2.  The BU's solution was to 
downgrade to SU2. We kind of pushed them and they came back with a fix. 
Apparently between CUCM 12.5 SU2 and 12.5 SU3, CUCM forced HTTPs for AXL 
connections.  to fix it, TAC had to root into my nodes and make a change to the 
haproxy.conf file to stop forcing HTTPS.

This whole mess took me right up to the last day of support and I think 
everyone at cisco hated me, but they were clear there would be no more support 
for this monster. meh

Thanks,
Nick

On Mon, Jan 11, 2021, at 2:24 PM, Brian Meade wrote:
> Unity Connection Live Record may be an option you could try and have it 
> conference in that number.
> 
> I think MediaSense is still around for video call handlers/video 
> voicemail/video on hold if I remember correctly.  I think they only killed it 
> for recording calls.
> 
> On Mon, Jan 11, 2021 at 1:56 PM Lelio Fulgenzi  wrote:
>> I sure was sad when they EOL’ed Media Sense. I really wanted to do video 
>> call handlers and video voicemail and greetings. 

>> __ __

>> Take a look at https://www.mns.vc/ they might have what you’re looking 
>> for.

>> __ __

>> Lelio____

>> __ __

>> __ __


>> *From:* cisco-voip  *On Behalf Of *Nick 
>> Barnett
>> *Sent:* Monday, January 11, 2021 12:54 PM
>> *To:* cisco-voip 
>> *Subject:* [cisco-voip] Alternatives for MediaSense simple recording?

>> __ __

>> *CAUTION:* This email originated from outside of the University of Guelph. 
>> Do not click links or open attachments unless you recognize the sender and 
>> know the content is safe. If in doubt, forward suspicious emails to 
>> IThelp@uoguelph.ca

>> __ __

>> Hey folks. What are people using now that MediaSense is EOL? It was fine for 
>> what it was. It just recorded anything you threw into it. it weaseled it's 
>> way into some weird apps we have, and now I'm kinda stuck.  We have an 
>> iphone app that was developed to work in areas with poor data connectivity. 
>> It creates a conference call to a PSTN number that routes into our system 
>> and is a route pattern attached to a SIP trunk directly to MediaSense. 

>> __ __

>> From there, we use APIs to pull the file down and save it using meta data 
>> from the initial call.

>> __ __

>> We aren't using ANY of the recording profiles or advanced features of 
>> mediasense. Our new recording system is NICE Engage and they don't offer any 
>> way to record via route patterns.

>> __ __

>> Are there any open source, or really ANYTHING else out there that can do 
>> this simple procedure? The most basic of requirements are 1) non proprietary 
>> audio format 2) retrievable with an API or script. My cisco account team can 
>> only recommend Webex for recording which doesn't look to allow recording 
>> with a route pattern. Our VAR sells NICE which requires an extra application 
>> to kick of a recording like this. 

>> __ __

>> What are you guys using? Any suggestions for me?

>> __ __

>> Thanks,

>> Nick

>> __ __

>> P.S. just to be clear, MeidaSense is not our quality assurance platform. We 
>> use NICE Engage for that and it's fine for now... just looking for something 
>> to fill the gap left by a disappearing MediaSense and our route pattern 
>> recording method.

>> __ __

>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip


Thanks,
Nick
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[cisco-voip] Alternatives for MediaSense simple recording?

[Nick Barnett]

Hey folks. What are people using now that MediaSense is EOL? It was fine for 
what it was. It just recorded anything you threw into it. it weaseled it's way 
into some weird apps we have, and now I'm kinda stuck.  We have an iphone app 
that was developed to work in areas with poor data connectivity. It creates a 
conference call to a PSTN number that routes into our system and is a route 
pattern attached to a SIP trunk directly to MediaSense. 

>From there, we use APIs to pull the file down and save it using meta data from 
>the initial call.

We aren't using ANY of the recording profiles or advanced features of 
mediasense. Our new recording system is NICE Engage and they don't offer any 
way to record via route patterns.

Are there any open source, or really ANYTHING else out there that can do this 
simple procedure? The most basic of requirements are 1) non proprietary audio 
format 2) retrievable with an API or script. My cisco account team can only 
recommend Webex for recording which doesn't look to allow recording with a 
route pattern. Our VAR sells NICE which requires an extra application to kick 
of a recording like this. 

What are you guys using? Any suggestions for me?

Thanks,
Nick

P.S. just to be clear, MeidaSense is not our quality assurance platform. We use 
NICE Engage for that and it's fine for now... just looking for something to 
fill the gap left by a disappearing MediaSense and our route pattern recording 
method.
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[cisco-voip] Single user with multiple device profiles?

[Nick Barnett]

For some reason I thought I'd seen this before, but it's eluding me. I have a 
need for a single user to have multiple extension mobility device profiles. I 
can't even find where it is supported or unsupported. Anyone have any advice on 
how to get this accomplished, whether it's a trick or a formal procedure?

Thanks,
Nick
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Re: [cisco-voip] Where does CUCM/informix store the recordingMediaSource and recordingFlag for phones/lines?

[Nick Barnett]

Perfect! Thank you!

If anyone is searching this in the future, here is the ugly query I came up 
with that works as an executeSQL AXL call. If you try to run it from the CLI 
you'll need to mess with the parentheses because the CLI doesn't like them. I 
also limited the rows to 10 so it doesn't blow up your server (I don't even 
know if that's possible on AXL, but worth the warning I suppose).


SELECT FIRST 10 device.name as Device, numplan.dnorpattern as PhoneNumber, 
routepartition.name as Partition, typepreferredmediasource.name as 
RecordingSource
FROM (((devicenumplanmap INNER JOIN device ON devicenumplanmap.fkdevice = 
device.pkid)
INNER JOIN numplan ON devicenumplanmap.fknumplan = numplan.pkid) INNER JOIN 
typepreferredmediasource ON devicenumplanmap.tkpreferredmediasource = 
typepreferredmediasource.enum)
INNER JOIN routepartition ON numplan.fkroutepartition = routepartition.pkid
WHERE typepreferredmediasource.name =’Gateway Preferred’
AND device.name like 'SEP%'

This worked for me, you may want to add different where clauses or change 
things around.

Nick

On Wed, Sep 11, 2019, at 10:25 AM, Tucci, Ben via cisco-voip wrote:
> That should be tkpreferredmediasource on the devicenumplanmap; which is where 
> the line info is stored for a given device. You could update based on 
> criteria after joining the device on fkdevice or the dnorpattern fron numplan 
> (fknumplan.)
> 
> 
> sql select * from typepreferredmediasource
> enum name moniker
>  = ==
> 1 Gateway Preferred PREFERRED_MEDIA_SOURCE_GATEWAY
> 2 Phone Preferred PREFERRED_MEDIA_SOURCE_PHONE
> 
> sql select FIRST 1 pkid, fkdevice, tkpreferredmediasource from 
> devicenumplanmap
> pkid fkdevice tkpreferredmediasource
>   
> ==
> 00045024-647d-453d-a281-e3ccc5556fc7 4bb930c6-0cd3-cdbd-8080-39e624bf8028 1
> 
> 
> 
> 
> 
> *From:* cisco-voip  on behalf of Nick 
> Barnett 
> *Sent:* Wednesday, September 11, 2019 8:12 AM
> *To:* Charles Goldsmith 
> *Cc:* voip puck 
> *Subject:* Re: [cisco-voip] Where does CUCM/informix store the 
> recordingMediaSource and recordingFlag for phones/lines? 
> 
> Valid question. We hit bug  CSCvr11455  
> <https://urldefense.proofpoint.com/v2/url?u=https-3A__bst.cloudapps.cisco.com_bugsearch_bug_CSCvr11455_-3Frfs-3Diqvred=DwMFAg=RyOedRvjc7OSsfc0bTI76Q=7N3S3VwV170meOMF2t3MxSPtfF4yZYjo-cGnk0FWjNE=Tc1mvbrADjg7eEZZsrSTqiBiUna2bqF5EyNQb0MY3Xs=Pn3GCVMLbpN4HdI98Tj9dFeAuDLszykflweRcgxKzJY=>a
>  week ago and it crippled our company for a few days because it broke A Cisco 
> DB and affected Extension Mobility. We are in an "enhanced" change freeze 
> window until Cisco can get us a patch. Just trying to keep this other project 
> rolling while we're stuck on the support side. The last time I exported all 
> phones it took almost a full day and I can't risk that right now. This option 
> would definitely work in most cases.
> 
> Not to mention, I need to automate some stuff for this switchover, so I'd 
> like to understand what I'm dealing with.
> 
> Thanks,
> Nick
> 
> On Wed, Sep 11, 2019, at 9:59 AM, Charles Goldsmith wrote:
>> I don't have an answer to your question, but why not just export all phones 
>> and filter that column in excel?
>> 
>> On Wed, Sep 11, 2019 at 9:57 AM Nick Barnett  wrote:
>>> __
>>> We're changing our recording platform and have to move from BiB Phone 
>>> Preferred to Gateway Preferred. I know we have many phones already 
>>> (erroneously) set to gateway preferred. The way we are set up, this isn't 
>>> causing an issue but will in the future. I need to find out how many phones 
>>> we have set this way. I can't find a way to search on this setting, so I 
>>> turned to SQL and AXL.
>>> 
>>> 
>>> When working with the AXL API, I can use a getPhone call and pull down the 
>>> appropriate config. I can also use an updatePhone call to change the 
>>> recording media source from "Phone Preferred" to "Gateway Preferred" and 
>>> recordingFlag to and from "Automatic Call Recording Enabled" and "Selective 
>>> Call Recording Enabled." When I look at the numplan, device and 
>>> devicenumplanmap, I don't see where the recording media source or 
>>> recordingFlag are stored.
>>> 
>>> 
>>> I've tried pulling down the rows of the previously mentioned tables via a 
>>> SQL select, then changing the settings in CUCM and pulling the same tables 
>>> down to see if anything changed. Nothing changes so I think it has to be 
>>> storing somewhere else or I'm doin

Re: [cisco-voip] Where does CUCM/informix store the recordingMediaSource and recordingFlag for phones/lines?

[Nick Barnett]

Valid question. We hit bug CSCvr11455  
<https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvr11455/?rfs=iqvred>a week 
ago and it crippled our company for a few days because it broke A Cisco DB and 
affected Extension Mobility. We are in an "enhanced" change freeze window until 
Cisco can get us a patch. Just trying to keep this other project rolling while 
we're stuck on the support side. The last time I exported all phones it took 
almost a full day and I can't risk that right now. This option would definitely 
work in most cases.

Not to mention, I need to automate some stuff for this switchover, so I'd like 
to understand what I'm dealing with.

Thanks,
Nick

On Wed, Sep 11, 2019, at 9:59 AM, Charles Goldsmith wrote:
> I don't have an answer to your question, but why not just export all phones 
> and filter that column in excel?
> 
> On Wed, Sep 11, 2019 at 9:57 AM Nick Barnett  wrote:
>> __
>> We're changing our recording platform and have to move from BiB Phone 
>> Preferred to Gateway Preferred. I know we have many phones already 
>> (erroneously) set to gateway preferred. The way we are set up, this isn't 
>> causing an issue but will in the future. I need to find out how many phones 
>> we have set this way. I can't find a way to search on this setting, so I 
>> turned to SQL and AXL.
>> 
>> 
>> When working with the AXL API, I can use a getPhone call and pull down the 
>> appropriate config. I can also use an updatePhone call to change the 
>> recording media source from "Phone Preferred" to "Gateway Preferred" and 
>> recordingFlag to and from "Automatic Call Recording Enabled" and "Selective 
>> Call Recording Enabled." When I look at the numplan, device and 
>> devicenumplanmap, I don't see where the recording media source or 
>> recordingFlag are stored.
>> 
>> 
>> I've tried pulling down the rows of the previously mentioned tables via a 
>> SQL select, then changing the settings in CUCM and pulling the same tables 
>> down to see if anything changed. Nothing changes so I think it has to be 
>> storing somewhere else or I'm doing something wrong.
>> 
>> 
>> Does anyone have either a SQL query I can look at that shows how this is 
>> stored, or a better understanding of how this parts works? Maybe there is a 
>> way to do this in CUCM that I'm unaware of?
>> 
>> Thanks in advance,
>> Nick
>> ___
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>> cisco-voip@puck.nether.net
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Thanks,
Nick
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[cisco-voip] Where does CUCM/informix store the recordingMediaSource and recordingFlag for phones/lines?

[Nick Barnett]

We're changing our recording platform and have to move from BiB Phone Preferred 
to Gateway Preferred. I know we have many phones already (erroneously) set to 
gateway preferred. The way we are set up, this isn't causing an issue but will 
in the future. I need to find out how many phones we have set this way. I can't 
find a way to search on this setting, so I turned to SQL and AXL.


When working with the AXL API, I can use a getPhone call and pull down the 
appropriate config. I can also use an updatePhone call to change the recording 
media source from "Phone Preferred" to "Gateway Preferred" and recordingFlag to 
and from "Automatic Call Recording Enabled" and "Selective Call Recording 
Enabled." When I look at the numplan, device and devicenumplanmap, I don't see 
where the recording media source or recordingFlag are stored.


I've tried pulling down the rows of the previously mentioned tables via a SQL 
select, then changing the settings in CUCM and pulling the same tables down to 
see if anything changed. Nothing changes so I think it has to be storing 
somewhere else or I'm doing something wrong.


Does anyone have either a SQL query I can look at that shows how this is 
stored, or a better understanding of how this parts works? Maybe there is a way 
to do this in CUCM that I'm unaware of?

Thanks in advance,
Nick___
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[cisco-voip] Automated cluster building? (devops tools)

[Nick Barnett]

I'm looking into what we need in a non-prod environment and I can't seem to 
find what I'm looking for. Has anyone used standard devops tools to automate 
the build of clusters? I doubt I'm going to find anything that would be an 
"IaaS for CUCM" that I can run at my company, but that's what I'm after.


What are other people doing for the problem of needing repeatable and quick 
development environments? There has to be a better way than building a new 
cluster from scratch, which can take hours. I can hit a button and deploy a 
full cluster of linux servers, all of the containers they need, DNS, the 
virtual network to connect them all, and everything else needed to be 
production ready (and it will be done in a few minutes), but I have to spend 
DAYS making a dev CUCM environment. What am I missing?


Thanks,
Nick


Thanks,
Nick
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Re: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Nick Barnett]

We think it is happening frequently WITHOUT this command being ran. Weird
stuff happens... like deleting a speed dial and it never goes away... or
changing the distribution order on a route list that auotmatically reverts
back after a few seconds... or maybe the GUI shows it never reverted back
however it is clearly not performing the correct algo. I can duplicate the
RTT issue by raising the packet size to 1200 and doing a repeat 100
packets. it WILL give me times over 80ms. BUT, the SDL traffic is supposed
to be QOS in a certain way and I'm sure that the pings I'm doing are NOT
being classified and queued properly. It is very frustrating that I know
what I'm talking (enough to discuss with them, but it has been 7 years
since I was 100% router jockey) about and can't get them to pay attention
to a probable network issue.

I have an IP SLA running that shows average latency in the 20ms range. IP
SLA is a fake red herring if you ask me... it only looks at an AVERAGE
every 5 minutes and if there are no issues, of course it will look great.

Thanks,
Nick

On Tue, Nov 6, 2018 at 12:42 PM Ryan Huff  wrote:

> You are able to correlate the out-of-band RTT to only when the
> dbreplication stat command is ran, or are there other times the RTT is OOB
> that isn't related to querying the replication status?
>
>
> Thanks,
>
> -R
> --
> *From:* cisco-voip  on behalf of Nick
> Barnett 
> *Sent:* Tuesday, November 6, 2018 11:57 AM
> *To:* Cisco VoIP Group
> *Subject:* [cisco-voip] WAN Delays > 80ms for CUCM cluster?
>
> We all know the max latency is 80ms, but ours occasionally goes over. I'm
> trying to track down why but the network team cannot find an issue. We are
> able to reproduce the issue repeatedly by running "utils dbreplication
> runtimestate." Whether this is causing the issue (I doubt it) or that
> command just takes long enough to run that it will eventually find a time
> that is > 80ms (my guess Is yes)... I'm not 100% sure.
>
> We opened a case with TAC to find out what that command is actually doing,
> but they won't divulge the info that our network team needs.
>
> My theory is that it's actually calling some shell script in redhat under
> the CLI appliance layer. Has anyone investigated that? Do we know what this
> command is actually doing? Specifically, i want to know where it's getting
> those ping times... is it running a generic ping with generic datagram
> data? Is it sending a 1497 packet of 0x and then 0x? Basically, I'm
> trying to give the network team something to go on because they are saying
> it's not them. (Of course they could run a packet capture and tell me
> (mostly) what it's doing, but it's hard to get their attention when they
> don't think it's on their end).
>
> Thanks,
> Nick
>
> P.S.  We have frequent DB replication issues... at least a few times per
> quarter. This is so annoying and I'm pretty sure it's due to this latency,
> but I can't get anyone to pay attention.
>
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Re: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Nick Barnett]

Yes, I agree, this is a super common "discussion" between app and network
teams... I'm a converted network engineer (like I bet many people are these
days)... so know all the tricks to push it back on the app :)

On Tue, Nov 6, 2018 at 12:25 PM Wes Sisk (wsisk)  wrote:

> Nick,
>
> The command is invoking database commands that Cisco does not own. They
> are not being obtuse; they genuinely do not know.
>
> It will cause a spike in database communication between nodes.
>
> My first guess is very much in line with yours that the burst in traffic
> exceeds certain QoS queues.
>
> IMHO - and I emphasize the MY in that - this a rather classic discussion
> point between application teams and network teams.
>
> What Matt suggests in a subsequent response is the the rather data
> intensive way of getting that information. Fortunately wireshark has graphs
> for round trip time.
>
> -Wes
>
> On Nov 6, 2018, at 11:57 AM, Nick Barnett  wrote:
>
> We all know the max latency is 80ms, but ours occasionally goes over. I'm
> trying to track down why but the network team cannot find an issue. We are
> able to reproduce the issue repeatedly by running "utils dbreplication
> runtimestate." Whether this is causing the issue (I doubt it) or that
> command just takes long enough to run that it will eventually find a time
> that is > 80ms (my guess Is yes)... I'm not 100% sure.
>
> We opened a case with TAC to find out what that command is actually doing,
> but they won't divulge the info that our network team needs.
>
> My theory is that it's actually calling some shell script in redhat under
> the CLI appliance layer. Has anyone investigated that? Do we know what this
> command is actually doing? Specifically, i want to know where it's getting
> those ping times... is it running a generic ping with generic datagram
> data? Is it sending a 1497 packet of 0x and then 0x? Basically, I'm
> trying to give the network team something to go on because they are saying
> it's not them. (Of course they could run a packet capture and tell me
> (mostly) what it's doing, but it's hard to get their attention when they
> don't think it's on their end).
>
> Thanks,
> Nick
>
> P.S.  We have frequent DB replication issues... at least a few times per
> quarter. This is so annoying and I'm pretty sure it's due to this latency,
> but I can't get anyone to pay attention.
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>
>
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Re: [cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Nick Barnett]

Not a bad idea, but they have so much many more tools to do this. I'll keep
this in mind though. Thanks.

On Tue, Nov 6, 2018 at 11:06 AM Matt Jacobson 
wrote:

> You could use the CLI packet capture with some filters to maximize the
> capture window, run the dbreplication command once or twice, and then stop
> the capture. Pop open RTMT, download the capture(s), and then see what you
> find in Wireshark.
>
> On Tue, Nov 6, 2018 at 20:58 Nick Barnett  wrote:
>
>> We all know the max latency is 80ms, but ours occasionally goes over. I'm
>> trying to track down why but the network team cannot find an issue. We are
>> able to reproduce the issue repeatedly by running "utils dbreplication
>> runtimestate." Whether this is causing the issue (I doubt it) or that
>> command just takes long enough to run that it will eventually find a time
>> that is > 80ms (my guess Is yes)... I'm not 100% sure.
>>
>> We opened a case with TAC to find out what that command is actually
>> doing, but they won't divulge the info that our network team needs.
>>
>> My theory is that it's actually calling some shell script in redhat under
>> the CLI appliance layer. Has anyone investigated that? Do we know what this
>> command is actually doing? Specifically, i want to know where it's getting
>> those ping times... is it running a generic ping with generic datagram
>> data? Is it sending a 1497 packet of 0x and then 0x? Basically, I'm
>> trying to give the network team something to go on because they are saying
>> it's not them. (Of course they could run a packet capture and tell me
>> (mostly) what it's doing, but it's hard to get their attention when they
>> don't think it's on their end).
>>
>> Thanks,
>> Nick
>>
>> P.S.  We have frequent DB replication issues... at least a few times per
>> quarter. This is so annoying and I'm pretty sure it's due to this latency,
>> but I can't get anyone to pay attention.
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[cisco-voip] WAN Delays > 80ms for CUCM cluster?

[Nick Barnett]

We all know the max latency is 80ms, but ours occasionally goes over. I'm
trying to track down why but the network team cannot find an issue. We are
able to reproduce the issue repeatedly by running "utils dbreplication
runtimestate." Whether this is causing the issue (I doubt it) or that
command just takes long enough to run that it will eventually find a time
that is > 80ms (my guess Is yes)... I'm not 100% sure.

We opened a case with TAC to find out what that command is actually doing,
but they won't divulge the info that our network team needs.

My theory is that it's actually calling some shell script in redhat under
the CLI appliance layer. Has anyone investigated that? Do we know what this
command is actually doing? Specifically, i want to know where it's getting
those ping times... is it running a generic ping with generic datagram
data? Is it sending a 1497 packet of 0x and then 0x? Basically, I'm
trying to give the network team something to go on because they are saying
it's not them. (Of course they could run a packet capture and tell me
(mostly) what it's doing, but it's hard to get their attention when they
don't think it's on their end).

Thanks,
Nick

P.S.  We have frequent DB replication issues... at least a few times per
quarter. This is so annoying and I'm pretty sure it's due to this latency,
but I can't get anyone to pay attention.
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Re: [cisco-voip] SIP OPTIONS pings are blocked on Cisco CUBE 3945E - Resource failure, dropping OPTIONS

[Nick Barnett]

fo/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 0
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 8
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 9
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 4
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 2
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 15
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/10240/sipSPIGetCallConfig: RTP 
>>>>>>> Preferred
>>>>>>> Codecs supported by GW 255
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/critical/32768/ccsip_ipip_media_forking_update_preferred_codec:
>>>>>>> MF: Not a Forked SIP leg..
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/verbose/1/ccsip_set_srtp_config: No Srtp
>>>>>>> configure for this leg.
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/verbose/12288/sipSPIGetModemInfoPerCall:
>>>>>>> peer_callID=0
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Error/ccsip_ipip_media_forking_anchor_leg_config:
>>>>>>>
>>>>>>>
>>>>>>>  MF: *Dial-peer is absent*..
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/36864/sipSPIMFChangeState: MF: Prev 
>>>>>>> state
>>>>>>> = 0 & New state = -1
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Info/info/32768/ccsip_ipip_media_forking_anchor_leg_reset:
>>>>>>> MF: Anchor leg config reset done...
>>>>>>>
>>>>>>> Oct  9 17:50:04.068:
>>>>>>> //3652/95FFAA748E45/SIP/Error/ccsip_ipip_media_forking_intra_frame_request_config:
>>>>>>>
>>>>>>>
>>>>>>>  MF:video profile Dial-peer is absent..
>>>>>>>
>>>>>>>
>>>>>>> OPTIONS looks like following:
>>>>>>>
>>>>>>> OPTIONS sip:domain.name.here:5060 SIP/2.0
>>>>>>>
>>>>>>> From: ;tag=4a6000292f6a
>>>>>>>
>>>>>>> To: 
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I do not have any script in use there, the configuration in pretty
>>>>>>> basic.
>>>>>>> What i have found is that second OPTIONS (the one that is
>>>>>>> left/dropped without OK) also generates following output:
>>>>>>>
>>>>>>> Oct  9 17:50:38.070:
>>>>>>> //-1//SIP/Info/verbose/4096/ccsip_new_msg_preprocessor:
>>>>>>> Checking Invite Dialog
>>>>>>>
>>>>>>> Oct  9 17:50:38.070:
>>>>>>> //3653/9862338A8E46/SIP/Info/verbose/4096/sipSPIFindCcbUASReqTable:
>>>>>>> *CCB found in UAS Request table. ccb=0x2766B958
>>>>>>>
>>>>>>> Oct  9 17:50:38.070:
>>>>>>> //3653/9862338A8E46/SIP/Info/info/4096/sipSPICheckFromToRequest

Re: [cisco-voip] SIP OPTIONS pings are blocked on Cisco CUBE 3945E - Resource failure, dropping OPTIONS

[Nick Barnett]

Are you using Customer Call Back? Which dial peer is the options ping
hitting? Does that dial peer have the CCB script on it? If so... maybe make
another dial peer for options pings that does not have the script enabled.
This is just a hunch...

On Tue, Oct 9, 2018 at 6:50 AM Maciej Bylica  wrote:

> Hi
>
> I have really strange problem with SIP OPTIONS pings.
> The Cisco i have (CUBE 3945 ios Version 15.3(3)M5) responds only to the
> first OPTIONS packet that is received from the endpoint.
> The rest of OPTIONs are dropped with following debug output:
>
> Oct  9 12:52:06 10.10.10.10 8694907: Oct  9 10:55:58.194:
> //-1//SIP/Event/sipSPIEventInfo: Queued event from SIP SPI :
> SIPSPI_EV_CC_OPTIONS_RESP
> Oct  9 12:52:06 10.10.10.10 8694908: Oct  9 10:55:58.194:
> //148025/BCB3C79A92C0/SIP/Info/info/4096/sact_idle_new_message_options:
> ccsip_api_options_ind returned: SIP_SUCCESS
> Oct  9 12:52:06 10.10.10.10 8694909: Oct  9 10:55:58.194:
> //148025/BCB3C79A92C0/SIP/State/sipSPIChangeState: 0x258D7210 : State
> change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OPTIONS_WAIT,
> SUBSTATE_NONE)
> Oct  9 12:52:06 10.10.10.10 8694910: Oct  9 10:55:58.194:
> //148025/BCB3C79A92C0/SIP/Error/sipSPIUaddCcbToTable:
> Oct  9 12:52:06 10.10.10.10 8694911:  *Could not add ccb to table*.
> ccb=0x258D7210
> key=c3c4f5582a4bfa1ff4b7e741c3cb6c6822f738b4cd7e78633fc70f5441197d3
> Oct  9 12:52:06 10.10.10.10 8694912: Oct  9 10:55:58.194:
> //148025/BCB3C79A92C0/SIP/Error/sact_idle_new_message_options:
> Oct  9 12:52:06 10.10.10.10 8694913:  *Resource failure, dropping OPTIONS*
>
> The true is that Cisco receives quite significant amount of SIP OPTIONs
> from the endpoint in short time, like 10 OPTIONS packeges within
> miliseconds.
> The after-effect i want to achieve is a response for each incoming OPTIONS
>
> Example of a successful response:
> Oct  9 11:30:37 10.10.10.10 8625106: Oct  9 09:34:28.569:
> //-1//SIP/Event/sipSPIEventInfo: Queued event from SIP SPI :
> SIPSPI_EV_CC_OPTIONS_RESP
> Oct  9 11:30:37 10.10.10.10 8625107: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Info/info/4096/sact_idle_new_message_options:
> ccsip_api_options_ind returned: SIP_SUCCESS
> Oct  9 11:30:37 10.10.10.10 8625108: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/State/sipSPIChangeState: 0x258B1110 : State
> change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OPTIONS_WAIT,
> SUBSTATE_NONE)
> Oct  9 11:30:37 10.10.10.10 8625109: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Info/verbose/4096/sipSPIUaddCcbToTable: Added to
> table. ccb=0x258B1110
> key=c3c4f5582a4bfa1ff4b7e741c3cb6c6822f738b4cd7e78633fc70f5441197d3 balance
> 1
> Oct  9 11:30:37 10.10.10.10 8625110: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Info/verbose/4096/sipSPIUaddccCallIdToTable:
> Adding call id 23DA9 to table
> Oct  9 11:30:37 10.10.10.10 8625111: Oct  9 09:34:28.569:
> //-1//SIP/Info/info/4096/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 3 for event 38
> Oct  9 11:30:37 10.10.10.10 8625112: Oct  9 09:34:28.569:
> //-1//SIP/Info/info/1024/httpish_msg_create: created
> msg=0x203FFDA4 with refCount = 1
> Oct  9 11:30:37 10.10.10.10 8625113: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Info/info/4096/sipSPISendOptionsResponse:
> Associated container=0x2673A528 to Options Response
> Oct  9 11:30:37 10.10.10.10 8625114: Oct  9 09:34:28.569:
> //-1//SIP/Info/verbose/8192/sipSPIAppHandleContainerBody:
> sipSPIAppHandleContainerBody len 164
> Oct  9 11:30:37 10.10.10.10 8625115: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Transport/sipSPITransportSendMessage:
> msg=0x203FFDA4, addr=11.11.11.11, port=5060, sentBy_port=5060, local_addr=,
> is_req=0, transport=1, switch=0, callBack=0x4F48054
> Oct  9 11:30:37 10.10.10.10 8625116: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Info/info/2048/sipSPIGetExtensionCfg: SIP
> extension config:1, check sys cfg:1
> Oct  9 11:30:37 10.10.10.10 8625117: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Transport/sipSPITransportSendMessage: Proceedable
> for sending msg immediately
> Oct  9 11:30:37 10.10.10.10 8625118: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Transport/sipTransportLogicSendMsg: Trying to
> send resp=0x203FFDA4 to default port=5060
> Oct  9 11:30:37 10.10.10.10 8625119: Oct  9 09:34:28.569:
> //-1//SIP/Transport/sipConnectionManagerGetConnection:
> connection required for raddr:11.11.11.11, rport:5060 with laddr:
> Oct  9 11:30:37 10.10.10.10 8625120: Oct  9 09:34:28.569:
> //-1//SIP/Transport/sipInstanceGetConnectionId: Registering
> gcb=0x258B1110 with connection=0x2426673C context list
> Oct  9 11:30:37 10.10.10.10 8625121: Oct  9 09:34:28.569:
> //146857/5A42A0608E30/SIP/Transport/sipTransportLogicSendMsg: Connection
> obtained...sending msg=0x203FFDA4
> Oct  9 11:30:37 10.10.10.10 8625122: Oct  9 09:34:28.569:
> //-1//SIP/Transport/sipTransportPostSendMessage: 

Re: [cisco-voip] phones working on a dumb switch with voice/data vlans switched properly

[Nick Barnett]

It’d be interesting to see what a wireshark trace looks like. I think Dave
is spot on. If you span the PC port on the phone then run a capture, you
SHOULD probably be able to see the VLAN tag for those phone packets. I
would expect the packets destined for the laptop behind the phone to be
untagged. You could probably see both in a single trace. Everything seems
to be working, it's just fun to dig in sometimes.

On Wed, Oct 3, 2018 at 7:59 PM Lelio Fulgenzi  wrote:

>
> Well, I certainly learned something today. Thx for the feedback.
>
> *-sent from mobile device-*
>
>
> *Lelio Fulgenzi, B.A.* | Senior Analyst
>
> Computing and Communications Services | University of Guelph
>
> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
> N1G 2W1
>
> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>
>
>
> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>
>
>
> [image: University of Guelph Cornerstone with Improve Life tagline]
>
> On Oct 3, 2018, at 8:23 PM, Dave Goodwin 
> wrote:
>
> Not necessarily. Unmanaged switches that do not support 802.1q will often
> just forward frames as is, tags and all, and only operate on the other
> parts of the L2 header they pay attention to (source/destination address).
> I have also seen unmanaged switches that did not claim support for 802.1q
> and ATE my tags.
>
> On Wed, Oct 3, 2018 at 6:16 PM Lelio Fulgenzi  wrote:
>
>>
>> Ok. I hear ya. But wouldn’t the switch need to support 802.1q to support
>> those tagged packets?
>>
>> This switch didn’t specify that it supported 802.1q.
>>
>>
>>
>> *-sent from mobile device-*
>>
>>
>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>
>> Computing and Communications Services | University of Guelph
>>
>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>> N1G 2W1
>>
>> 519-824-4120 Ext. 56354 <519-824-4120;56354> | le...@uoguelph.ca
>>
>>
>>
>> www.uoguelph.ca/ccs | @UofGCCS on Instagram, Twitter and Facebook
>>
>>
>>
>> [image: University of Guelph Cornerstone with Improve Life tagline]
>>
>> On Oct 3, 2018, at 5:47 PM, Dave Goodwin 
>> wrote:
>>
>> I haven't really tried the kind of thing you described in quite a long
>> time. But I assume the reason the phone gets an IP address is because the
>> CDP VVLAN information is being successfully sent through the unmanaged
>> non-Cisco switch so the phone actually sees it. The phone learns its VVLAN
>> ID, so starts tagging all its frames and therefore gets an IP from the
>> VVLAN. The other devices plugged into the switch (or behind the phone)
>> don't normally need to tag, so the frames are coming through up to the
>> Cisco switch untagged and are therefore in the native/access VLAN on the
>> port and getting one of those IPs.
>>
>> So I guess that unmanaged switch passes any CDP through to all ports, and
>> also passes any tagged traffic through unaltered.
>>
>> On Wed, Oct 3, 2018 at 5:17 PM Lelio Fulgenzi  wrote:
>>
>>>
>>> OK - I'm probably showing my ignorance here, but I was quite surprised
>>> to find out that plugging in a dumb Dlink DES-105 switch into our cisco
>>> switch with access layer programming, so data vlan and voice vlan, extended
>>> things such that when a phone is plugged in, it got an IP address on the
>>> voice vlan, and plugging a non-phone device got an ip address on the data
>>> vlan and then plugging a similar device into the back of the phone also got
>>> an ip address on the data vlan. We plugged in multiple phones as well. All
>>> worked fine. *phones powered by brick*
>>>
>>> I can appreciate a passthrough device, I've used them before as ethernet
>>> extenders. By what I'm not understanding is how traffic is being classed
>>> properly through to this dumb switch.
>>>
>>> We're using a new 9300 series switch, but I'm not sure that would make a
>>> difference.
>>>
>>>
>>>
>>> ---
>>> 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] FirstName Lastname in delivered CUC Unified Messaging voicemails

[Nick Barnett]

 When a call comes into Albert Jones CUC vmailbox from Barb Smith's mobile
phone, an email is delivered to Albert Jones email mailbox with subject
line "VM from Smith, Barb 15553219876"  I'm 99.9% sure that this name info
is just coming from CNAM info via the PSTN.

When looking in CUC, System Settings, Subject Line Formats, you can change
the basic layout of the subject lines... it looks like there is a %NAME%
variable, but nothing about firstname or lastname. I was just looking for a
way to make everything match...

I'm not sure why it was decided that we're going to the new "industry
standard" of Firstname Lastname, but it apparently is a big deal that
everything follow that new rule. It doesn't look possible for this piece.

Thanks,
Nick


On Tue, Jun 5, 2018 at 1:23 PM, Sadelski, Richard <
richard.sadel...@presencehealth.org> wrote:

>
>
>
>
> Have you tried changing the Display Name field for the users voicemail
> box?  Not sure if that is what you are looking to change.
>
>
>
> Thanks
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Nick Barnett
> *Sent:* Tuesday, June 05, 2018 12:15 PM
> *To:* Cisco VoIP Group 
> *Subject:* [cisco-voip] FirstName Lastname in delivered CUC Unified
> Messaging voicemails
>
>
>
> Our email team recently change the display of names in MSFT land to be
> "Firstname Lastname" instead of the standard "Lastname, Firstname" that
> everyone is familiar with. CUC's Unified Messaging sends the notifications
> out as "Last, First"
>
>
>
> I've been looking through docs and can't seem to find a way to change this
> in CUC.
>
>
>
> Has anyone ever faced this or maybe have some pointers on where to poke
> around?
>
>
>
> Thanks,
>
> Nick
>
>
>
> p.s. - we have a mix of people with Viewmail and those with just Unified
> Messaging, but the same thing is happening for both... which makes sense
> since CUC has no idea if the user's client has viewmail or not.
>
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[cisco-voip] FirstName Lastname in delivered CUC Unified Messaging voicemails

[Nick Barnett]

Our email team recently change the display of names in MSFT land to be
"Firstname Lastname" instead of the standard "Lastname, Firstname" that
everyone is familiar with. CUC's Unified Messaging sends the notifications
out as "Last, First"

I've been looking through docs and can't seem to find a way to change this
in CUC.

Has anyone ever faced this or maybe have some pointers on where to poke
around?

Thanks,
Nick

p.s. - we have a mix of people with Viewmail and those with just Unified
Messaging, but the same thing is happening for both... which makes sense
since CUC has no idea if the user's client has viewmail or not.
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Re: [cisco-voip] CUC and Exchange Online migration

[Nick Barnett]

One gotcha is the user smtp address in cuc needs to be publicly reachable
if you plan to add the required alias for view mail. If not, ExO will not
allow you to use the email address.

On Tue, Mar 27, 2018, 2:22 AM James Andrewartha via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> Hi voipers,
>
> We're about to move all voicemail from Exchange 2013 on-prem to CUC, and
> then migrate to Exchange Online. We have the Unified Messaging from CUC
> to Exchange 2013 working fine, I found one post that said that things
> will just work(tm) when we start moving users to Exchange Online, is
> that correct or would I need to set up a separate Unified Messaging
> Service for it?
>
> Also is there an SQL query I can run in CUCM to determine which DNs have
> a custom Voice Mail Profile set?
>
> Thanks,
>
> --
> James Andrewartha
> Network & Projects Engineer
> Christ Church Grammar School
> Claremont, Western Australia
> Ph. (08) 9442 1757
> Mob. 0424 160 877
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[cisco-voip] Cucilync keeps locking people's AD accounts

[Nick Barnett]

Years ago when cucilync 8.X was deployed, there was a registry setting that
made a popup window appear when the user changed their Windows password. I
can't find this setting for CuciLync 11.6.  Does anyone know if it exists?
Is it an option in jabber-config.xml that I'm missing somehow? Any help is
appreciated.

Thanks,
Nick
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Re: [cisco-voip] Automated PSTN ingress call regression testing?

[Nick Barnett]

Thanks. I am aware of the multiple carriers... but I think being able to
cover the edge services would be a huge help in this situation. I'm just
anticipating what they will ask for. I kind of cast a wide net with this
email because I was not sure what kind of services were out there. If some
provider offered a prepackaged, automated testing service that featured
multiple carrier numbers, I'd buy it in an instant. I just need to remember
baby steps :)

By "prone to issues", I just mean that it will test connected calls, but
getting that next layer of "it connected, but is it working" would be
difficult. Not necessarily "issues". I suppose "obstacles" would have been
a better word.  The issues I see with the freePBX would be similar, but
also includes perimeter security and things of that nature.

I don't have UCCX, but I'm fairly ok at cobbling together AXL and JTAPI to
do some stuff... maybe I'll just start there since it's basically free.

On Wed, Mar 7, 2018 at 9:33 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Even if you do the Free IP PBX or Twilio API, you're only calling from one
> carrier.  In the scenario you described, you mentioned:
>
> "Verizon wireless customers cannot call Sprint toll free numbers from area
> code 555"
>
> Which is very specific.  Would you imagine that you would have owned a 555
> number on Verizon to have caught that scenario faster?  What if the area
> code was 666?  Or the originating carrier was AT?  The different
> combinations you would have to account for are very high.
>
> If you only care about your edge service and inward, and not far end
> carriers, then a Twilio API app sounds like a good plan.  Heck, you could
> even just write a UCCX script to call out and back in via tromboning off
> the PSTN.
>
> I'm curious, what did you mean by "prone to issues," when referring to the
> API?
>
> On Wed, Mar 7, 2018 at 1:57 PM Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> A client has a need for an off site solution that will make test calls to
>> their numbers and report when there are issues. I understand that this is
>> very vague, but they are interested in hearing about any and all solutions.
>>
>> They have several SIP carriers and a nationwide presence, but the SIP
>> trunking is centralized. They've had enough issues with one DID service
>> failing and their customers having to report the issue. Ideally, the SIP
>> providers would be able to automatically do "something" when they stop
>> receiving options pings, or when a certain sip response is received... but
>> it doesn't work that way with the behemoth phone companies.
>>
>> The way it works now is that MOST issues are able to be caught
>> successfully with internal monitoring... but others such certain NPA-NXX
>> can't call another NPA-NXX, or carrier interconnects such as "Verizon
>> wireless customers cannot call Sprint toll free numbers from area code
>> 555"  These odd scenarios are what we are looking to solve. I understand
>> this is potentially huge, but I think if we could automate calls to about
>> 10 different numbers, that would cover enough of the ingress and carrier
>> combinations that it would make a HUGE difference.
>>
>> I've thought of spinning up an Asterisk and somehow automating the echo
>> test feature. I've also thought about using the Twilio API to test if calls
>> are successful. Both of these are complicated and prone to issues... so if
>> there is a hosted or cloud solution that is already available, please let
>> me know.
>>
>> Any suggestions or more than welcome.
>>
>> Thanks,
>> Nick
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[cisco-voip] Automated PSTN ingress call regression testing?

[Nick Barnett]

A client has a need for an off site solution that will make test calls to
their numbers and report when there are issues. I understand that this is
very vague, but they are interested in hearing about any and all solutions.

They have several SIP carriers and a nationwide presence, but the SIP
trunking is centralized. They've had enough issues with one DID service
failing and their customers having to report the issue. Ideally, the SIP
providers would be able to automatically do "something" when they stop
receiving options pings, or when a certain sip response is received... but
it doesn't work that way with the behemoth phone companies.

The way it works now is that MOST issues are able to be caught successfully
with internal monitoring... but others such certain NPA-NXX can't call
another NPA-NXX, or carrier interconnects such as "Verizon wireless
customers cannot call Sprint toll free numbers from area code 555"  These
odd scenarios are what we are looking to solve. I understand this is
potentially huge, but I think if we could automate calls to about 10
different numbers, that would cover enough of the ingress and carrier
combinations that it would make a HUGE difference.

I've thought of spinning up an Asterisk and somehow automating the echo
test feature. I've also thought about using the Twilio API to test if calls
are successful. Both of these are complicated and prone to issues... so if
there is a hosted or cloud solution that is already available, please let
me know.

Any suggestions or more than welcome.

Thanks,
Nick
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[cisco-voip] AXL or SQL to run a CTI Search like RTMT?

[Nick Barnett]

We consistently have issues around here where contact center users receive
error 10159 "This is a shared line"  This happens because our contact
center uses extension mobility. To remedy the error, you have to launch
RTMT, go to Voice/Video, CTI then CTI Search. At that point, you put the
users number in the search box and hit Finish. It comes back with a table
of devices using that number. From there, you have to go to the application
user and disassociate all (usually just 2) of the devices returned from the
RTMT search.

Is RTMT the only place that has this info? I just went through the AXL
toolkit and didn't see anything that looked promising. Any ideas on if this
info can be pulled down from a SQL query?

I'm working on automating this whole process because that's the new thing
everyone likes to do :)  I already have the code that will remove the
appropriate device from the app user and reassociate it successfully. I
just need to know which devices need to be removed and added back, and the
only place I can find that info is RTMT.

Thanks,
Nick
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Re: [cisco-voip] CUCM 10.5 and Office 365

[Nick Barnett]

I'd also add when comparing Exchange/UM to CUC with Cisco/UM, caller input
can get really complicated if you use non-DIDs inside your company
(especially since the SBC connection solutions are EOL). We have a "global"
opt out of "press 0 to speak to blah blah blah operator group", but the
actual number dialed hits CVP/ICM and routes around with non-PSTN numbers.
We also have some complex ToD routing done through CUC using the same
non-DID type numbers (hitting hunt pilots, CTI route points, etc). We'd
have to redesign our entire dial plan to accomodate ExO/UM.

On Wed, Sep 27, 2017 at 2:45 AM, Nathan Reeves 
wrote:

> In situations where a customer has gone O365 (and they've got licenses for
> Unity Connection Via CUWL Standard or Basic Messaging Licenses), I'd be
> using Unity for Voicemail and AA's, and using the Single Inbox
> functionality of Unity to get Voicemail's synced into the Users Exchange
> Mailbox.  Note that Single inbox is still available post 1 July 2018, it's
> only the SBC Functionality which is being deprecated.
>
> The only thing from memory you lose with Unity Connection Vs Exchange UM
> is missed call notifications.  Only issue I suppose is the additional tasks
> required to manage the unity mailboxes vs Checking the box in AD for
> Exchange.
>
> Nathan
>
> On Wed, Sep 27, 2017 at 2:47 PM, James Andrewartha <
> jandrewar...@ccgs.wa.edu.au> wrote:
>
>> On 27/09/17 13:12, Nathan Reeves wrote:
>> > To be honest it was the best thing to happen.  After inheriting a
>> > customer setup where they were using AA's on O365, with Cube acting as
>> > the SBC, we saw endless issues with calls being answered but no audio
>> > heard.  Wouldn't work for a varied length of time only to just start
>> > working again.  Managed to move them over to Unity Call Handlers and
>> > we've not skipped a beat since.  Like you, being in AU, we also saw
>> > latency through to the O365 tenancy, though I'm suspecting the issues we
>> > saw were not latency related.
>>
>> Hmm. AAs are less of an issue, we can leave them on the on-premises
>> Exchange server that a hybrid configuration leaves behind for
>> management. Voicemail is the main thing, at the moment it seems our
>> choices are go to Exchange 2016 on-premises or switch to Unity
>> Connection (we already have CUWL Standard for some reason) then go to
>> Exchange Online. Is there a third option?
>>
>> --
>> James Andrewartha
>> Network & Projects Engineer
>> Christ Church Grammar School
>> Claremont, Western Australia
>> Ph. (08) 9442 1757
>> Mob. 0424 160 877
>>
>
>
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Re: [cisco-voip] Changing CUC UM Account with API?

[Nick Barnett]

Anthony, that's what I experienced too, just an endless loop of not being
able to find what I was looking for.
Nathan, you have WAY more patience than I do right now. Thank you!

On Sun, Oct 1, 2017 at 1:35 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> Oh man, you just solved my major problem with Postman desktop app.  I was
> getting the domain not allowed error, and just quit after that. I have not
> experienced that before, but now I know.  Thanks!
>
> On Sun, Oct 1, 2017 at 12:25 AM Nathan Reeves <nathan.a.ree...@gmail.com>
> wrote:
>
>> Taken a quick look and yeah, Anthony is correct in regards to the use of
>> DELETE.
>>
>> *To Delete the existing UM Service account:*
>> - A GET to https:///vmrest/users/ will dump the users
>> on the Connection Server.
>> - For each user you'll find an objectId listed. A GET to https://
>> /vmrest/users//externalserviceaccounts
>> will dump any attached external services account for the user.
>> - You'll want to then grab the '' value in the returned data
>> (looking like 
>> '/vmrest/users//externalserviceaccounts/'
>> and execute a DELETE against that URI (obv adding 
>> 'https:///'
>> to the URI).
>> - Note that you need to use basic auth to login as an appadmin for the
>> request.
>> - CURL wise: curl --basic --user : -k -X DELETE
>> https:///vmrest/users//
>> externalserviceaccounts/
>>
>> *To add the new UM Service to the user account:*
>> - Update the XML below with:
>>- %umservice-objectid% - The guid of the um service (easiest way is to
>> grab the guid from the CUC Admin pages for the UM service listed under
>> Unified Messaging > Unified Messaging Services).
>>- %user-objectid% - As per the id you found prev when you did the
>> delete.
>>
>> 
>>  
>> *%umservice-objectid%*
>>  *%user-objectid%*
>>  /vmrest/users/*%user-objectid%*
>>  true
>>  false
>>  true
>>  false
>>  0
>>  
>>  true
>>  true
>> 
>>
>> You'll need to then POST this XML to url: 'https:///
>> vmrest/users//externalserviceaccounts'.  Note that again
>> you need to provide basic auth.
>>
>> Note that the XML Above worked for me when adding a UM Service configured
>> to point to Exchange, so ~in theory~ this should work for O365 as well.
>> Not got a tenancy I can test against at the moment in O365 to confirm.
>>
>> If you use Postman to run this stuff, make sure you use the desktop
>> version (not the chrome extension version), and make sure you add the
>> 'Origin' header.  The value should be the url of the CUC Server (eg '
>> http://172.20.2.25').  This should stop you receiving the 'domain not
>> allowed' error message'
>>
>> Hope this assists.
>>
>> Nathan
>>
>>
>> On Fri, Sep 29, 2017 at 7:56 AM, Nathan Reeves <nathan.a.ree...@gmail.com
>> > wrote:
>>
>>> While people are possibly playing around with the CUC Provisioning API,
>>> let me know if you ever get the import of CUCM Local users to CUC working
>>> correctly.  The last rollout I did I was trying to pull in local users but
>>> the api just didn't work.  Had to use LDAP imports instead.  Never looked
>>> too much deeper as I just needed to get on with things.
>>>
>>> But yeah, the API for CUC is really hit and miss.  lol, and the API for
>>> UCCX can be a bit the same.
>>>
>>> Nathan
>>>
>>> On Fri, Sep 29, 2017 at 5:07 AM, Anthony Holloway <
>>> avholloway+cisco-v...@gmail.com> wrote:
>>>
>>>> Wow, the documentation for the CUC API has gone to shit.  There's
>>>> literally a link on the developer.cisco.com site that sends you to a
>>>> wiki site, which itself then sends you to developer.cisco.com.  Nice.
>>>>
>>>> Anyway, I tried to look into this quick for you, but I got stuck with
>>>> the documentation on POSTing a new UM account for end users.  It literally
>>>> just says:
>>>>
>>>> "To create an a new external service account, POST an XML document 
>>>> (formatted
>>>> similar to the one above) to the following URL:
>>>> POST https:///vmrest/users//
>>>> externalserviceaccounts"
>>>> Source: http://docwiki.cisco.com/wiki/Cisco_Unity_
>>>> Connection_Provisioning_Interface_(CUPI)_API_--_User_
>>>> Unified_Messaging_Account
>>>>
>>>> The key part being "similar 

[cisco-voip] Changing CUC UM Account with API?

[Nick Barnett]

I can handle most things in CUCM with SOAP, but I always get confused when
trying to use VMREST in CUC. I cannot find a way to add and remove a UM
account via automation. We're stuck using a CSV and it's really putting a
cramp in our migration to Exchange Online.

The particular change I'm needing is here: Users->EditMenu->Unified
messaging accounts
1) Need to add an additional one that connects to Exchange Online (this
already exists in CUC)
2) Need to delete the old one that connects to on prem Exchange. (Not
delete the whole UM connector service, but just the account association to
a particular user)

Does anyone have any pointers on this? We have batches of people migrating
every week, sometimes multiple times per week... so I can't just make some
global change.

We're on CUC 10.5

Thanks!
Nick

P.S.  I hate that MSFT and CSCO both have a product called "Unified
Messaging" :)
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[cisco-voip] doing more with CUC notification devices?

[Nick Barnett]

One of my customers has asked for voicemail "logging."  What they mean by
that is that each time one of our sales people gets a voicemail in their
CUC, they want this added to an internal database. There is a lot more to
it, but this is the extreme basics of the request. My thoughts were to
create an addtional SMTP notification device and stand up another email
server. The 2nd SMTP device would be sent to something like
nick.barn...@internalserver.company.xyz   From there, they could have an
app that parsed the email and did whatever they wanted. That should
theoretically work, but it's nasty as it relies on email, and other
servers, and webapps and a bunch of other stuff that can break.

They basically want a push notification for each voicemail, but not to an
SMTP device.
What we really need is a webhook that is fired when someone gets a new
voicemail.

Is there any way to do this now? I'm on CUC 10.5... is anything like this
slated in a roadmap? Are there any 3rd party solutions for something like
this?

Thanks!
Nick
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[cisco-voip] Blocking IPs from the CUCM CLI?

[Nick Barnett]

I swear I've seen this before, but I can't find it. I might be mistaken,
but I was thinking you could firewall block IPs from the CUCM CLI. Am I
just dreaming?

Thanks,
Nick
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Re: [cisco-voip] e911, CER, wireless and soft phones?

[Nick Barnett]

ok, that's interesting for a whole lot of reasons right now. We are looking
at our next upgrade for next year and right now, we are a 100% cisco shop.

On Thu, Jun 1, 2017 at 12:28 AM, Ankur Srivastava via cisco-voip <
cisco-voip@puck.nether.net> wrote:

> In 11.5 Cisco has released a feature that helps you isolate the particular
> user, but this only works for Cisco WLC. I tried testing this with Aruba in
> our environment but cucm is not synching with it. I raised this with Cisco
> to open this feature up to other Third-party vendors. But I guess if more
> people raise the noise they will listen. All they are doing is sharing info
> over SNMP , but its currently programmed to not respond to any other box
> that doesn’t say Cisco
>
>
>
> Regards,
>
> Ankur
>
>
>
> *From: *cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of
> Lelio Fulgenzi <le...@uoguelph.ca>
> *Date: *Wednesday, May 31, 2017 at 4:53 PM
> *To: *Brian Meade <bmead...@vt.edu>
> *Cc: *voip puck <cisco-voip@puck.nether.net>
> *Subject: *Re: [cisco-voip] e911, CER, wireless and soft phones?
>
>
>
> Can any of these solutions do anything better than tell me what AP I'm on?
>
>
>
> And that includes telling me my location based on the location of the AP.
>
>
>
> I'd hope that there'd be some type of triangulation based on proximity to
> several APs. Or some sort of partnership with the BU that handles
> rfid/beacons.
>
>
>
>
>
>
>
> Sent from my iPhone
>
>
> On May 31, 2017, at 4:54 PM, Brian Meade <bmead...@vt.edu> wrote:
>
> Yea, but most people are deploying their entire campus wireless as a
> single /16 for better roaming.
>
>
>
> On Wed, May 31, 2017 at 4:46 PM, Ben Amick <bam...@humanarc.com> wrote:
>
> Doesn't CER have the ability to track by subnet?
>
> Ben Amick
> Telecom Analyst
>
>
> > On May 31, 2017, at 3:59 PM, Nick Barnett <nicksbarn...@gmail.com>
> wrote:
> >
> > I'm looking into options to properly track softphones with CER. Our CER
> is 100% Intrado right now, but we're looking at ditching hard phones and
> need to figure out tracking of softphones, especially when connected
> wirelessly.
> >
> > What are people using out there?
> >
> > Thanks,
> > Nick
>
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> qejqabbXaoVVZASyyO-Y-euvsdEEK6zAQsTLt6VIxGIH5gkjrlS
> 6NJOVICSHIdzrBPtPhPtdRcQsLZvAXKcff9ZuVtdBZDCm4QrIIth5dqWqJSk
> -l3PWApmU6CQjq9K_nhh7ec3HLCzATsSjDdqymoIToHMd9_
> 7wrwCHIcfBisEeROQGmGncRAIrJaBGBPdpb5O5mUm-wamrIlU6A_
> zMddI6zCVEVdCBKQGmGncRAIqnjh1VEVhh0cQg2gQ2M3d41ykvf-
> B3h1I43h1iHIPh0IjH3W4zr1EVdMMln8SHR
>
>
> Confidentiality Note: This message is intended for use only by the
> individual or entity to which it is addressed and may contain information
> that is privileged, confidential, and exempt from disclosure under
> applicable law. If the reader of this message is not the intended recipient
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> intended recipient, you are hereby notified that any dissemination,
> distribution or copying of this communication is strictly prohibited. If
> you have received this communication in error, please contact the sender
> immediately and destroy the material in its entirety, whether electronic or
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Re: [cisco-voip] e911, CER, wireless and soft phones?

[Nick Barnett]

Yep, we are wireless on /16, makes roaming a heck of a lot easier, but is
worthless for subnet e911 tracking.  We also have RedSky's cloud service.
I've reached out to them to try and get a demo of their warez.

On Wed, May 31, 2017 at 3:54 PM, Brian Meade <bmead...@vt.edu> wrote:

> Yea, but most people are deploying their entire campus wireless as a
> single /16 for better roaming.
>
> On Wed, May 31, 2017 at 4:46 PM, Ben Amick <bam...@humanarc.com> wrote:
>
>> Doesn't CER have the ability to track by subnet?
>>
>> Ben Amick
>> Telecom Analyst
>>
>> > On May 31, 2017, at 3:59 PM, Nick Barnett <nicksbarn...@gmail.com>
>> wrote:
>> >
>> > I'm looking into options to properly track softphones with CER. Our CER
>> is 100% Intrado right now, but we're looking at ditching hard phones and
>> need to figure out tracking of softphones, especially when connected
>> wirelessly.
>> >
>> > What are people using out there?
>> >
>> > Thanks,
>> > Nick
>> > ___
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>> > cisco-voip@puck.nether.net
>> > http://cp.mcafee.com/d/k-Kr6h0SyMqen4TbzDSjtPqabbXaoUsyqejqa
>> bbXaoVVZASyyO-Y-euvsdEEK6zAQsTLt6VIxGIH5gkjrlS6NJOVICSHIdzrB
>> PtPhPtdRcQsLZvAXKcff9ZuVtdBZDCm4QrIIth5dqWqJSk-
>> l3PWApmU6CQjq9K_nhh7ec3HLCzATsSjDdqymoIToHMd9_7wrwCHIcfBisEe
>> ROQGmGncRAIrJaBGBPdpb5O5mUm-wamrIlU6A_zMddI6zCVEVdCBKQGmGn
>> cRAIqnjh1VEVhh0cQg2gQ2M3d41ykvf-B3h1I43h1iHIPh0IjH3W4zr1EVdMMln8SHR
>>
>>
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>> intended recipient, you are hereby notified that any dissemination,
>> distribution or copying of this communication is strictly prohibited. If
>> you have received this communication in error, please contact the sender
>> immediately and destroy the material in its entirety, whether electronic or
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[cisco-voip] e911, CER, wireless and soft phones?

[Nick Barnett]

I'm looking into options to properly track softphones with CER. Our CER is
100% Intrado right now, but we're looking at ditching hard phones and need
to figure out tracking of softphones, especially when connected wirelessly.

What are people using out there?

Thanks,
Nick
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[cisco-voip] Automating association devices and profiles to PG_User application user?

[Nick Barnett]

We have automation that builds devices, EM_Profiles, DNs, and just about
everything else... except for the manual add of the controlled devices to
the pg user application user for UCCE. We also use Nice for recording and
there is a nice app user that needs these associations as well.

I have figured out how to do this by using a getAppUser AXL call, parsing
the returned data, inserting my NEW device/profile where appropriate (into
a new tag) and then submitting the information back as an updateAppUser. I
think this this the only method we have available to automating this
portion.

It kind of worries me to do it this way because I can see how the database
may see it as disassociating all devices from the PG User and then
re-associating all devices. Depending on processor utilization etc, I can
also see where they may be a short period of time where the PG user has no
associated devices.

Are my worries substantiated by any fact? Does anyone else do this? Are
there better ways to accomplish this task?

Thanks,
Nick
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Re: [cisco-voip] Execute sql query via vmrest on CUC (or alternatative approach)?

[Nick Barnett]

Thanks Anthony and Brian.  I think i can make this work, especially after
reading your firefox script and your non-code example... I think i can hack
this together with another project I made in python and probably get it to
work.

Thanks,
Nick

On Wed, Mar 8, 2017 at 3:40 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> I don't think the CUC API has an arbitrary SQL execution method call like
> AXL does.
>
> Since the SQL query is effectively searching everyone's mailbox message
> counts, and then just filtering the output to you, you could write that
> same process using the CUPI API.
>
> Here's a high level program flow in no particular actual language:
>
> results = Array()
> response = HTTP GET https:///vmrest/users
> for each user in response.users:
> userobjectid = response.ObjectId
> alias = response.Alias
> response = HTTP GET https:///vmrest/users/<
> userobjectid>/mailboxattributes
> count = response.NumMessages
> results.append(alias, count)
> results.sort(count, DESC)
> for each result in results:
> print result.alias, results.count
>
> Oh, and this is probably a good time to plug my Firefox GreaseMonkey User
> Script which shows you the breakdown of message counts per folder, and even
> let's you empty the deleted items.
>
> https://twitter.com/avholloway45633/status/828515885769953280
>
>
> On Wed, Mar 8, 2017 at 2:52 PM Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> I found this SQL query
>> <https://www.cisco.com/c/en/us/support/docs/unified-communications/unity-connection/118299-technote-cuc-00.html#anc8>
>> to return a count of all message boxes in CUC. I modified it to return the
>> top 10 by adding "FIRST 10" immediately after "select" on the first line:
>>
>> run cuc dbquery unitymbxdb1 select FIRST *10* alias as UserID, count (*)
>> as messages \
>>
>> from vw_message, unitydirdb:vw_mailbox, unitydirdb:vw_user \
>>
>> where mailboxobjectid in \
>>
>> (select mailboxid from vw_mailbox where unitydirdb: vw_user.objectid =
>> unitydirdb:vw_mailbox.userobjectid) \
>>
>> group by alias order by \
>>
>> messages desc
>>
>>
>> This works, but it's not very "dev ops friendly." I think I'd have to use
>> an expect script and code in my CLI password... which I really don't want
>> to do.
>>
>>
>> I looked through the VMREST kit for CUC 10.5 and I don't see anything
>> like this. I can usually find my way around the AXL kit in CUCM but I
>> frequently have issues finding what I need in the CUC VMREST calls.
>>
>>
>> Is there a way to execute this specific query via VMREST to CUC? Is there
>> a VMREST call already baked into CUC that will return similar information?
>>
>>
>> Thanks,
>>
>> Nick
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>>
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[cisco-voip] Execute sql query via vmrest on CUC (or alternatative approach)?

[Nick Barnett]

I found this SQL query

to return a count of all message boxes in CUC. I modified it to return the
top 10 by adding "FIRST 10" immediately after "select" on the first line:

run cuc dbquery unitymbxdb1 select FIRST *10* alias as UserID, count (*) as
messages \

from vw_message, unitydirdb:vw_mailbox, unitydirdb:vw_user \

where mailboxobjectid in \

(select mailboxid from vw_mailbox where unitydirdb: vw_user.objectid =
unitydirdb:vw_mailbox.userobjectid) \

group by alias order by \

messages desc


This works, but it's not very "dev ops friendly." I think I'd have to use
an expect script and code in my CLI password... which I really don't want
to do.


I looked through the VMREST kit for CUC 10.5 and I don't see anything like
this. I can usually find my way around the AXL kit in CUCM but I frequently
have issues finding what I need in the CUC VMREST calls.


Is there a way to execute this specific query via VMREST to CUC? Is there a
VMREST call already baked into CUC that will return similar information?


Thanks,

Nick
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Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[Nick Barnett]

I forgot to put a smiley face in that email, I wasn't trying to be a jerk
with my Trivial File Transfer Protocol jab :)

On Thu, Dec 1, 2016 at 9:54 PM, Nick Barnett <nicksbarn...@gmail.com> wrote:

> By definition, TFTP is trivial.
>
> The service either needed deactivated or the server needed to restart.
> Either way, the TFTP server is going down to regenerate the certs.
>
> On Thu, Dec 1, 2016 at 4:41 PM, Ryan Huff <ryanh...@outlook.com> wrote:
>
>> Anthony and James have highlighted one of the greater weaknesses of
>> thinking like an engineer.
>>
>> As an engineer, we look at TFTP service interruption and see all the
>> potential outcomes and things that could happen. We think about a firmware
>> download being interrupted on an endpoint and realize that it's simply a
>> phone reset to fix.
>>
>> That's great, if your end-users think like engineers and know what you
>> know.
>>
>> Although a nuclear power plant sitting in Japan or China is an extreme
>> example in my opinion, it is right on point. There are many, many
>> situations beyond a nuclear power plant where something as minor as a phone
>> firmware download being interrupted would be completely unacceptable to the
>> customer.
>>
>> In an SMB scenario with clearly defined maintenance windows, I can see
>> this not being such a big deal potentially. However if you're dealing with
>> a customer that counts endpoints in the tens of thousands (or even
>> thousands), it stands to reason that more than a few endpoints might be
>> impacted by something as, "trivial" as a TFTP service reset.
>>
>> It may be trivial in the permanency of the impact it could have on an
>> endpoint, but it is not trivial a enough to assume that it would not have
>> any impact to end-user performance, expectations or usability.
>>
>> -Ryan
>>
>> On Dec 1, 2016, at 5:26 PM, James Buchanan <james.buchan...@gmail.com>
>> wrote:
>>
>> If the endpoint is 8000 miles away from you and located in a nuclear
>> power plant, that TFTP interruption wasn't so trivial.
>>
>> On Thu, Dec 1, 2016 at 5:10 PM, Ben Amick <bam...@humanarc.com> wrote:
>>
>>> An endpoint in the middle of an upgrade has already entirely downloaded
>>> the firmware into memory, and would not be affected. If it is mid-download
>>> then it would have no affect other than breaking the operation and perhaps
>>> requiring a manual restart if it is coming off a factory reset
>>>
>>>
>>>
>>> *Ben Amick*
>>>
>>> Telecom Analyst
>>>
>>>
>>>
>>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>>> Behalf Of *Anthony Holloway
>>> *Sent:* Thursday, December 01, 2016 5:08 PM
>>> *To:* Nick Barnett <nicksbarn...@gmail.com>
>>> *Cc:* Cisco VoIP Group <cisco-voip@puck.nether.net>
>>> *Subject:* Re: [cisco-voip] Are there any gotchas to watch out for
>>> switching to FQDN server names from IP address server names?
>>>
>>>
>>>
>>> Is TFTP really that trivial?  What would happen to an endpoint, which is
>>> in the middle of a firmware upgrade, when you deactivate TFTP?
>>>
>>>
>>>
>>> On Thu, Dec 1, 2016 at 2:51 PM, Nick Barnett <nicksbarn...@gmail.com>
>>> wrote:
>>>
>>> I figured that a reboot would work, but TAC told me it wouldn't... and
>>> rather than experimenting, I just did what they said to do :)   Besides,
>>> deactivating TFTP is trivial and in a properly laid out deployment should
>>> have 0 impact.
>>>
>>>
>>>
>>> On Wed, Nov 30, 2016 at 8:28 AM, NateCCIE <natec...@gmail.com> wrote:
>>>
>>> A reboot does work. What the deal is the new https version of tftp (port
>>> 6972) does not restart with the service restart.  So it continues to use
>>> the old cert. But it does stop and start with a service deactivation and
>>> reactivation.  Before cucm 11 the tftp over http was only plain text (port
>>> 6970)
>>>
>>>
>>>
>>> Sent from my iPhone
>>>
>>>
>>> On Nov 30, 2016, at 1:12 AM, James Buchanan <james.buchan...@gmail.com>
>>> wrote:
>>>
>>> Hello,
>>>
>>> If I remember right, it actually has to be deactivated under Service
>>> Management. It's not just restarting the service.
>>>
>>> Thanks,
>>>
>>> James
>>>
>>>
>>>
>>&

Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[Nick Barnett]

By definition, TFTP is trivial.

The service either needed deactivated or the server needed to restart.
Either way, the TFTP server is going down to regenerate the certs.

On Thu, Dec 1, 2016 at 4:41 PM, Ryan Huff <ryanh...@outlook.com> wrote:

> Anthony and James have highlighted one of the greater weaknesses of
> thinking like an engineer.
>
> As an engineer, we look at TFTP service interruption and see all the
> potential outcomes and things that could happen. We think about a firmware
> download being interrupted on an endpoint and realize that it's simply a
> phone reset to fix.
>
> That's great, if your end-users think like engineers and know what you
> know.
>
> Although a nuclear power plant sitting in Japan or China is an extreme
> example in my opinion, it is right on point. There are many, many
> situations beyond a nuclear power plant where something as minor as a phone
> firmware download being interrupted would be completely unacceptable to the
> customer.
>
> In an SMB scenario with clearly defined maintenance windows, I can see
> this not being such a big deal potentially. However if you're dealing with
> a customer that counts endpoints in the tens of thousands (or even
> thousands), it stands to reason that more than a few endpoints might be
> impacted by something as, "trivial" as a TFTP service reset.
>
> It may be trivial in the permanency of the impact it could have on an
> endpoint, but it is not trivial a enough to assume that it would not have
> any impact to end-user performance, expectations or usability.
>
> -Ryan
>
> On Dec 1, 2016, at 5:26 PM, James Buchanan <james.buchan...@gmail.com>
> wrote:
>
> If the endpoint is 8000 miles away from you and located in a nuclear power
> plant, that TFTP interruption wasn't so trivial.
>
> On Thu, Dec 1, 2016 at 5:10 PM, Ben Amick <bam...@humanarc.com> wrote:
>
>> An endpoint in the middle of an upgrade has already entirely downloaded
>> the firmware into memory, and would not be affected. If it is mid-download
>> then it would have no affect other than breaking the operation and perhaps
>> requiring a manual restart if it is coming off a factory reset
>>
>>
>>
>> *Ben Amick*
>>
>> Telecom Analyst
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Anthony Holloway
>> *Sent:* Thursday, December 01, 2016 5:08 PM
>> *To:* Nick Barnett <nicksbarn...@gmail.com>
>> *Cc:* Cisco VoIP Group <cisco-voip@puck.nether.net>
>> *Subject:* Re: [cisco-voip] Are there any gotchas to watch out for
>> switching to FQDN server names from IP address server names?
>>
>>
>>
>> Is TFTP really that trivial?  What would happen to an endpoint, which is
>> in the middle of a firmware upgrade, when you deactivate TFTP?
>>
>>
>>
>> On Thu, Dec 1, 2016 at 2:51 PM, Nick Barnett <nicksbarn...@gmail.com>
>> wrote:
>>
>> I figured that a reboot would work, but TAC told me it wouldn't... and
>> rather than experimenting, I just did what they said to do :)   Besides,
>> deactivating TFTP is trivial and in a properly laid out deployment should
>> have 0 impact.
>>
>>
>>
>> On Wed, Nov 30, 2016 at 8:28 AM, NateCCIE <natec...@gmail.com> wrote:
>>
>> A reboot does work. What the deal is the new https version of tftp (port
>> 6972) does not restart with the service restart.  So it continues to use
>> the old cert. But it does stop and start with a service deactivation and
>> reactivation.  Before cucm 11 the tftp over http was only plain text (port
>> 6970)
>>
>>
>>
>> Sent from my iPhone
>>
>>
>> On Nov 30, 2016, at 1:12 AM, James Buchanan <james.buchan...@gmail.com>
>> wrote:
>>
>> Hello,
>>
>> If I remember right, it actually has to be deactivated under Service
>> Management. It's not just restarting the service.
>>
>> Thanks,
>>
>> James
>>
>>
>>
>> On Tue, Nov 29, 2016 at 11:36 PM, Derek Andrew <derek.and...@usask.ca>
>> wrote:
>>
>> Would a simple reboot accomplish the same as deactivating and activating?
>>
>>
>>
>> On Mon, Nov 28, 2016 at 2:19 PM, Nick Barnett <nicksbarn...@gmail.com>
>> wrote:
>>
>> I just thought I would share what happened with this, even though it is
>> super old. Changing the node names to FQDN was mostly painless. The one
>> thing that bit me was bug CSCuy13916. After changing the names of the
>> nodes, the TFTP service needs to be DEACTIVATED and then re-activated i

Re: [cisco-voip] Call Recording with CUCM9.1

[Nick Barnett]

And if you have centralized call ingress/egress, like with enterprise SIP
trunks, you can use media class forking on the CUBE to cut down on WAN
usage.

On Thu, Dec 1, 2016 at 8:22 AM, Ryan Huff  wrote:

> Cisco MediaSense is a great (and admittedly, simple) option for BIB
> recording; just need to purchase the RTU and put it on some virtual
> hardware.
>
> Sent from my iPhone
>
> On Dec 1, 2016, at 9:17 AM, Brian Meade  wrote:
>
> You're probably much better using BIB(built-in bridge)-based/network
> recording instead.  It uses the Built-in-bridge of the phones to send
> duplicate RTP streams to a recording server.  Most recording servers out
> there now prefer this method.
>
> It's technically possible to route the span sessions over the WAN.  ERSPAN
> on your switches would make this a lot easier but I'd really recommend
> getting away from that type of setup.
>
> On Thu, Dec 1, 2016 at 6:24 AM, Michel L. M. B. Perez <
> michelmbpe...@gmail.com> wrote:
>
>> Hi guys,
>>
>> I never did that, so that´s why i am asking for all entire list here. Is
>> it possible to record calls over WAN using CUCM?
>>
>> Today, my scenario is (branches):
>> I have one (old) machine doing the recording of the calls in all branch
>> sites, all Phones that needs to be recorded are using a span session on the
>> switches of Branches, the traffic is routed to the Recorder monitor port.
>> I have the same situation in all sites, HQ and Branches.
>>
>> My desire is to have here just one call recorder over my Vmware
>> infrastructure, and all of this recorded calls being rotued over IP of the
>> virtualized machine.
>>
>> I was reading this document: http://www.cisco.com
>> /c/en/us/td/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsgd
>> -712-cm/fsmr.html#wp1048991
>>
>> Do you think that this is possible?
>>
>> Kr.,
>> --
>> Michel Perez
>> Skype: michelmbperez
>> michelmbpe...@gmail.com
>> http://br.linkedin.com/in/michelmbperez
>>
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>>
>>
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>
>
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Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[Nick Barnett]

I just thought I would share what happened with this, even though it is
super old. Changing the node names to FQDN was mostly painless. The one
thing that bit me was bug CSCuy13916. After changing the names of the
nodes, the TFTP service needs to be DEACTIVATED and then re-activated in
order to fully update the certificates.  Before taking those steps, I kept
getting certificate errors from CuciLync, but afterwards, everything worked
as designed.

Other than that, any CTI route points (and any other device as well) that
exist will fall to another node in the CMG. Not a big deal, just something
to be aware of.

Thanks,
Nick

On Wed, Aug 31, 2016 at 3:13 PM, Nick Barnett <nicksbarn...@gmail.com>
wrote:

> We are on 10.0 and this cluster has been upgraded over the years from 8.0
> to 8.6 to 10.0.  I know it used to be common practice to rip the host name
> out of a new node and put in the IP address. That's how we are set up...
> but now that I need to do some work with certs so that jabber and cucilync
> work properly, it's time to fix this.
>
> Is there anything I should watch out for? Anything that may bite me in
> rare cases? We have CER, CVP, CUC, UCCE and a rarely used IMP.
>
> I checked that each node has DNS enabled by looking at "show network eth0"
> on each sub. I also then looked up each FQDN from each node and they all
> resolve properly. As far as I know, that's about it.
>
> Thanks in advance!
>
> nick
>
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Re: [cisco-voip] Query or report to find devices that have video calling enabled?

[Nick Barnett]

Also very helpful, thanks for the examples!

On Mon, Nov 28, 2016 at 1:09 PM, Alan Libbee <alan.lib...@umuc.edu> wrote:

> Actually, you can query and filter based on the xml data. I wrote some
> sample queries for you, hope this helps.
>
> -Alan
>
> This will show you devices with an override for video enabled:
> run sql select device.name, device.description, devicepool.name from
> device inner join devicexml4k on device.pkid = devicexml4k.fkdevice inner
> join devicepool on device.fkdevicepool = devicepool.pkid where
> devicexml4k.xml like '%1%'
>
> This will show you devices with an override for video disabled:
> run sql select device.name, device.description, devicepool.name from
> device inner join devicexml4k on device.pkid = devicexml4k.fkdevice inner
> join devicepool on device.fkdevicepool = devicepool.pkid where
> devicexml4k.xml like '%0%'
>
> This will include the xml data if you like:
> run sql select device.name, device.description, devicepool.name,
> devicexml4k.xml from device inner join devicexml4k on device.pkid =
> devicexml4k.fkdevice inner join devicepool on device.fkdevicepool =
> devicepool.pkid where devicexml4k.xml like '%1 videoCapability>%'
>
> On Mon, Nov 28, 2016 at 1:19 PM, Brian Meade <bmead...@vt.edu> wrote:
>
>> That's stored in the devicexml16k , devicexml8k, and devicexml4k tables.
>>
>> Only non-default values are stored there.  It's in XML so can't make a
>> query to check for a certain item.  You have to use some post-processing to
>> pull out the values you care about.
>>
>> On Mon, Nov 28, 2016 at 1:04 PM, Nick Barnett <nicksbarn...@gmail.com>
>> wrote:
>>
>>> Hello. We are a Cisco phone shop, but use Lync/Skype for IM We also
>>> use cucilync. In order to reduce confusion with end users, we've elected to
>>> disable video calling from CSF devices and let them do video calls inside
>>> Lync/Skype.
>>>
>>> I batched in a change to all CSF devices to disable video calling. The
>>> batch job reports as successful with no errors. Every now and then I come
>>> across a CSF device that still has video calling enabled.
>>>
>>> I was poking around in the informix trying to find a query to run that
>>> will list all CSF devices that still have video calling enabled, but I'm
>>> striking out. It seems that this information is not housed in the "device"
>>> table. I also went through the data dictionary to no avail.
>>>
>>> Maybe there is an easier way to do this, but I just need a report, or
>>> query that will show these improperly configured devices. Does anyone know
>>> where the "Product Specific Configuration Layout" at the bottom of the
>>> CSF device config is actually housed and how it is related to the actual
>>> CSF device?
>>>
>>> Thanks,
>>> Nick
>>>
>>> ___
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>>>
>>>
>>
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>>
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Re: [cisco-voip] Query or report to find devices that have video calling enabled?

[Nick Barnett]

This is incredibly helpful. Thank you. I didn’t find any data in the 16k
and 8k tables, but 4k has a LOT. Do you know what the 8k and 16k tables are
actually used for?



I wrote this AXL call that seems to list out exactly what I was looking
for. All of our CSF devices start with “UCCSF”



http://schemas.xmlsoap.org/soap/envelope/;
xmlns:ns="http://www.cisco.com/AXL/API/10.0;>

   

   

  

 

SELECT name, description FROM device

WHERE name LIKE 'UCCSF%'

 AND name IN

(SELECT device.name

 FROM devicexml4k INNER JOIN device ON
devicexml4k.fkdevice = device.pkid

 WHERE device.name LIKE 'UCCSF%'

  AND devicexml4k.xml LIKE
'%1/videoCapability%')



  

   



On Mon, Nov 28, 2016 at 12:19 PM, Brian Meade <bmead...@vt.edu> wrote:

> That's stored in the devicexml16k , devicexml8k, and devicexml4k tables.
>
> Only non-default values are stored there.  It's in XML so can't make a
> query to check for a certain item.  You have to use some post-processing to
> pull out the values you care about.
>
> On Mon, Nov 28, 2016 at 1:04 PM, Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> Hello. We are a Cisco phone shop, but use Lync/Skype for IM We also
>> use cucilync. In order to reduce confusion with end users, we've elected to
>> disable video calling from CSF devices and let them do video calls inside
>> Lync/Skype.
>>
>> I batched in a change to all CSF devices to disable video calling. The
>> batch job reports as successful with no errors. Every now and then I come
>> across a CSF device that still has video calling enabled.
>>
>> I was poking around in the informix trying to find a query to run that
>> will list all CSF devices that still have video calling enabled, but I'm
>> striking out. It seems that this information is not housed in the "device"
>> table. I also went through the data dictionary to no avail.
>>
>> Maybe there is an easier way to do this, but I just need a report, or
>> query that will show these improperly configured devices. Does anyone know
>> where the "Product Specific Configuration Layout" at the bottom of the
>> CSF device config is actually housed and how it is related to the actual
>> CSF device?
>>
>> Thanks,
>> Nick
>>
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[cisco-voip] Query or report to find devices that have video calling enabled?

[Nick Barnett]

Hello. We are a Cisco phone shop, but use Lync/Skype for IM We also use
cucilync. In order to reduce confusion with end users, we've elected to
disable video calling from CSF devices and let them do video calls inside
Lync/Skype.

I batched in a change to all CSF devices to disable video calling. The
batch job reports as successful with no errors. Every now and then I come
across a CSF device that still has video calling enabled.

I was poking around in the informix trying to find a query to run that will
list all CSF devices that still have video calling enabled, but I'm
striking out. It seems that this information is not housed in the "device"
table. I also went through the data dictionary to no avail.

Maybe there is an easier way to do this, but I just need a report, or query
that will show these improperly configured devices. Does anyone know where
the "Product Specific Configuration Layout" at the bottom of the CSF device
config is actually housed and how it is related to the actual CSF device?

Thanks,
Nick
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[cisco-voip] change displayed number when dialing out?

[Nick Barnett]

Long story short, I have to use steering digits to send calls to the
correct CUBE and then select the proper SIP trunk to egress to PSTN.  This
works well for our setup, but now the users occasionally complain that
their phone is displaying "weird numbers."

For instance, calling out of our corp office 800-555-1212, the user has to
dial a 9 so they are used to seeing 918005551212.  Now they see
10010718005551212.

My outbound dial peer on the cube matches those steering digits and then
strips them to send to the PSTN. If I had a way of stripping the digits
with the inbound dialpeer, the steering digits would likely be removed...
but I can't think of a way to do that.

I have 1 cluster, 2 CUBEs and each CUBE has 3 sip trunks from different
carriers.

Any ideas on how to "fix" this "problem" ?

Thanks,
Nick

To expand on the "long story short", nearly all of our business uses EM and
they are located all over the country. We have a corp data center and a
remote DC. Half of all phones register in each DC. Each DC has a CUBE and 3
SIP trunks. I wanted to use standard local route groups and DPs to guide
the outbound calls to the proper sip trunk, but EM profiles don't have
device pools. This may not sound like an issue, but some of our offices
have DIDs from multiple carriers. If I send calls from a Carrier1 DID out
the Carrier2 SIP trunk, I have to pay alien TN charges. Despite all of
this, the solution in place works except for this minor annoyance.
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Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[Nick Barnett]

awesome info, thank you! I'm pretty sure our gateways have DNS, but I'll be
making sure... and I've dealt with enough dbreplication issues lately that
I'll be vigilant on that as well. It may take me 8 hours to do it that way
though blech. I just had to do a cluster reboot earlier this week and
it took just under an hour with 9 nodes.

On Wed, Aug 31, 2016 at 4:15 PM, <dan...@ohnesorge.me> wrote:

> One of the most important points that people tend to forget when changing
> the processnode (System>Server) entries is that MGCP and SCCP gateways will
> download a config file (like a phone) and will need to resolve these
> entries. For what ever reason I've seen so many customers not add any ip
> name server to their routers so this one can bite you in the ass.
>
> Now with regards to actually changing the entries, I have done this way
> too many times. What you REALLY need to do is change the entry one by one,
> then restart all the nodes in the cluster one by one. Then change the next
> entry and repeat! I know this sounds totally unnecessary but the
> processnode has the ability to stuff up your dbreplication to the point
> where TAC will suggest a rebuild.
>
> Thanks,
> Daniel
>
> Sent from my iPhone
>
> On 1 Sep 2016, at 06:39, Ryan Huff <ryanh...@outlook.com> wrote:
>
> Nick,
>
>
> If the UC servers already have DNS entries (means they already have a
> domain name too); then the servers are already using FQDNs, at least for
> internal referencing. If you're saying the you want to change the
> processNode names (the CM Server references) then as long as the FQDNs are
> resolvable in the forward and reverse direction, it should be fine.
>
>
> If you need to change the hostname or domain names of the servers to
> something more palatable (a crossroads often encountered when dealing with
> Jabber and end users and UC servers that were IP addresses first); that is
> a horse of a much different color; please *carefully *consult
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/
> install/10_0_1/ipchange/CUCM_BK_C3782AAB_00_change-
> ipaddress-hostname-100/CUCM_BK_C3782AAB_00_change-ipaddress-hostname-100_
> chapter_0100.html (especially in the case of IM & Presence HA)
>
>
> If you are also talking about changing the IP Phone URL references under
> Enterprise Parameters (from IP address to FQDN); your phone networks will
> need DNS capabilities to resolve those FQDNs as well. As a matter of
> practice, I always ensure IP phone networks have DNS capabilities, but it
> can be uncommonly found out in the wild.
>
>
> Beyond that, if you are simply just changing the processNode references
> for IP addresses to FQDNs (presumably, so CUCM requests come from an FQDN
> and not an IP address) and everything is already resolving correctly, you
> should be g2g.
>
>
> Thanks,
>
>
> = Ryan =
>
>
>
>
> --
> *From:* cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Nick
> Barnett <nicksbarn...@gmail.com>
> *Sent:* Wednesday, August 31, 2016 4:13 PM
> *To:* Cisco VoIP Group
> *Subject:* [cisco-voip] Are there any gotchas to watch out for switching
> to FQDN server names from IP address server names?
>
> We are on 10.0 and this cluster has been upgraded over the years from 8.0
> to 8.6 to 10.0.  I know it used to be common practice to rip the host name
> out of a new node and put in the IP address. That's how we are set up...
> but now that I need to do some work with certs so that jabber and cucilync
> work properly, it's time to fix this.
>
> Is there anything I should watch out for? Anything that may bite me in
> rare cases? We have CER, CVP, CUC, UCCE and a rarely used IMP.
>
> I checked that each node has DNS enabled by looking at "show network eth0"
> on each sub. I also then looked up each FQDN from each node and they all
> resolve properly. As far as I know, that's about it.
>
> Thanks in advance!
>
> nick
>
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>
>
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Re: [cisco-voip] Are there any gotchas to watch out for switching to FQDN server names from IP address server names?

[Nick Barnett]

Thanks Ryan. Yes, I'm just trying to change the process node names. Right
now, when someone logs in with cucilync, it prompts them for several
certificates. Those certs are references a CN that is an IP address. I'm
thinking that if I change the node name to an FQDN, and assuming I have my
cert chain signed properly and deployed, hopefully the end user will NOT
see these cert warnings any more. Does that sound about right?

On Wed, Aug 31, 2016 at 3:39 PM, Ryan Huff <ryanh...@outlook.com> wrote:

> Nick,
>
>
> If the UC servers already have DNS entries (means they already have a
> domain name too); then the servers are already using FQDNs, at least for
> internal referencing. If you're saying the you want to change the
> processNode names (the CM Server references) then as long as the FQDNs are
> resolvable in the forward and reverse direction, it should be fine.
>
>
> If you need to change the hostname or domain names of the servers to
> something more palatable (a crossroads often encountered when dealing with
> Jabber and end users and UC servers that were IP addresses first); that is
> a horse of a much different color; please *carefully *consult
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/
> install/10_0_1/ipchange/CUCM_BK_C3782AAB_00_change-
> ipaddress-hostname-100/CUCM_BK_C3782AAB_00_change-ipaddress-hostname-100_
> chapter_0100.html (especially in the case of IM & Presence HA)
>
>
> If you are also talking about changing the IP Phone URL references under
> Enterprise Parameters (from IP address to FQDN); your phone networks will
> need DNS capabilities to resolve those FQDNs as well. As a matter of
> practice, I always ensure IP phone networks have DNS capabilities, but it
> can be uncommonly found out in the wild.
>
>
> Beyond that, if you are simply just changing the processNode references
> for IP addresses to FQDNs (presumably, so CUCM requests come from an FQDN
> and not an IP address) and everything is already resolving correctly, you
> should be g2g.
>
>
> Thanks,
>
>
> = Ryan =
>
>
>
>
> --
> *From:* cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Nick
> Barnett <nicksbarn...@gmail.com>
> *Sent:* Wednesday, August 31, 2016 4:13 PM
> *To:* Cisco VoIP Group
> *Subject:* [cisco-voip] Are there any gotchas to watch out for switching
> to FQDN server names from IP address server names?
>
> We are on 10.0 and this cluster has been upgraded over the years from 8.0
> to 8.6 to 10.0.  I know it used to be common practice to rip the host name
> out of a new node and put in the IP address. That's how we are set up...
> but now that I need to do some work with certs so that jabber and cucilync
> work properly, it's time to fix this.
>
> Is there anything I should watch out for? Anything that may bite me in
> rare cases? We have CER, CVP, CUC, UCCE and a rarely used IMP.
>
> I checked that each node has DNS enabled by looking at "show network eth0"
> on each sub. I also then looked up each FQDN from each node and they all
> resolve properly. As far as I know, that's about it.
>
> Thanks in advance!
>
> nick
>
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[cisco-voip] upgrading cucilync 8.6 to 11.6.0?

[Nick Barnett]

I'm working on packaging Cucilync 11.6.0 so it will work with the new Skype
for business client. we are using windows 7 64bit, but the skype client is
32bit. I've never really dealt with installing Jabber (since we use Lync
mostly) and I'm having a tough time understanding the boot process for the
new Cucilync. I understand that after v9, cucilync has the ability to find
the cluster with DNS SRV records. I have this part working... I think. I
can see the DNS request hit the wire with a packet capture and my cucilync
properly registers in softphone mode. When i change the device pool of the
CSF, the registration follows to the correct sub (something that the old
version did not do in our environment).

I've set up a new service profile and created profiles for VM, mailstore,
CTI... I've also applied this to my end user settings.

I also created a new cucilync-config.xml file and uploaded it to TFTP. I
installed cucilync with the switch to recognize this new xml file. The xml
file has a couple of policy settings, like "not docked." That's the whole
point of this exercise, people hate the new docking UI and I need to get
rid of it.

I'm sure there are more issues at hand here, but I can't drag a call from
skype to cucilync. My first thoughts were that CTI is not working, but
that's just a guess.

I started plowing through the files in the %appdata%\cisco\Unified
Communications\Jabber\CSF\Config\ folder, and now I'm even more confused. I
see OLD data (i deleted all of the registry entries as well as all files
and directories from 8.6). i know this is old data because I'm taking this
opportunity to point the cucilync instances to our new tftp servers... they
were previously installed to point to a set of 3 TFTP servers, that's not
the current config.

Am I over thinking this? Have I missed something dumb and obvious? Is there
a decent document on how the app is supposed to boot? Finding stuff on
cucilync is rough, it feels like cucilync doesn't even exist since
everything is referred to as Jabber.

Thanks for listening to me brain dump :)

Nick
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Re: [cisco-voip] Route Dial-Peer Based On Response

[Nick Barnett]

In my scenario, I have an srv record setup with 4 ccm nodes, 1 and 2 are
equal weight and preferred, 3 and 4 have equal weights but are higher
weighted than the 1/2 pair.  ccm1 and 2 are at a main data center, ccm 3
and 4 are at a remote DC. I want calls load balanced to 1 and 2, and in the
case those are unavailable, load balance between 3 and 4.  This works as
expected and is a great alternative to having 4 different (unneeded) dial
peers.

The problem came to light when I added keepalive to this DP. everything is
fine and dandy when 1/2 are available, but in testing failover (acl on an
uplink to block comm to 1/2), the keepalive took the DP out of service.
Perhaps there are timers that could be shortened or extended to provide a
different response, but I decided I didn't really need keepalive when using
SRV in this manner.

On Thu, Jul 21, 2016 at 2:31 PM, NateCCIE <natec...@gmail.com> wrote:

> I have only used SRV destination a little bit, but my CVP friends use them
> extensively.  I have never seen the option ping with SRV records act
> differently than I would have expected.
>
> Do you have experiences where the whole dialpeer went down and other
> members of the SRV were still accessable?
>
> Sent from my iPhone
>
> On Jul 21, 2016, at 9:11 AM, Nick Barnett <nicksbarn...@gmail.com> wrote:
>
> That may work fine based on how the SRV and options pings are
> configured... but if you are counting on an SRV record to point to another
> CCM sub when the primary is down (etc...), options pings will likely take
> the whole DP down when a single host in the SRV goes unreachable.
>
> On Wed, Jul 20, 2016 at 6:12 PM, Erick Bergquist <erick...@gmail.com>
> wrote:
>
>> You could also look at adding keep alive (options ping) to the dial peers
>> and call manager sip trunk with options mentioned above.
>>
>> Do you mind sharing the tcl script?
>>
>> Erick
>>
>>
>> On Wednesday, July 20, 2016, Pawlowski, Adam <aj...@buffalo.edu> wrote:
>>
>>> Nathan,
>>>
>>>
>>>
>>> Thanks, this looks to be exactly what I’m looking for,
>>> this way I don’t convey the wrong message. It doesn’t seem like I can have
>>> more than one option other than hunt or don’t hunt, and it seems to be
>>> proper to let the telephony provider handle it. Cool. Thanks again.
>>>
>>>
>>>
>>> Adam
>>>
>>>
>>>
>>> *From:* Nathan Richardson [mailto:nrichard...@gci.com]
>>> *Sent:* Wednesday, July 20, 2016 12:34 PM
>>> *To:* Pawlowski, Adam; 'cisco-voip@puck.nether.net'
>>> *Subject:* RE: Route Dial-Peer Based On Response
>>>
>>>
>>>
>>> One thing that may help is to configure the “voice hunt” settings. For
>>> example, you could put in “no voice hunt temp-fail” which would make the
>>> router stop routing if it receives a cause code 41 from the CM so it would
>>> skip your TCL script in that scenario and should send that code back to
>>> your ITSP. It may even work to combine “no voice hunt all” with “voice hunt
>>> unassigned-number” or something like that.
>>>
>>>
>>>
>>>
>>> http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_v2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1190281
>>>
>>>
>>>
>>> -Nathan Richardson
>>>
>>>
>>>
>>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>>> Behalf Of *Pawlowski, Adam
>>> *Sent:* Wednesday, July 20, 2016 5:33 AM
>>> *To:* 'cisco-voip@puck.nether.net' <cisco-voip@puck.nether.net>
>>> *Subject:* [cisco-voip] Route Dial-Peer Based On Response
>>>
>>>
>>>
>>> [External Email]
>>>
>>> Hey all,
>>>
>>>
>>>
>>> I’ve set up our CUBE routers to try and be a bit more
>>> slick, so I am making use of e164 pattern maps, dial peer groups, and DNS
>>> SRV lookups for redundancy/randomization. All that actually seems to be
>>> working rather well. I have a requirement to make any inactive/unallocated
>>> number in my UCM play a custome intercept. I did this, at least for now, by
>>> setting up a secondary dial peer that matches with a higher preference than
>>> my UCM peer, and it plays an announcement with a TCL script.
>>>
>>>
>>>
>>> I’d like to set this up so that if the UCM peer is down,
>>> or if it receives some other code indicating a temporary failure, etc, I
>>> either would like 

Re: [cisco-voip] Route Dial-Peer Based On Response

[Nick Barnett]

That may work fine based on how the SRV and options pings are configured...
but if you are counting on an SRV record to point to another CCM sub when
the primary is down (etc...), options pings will likely take the whole DP
down when a single host in the SRV goes unreachable.

On Wed, Jul 20, 2016 at 6:12 PM, Erick Bergquist  wrote:

> You could also look at adding keep alive (options ping) to the dial peers
> and call manager sip trunk with options mentioned above.
>
> Do you mind sharing the tcl script?
>
> Erick
>
>
> On Wednesday, July 20, 2016, Pawlowski, Adam  wrote:
>
>> Nathan,
>>
>>
>>
>> Thanks, this looks to be exactly what I’m looking for,
>> this way I don’t convey the wrong message. It doesn’t seem like I can have
>> more than one option other than hunt or don’t hunt, and it seems to be
>> proper to let the telephony provider handle it. Cool. Thanks again.
>>
>>
>>
>> Adam
>>
>>
>>
>> *From:* Nathan Richardson [mailto:nrichard...@gci.com]
>> *Sent:* Wednesday, July 20, 2016 12:34 PM
>> *To:* Pawlowski, Adam; 'cisco-voip@puck.nether.net'
>> *Subject:* RE: Route Dial-Peer Based On Response
>>
>>
>>
>> One thing that may help is to configure the “voice hunt” settings. For
>> example, you could put in “no voice hunt temp-fail” which would make the
>> router stop routing if it receives a cause code 41 from the CM so it would
>> skip your TCL script in that scenario and should send that code back to
>> your ITSP. It may even work to combine “no voice hunt all” with “voice hunt
>> unassigned-number” or something like that.
>>
>>
>>
>>
>> http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_v2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1190281
>>
>>
>>
>> -Nathan Richardson
>>
>>
>>
>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Pawlowski, Adam
>> *Sent:* Wednesday, July 20, 2016 5:33 AM
>> *To:* 'cisco-voip@puck.nether.net' 
>> *Subject:* [cisco-voip] Route Dial-Peer Based On Response
>>
>>
>>
>> [External Email]
>>
>> Hey all,
>>
>>
>>
>> I’ve set up our CUBE routers to try and be a bit more
>> slick, so I am making use of e164 pattern maps, dial peer groups, and DNS
>> SRV lookups for redundancy/randomization. All that actually seems to be
>> working rather well. I have a requirement to make any inactive/unallocated
>> number in my UCM play a custome intercept. I did this, at least for now, by
>> setting up a secondary dial peer that matches with a higher preference than
>> my UCM peer, and it plays an announcement with a TCL script.
>>
>>
>>
>> I’d like to set this up so that if the UCM peer is down,
>> or if it receives some other code indicating a temporary failure, etc, I
>> either would like to bypass this peer so the code goes back to the ITSP, or
>> I can play a message saying something about technical difficulties, etc.
>> I’m not sure it’s possible to do this? The other way of doing this would be
>> to have the UCM itself with a translation or something to roll to an
>> audiotext mailbox, which is how we do this today, but it requires either
>> that we maintain translations for all numbers, or a generic one that will
>> answer to all extensions queried at the system which I don’t want to do
>> either.
>>
>>
>>
>> Any thoughts?
>>
>>
>>
>> Regards,
>>
>>
>>
>> Adam Pawlowski
>>
>> SUNY Buffalo NCS
>>
>>
>>
>
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Re: [cisco-voip] DS3 voice delivery?

[Nick Barnett]

Thank you everybody. The mux jumped into maintenance mode for some reason.
We were able to find a 20 year old manual that had DB9 ->RJ45 pinouts... I
cut the end off a blue cisco console cable and rewired the end. Then we
could session into the device and set it back into service. We're up for
now, hopefully it doesn't decide it needs to be in maintenance mode for
another 15 years :)

On Wed, Jun 8, 2016 at 8:51 AM, Nick Barnett <nicksbarn...@gmail.com> wrote:

> So, we have an old MUX that died. We have to either replace the MUX or use
> something else. Is it possible to use a DS3 SM on an ISR to terminate the
> DS3? Right now, we have a DS3 that hits a mux and breaks out to 28 PRIs...
> those PRIs go into a plethora of VWIC interfaces on a SIP router. Would it
> be possible to get another module for this router that lets us plug the
> coax in and skip all of this VWIC/MUX business?
>
> Thanks,
> Nick
>
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[cisco-voip] DS3 voice delivery?

[Nick Barnett]

So, we have an old MUX that died. We have to either replace the MUX or use
something else. Is it possible to use a DS3 SM on an ISR to terminate the
DS3? Right now, we have a DS3 that hits a mux and breaks out to 28 PRIs...
those PRIs go into a plethora of VWIC interfaces on a SIP router. Would it
be possible to get another module for this router that lets us plug the
coax in and skip all of this VWIC/MUX business?

Thanks,
Nick
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I just thought I would update with how I got this working. It was a multi
staged "fix." I rewrote the entire dial plan to use e164 pattern maps and
SRV records. This reduced my dial peer count from 150+ to less than 20.
Then I took the INSIDE dial peers and bound them at the DP level to the
inside interface. OUTSIDE PSTN facing DPs got bound to the outside
interface.

At this point, register requests were sourcing from the external interface
IP address and not the external VIP (I'm using CUBE HA).

To fix that, I placed a global bind in voice service voip to the outside
interface. This made the REGISTER requests source from the external VIP...
but it broke other stuff.

With the internal DPs bound inside and the external DPs bound outside...
the SIP REGISTER events were now using the global bind... but my SIP
OPTIONS pings from CUCM were also answering from that public IP on the
outside interface... so my SIP trunks to the cube from CUCM went out of
service.

I added voice class SIP URIs to my internal CUCM dial peers so that the
inside interface would answer SIP OPTIONS pings.

I put my credentials lines in sip-ua and auth lines in the external DPs.

Everything is up and running.

Thanks for everyone's help and suggestions.

On Wed, May 4, 2016, 4:54 PM Dave Goodwin <dave.good...@december.net> wrote:

> Is there anything wrong with adding voice-class sip bind commands to ALL
> the voip dial-peers, and then set the global binding to the interface that
> faces the ITSP requiring authentication (since it seems sip-ua REGISTER
> messages use the global bind)?
>
> -Dave
>
> On Wed, May 4, 2016 at 4:22 PM, Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> Thanks for everybody's ideas.
>>
>> Unfortunately, 15.6 is OUT because it is not on the CVP 10.0
>> compatibility matrix :(
>>
>> I'm going to look at using multiple registrars and see if I can trick it
>> into behaving... if that doesn't work, I guess I'll have to remove my
>> global binding...
>>
>>
>> On Wed, May 4, 2016 at 11:35 AM, Sreekanth <sreen...@cisco.com> wrote:
>>
>>> Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar
>>> server globally with the authentication and credential parameters. These
>>> REGISTER messages will be bound to the interface that is bound under voice
>>> service voip -> sip. However, in the 15.6(2)T version, the tenant
>>> configurations under the dial-peers will instruct the CUBE to send out
>>> REGISTER messages.
>>>
>>> I just checked with the router in my lab and actually, option 2 won't be
>>> possible. It won't instruct the CUBE to send out REGISTER messages. It will
>>> only instruct the CUBE to add authentication credentials and realm settings
>>> when sending out the INVITE messages towards the session target configured
>>> under the dial-peer.
>>>
>>> You will have to go with option 1.
>>> *1. Create the voice class tenant for the SIP trunk to ITSP and bind it
>>> with the right interface.*
>>> voice class tenant 1
>>>   registrar dns:cisco.com expires 3600
>>>   credentials username cisco password cisco realm cisco.com
>>>   authentication username cisco123 password 7 cisco123
>>>   sip-server dns:cisco.com
>>>   bind control source-interface GigabitEthernet0/2
>>>   bind media source-interface GigabitEthernet0/2
>>>   early-offer forced
>>>
>>> *2. Apply the voice class tenant to the dial-peer. Create specific
>>> dial-peers towards ITSP.*
>>> dial-peer voice X voip
>>>  voice-class sip tenant 1
>>>
>>> When this is done, CUBE will send REGISTER messages as well towards this
>>> ITSP with the traffic bound to gig0/2.
>>> This way you can have multiple ITSP trunks on 1 CUBE.
>>>
>>> Sreekanth
>>>
>>>
>>>
>>>
>>> On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
>>>
>>> I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
>>> upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
>>> versions kind of scares me, so maybe staging is in order.
>>>
>>> *I do have some limited auth commands on the dial peer, if this is what
>>> you were talking about... but I don't think it applies in this scenario. I
>>> don't have any options for credentials:*
>>> CUBE(config-dial-peer)#voice-class sip authenticate ?
>>>   redirecting-number  Use redirecting number credentials while
>>> authenticating
>>>
>>> CUBE(config-dial-peer)#voice-class sip cred
>>> CUB

[cisco-voip] UCCE agents on wireless IP Communicator?

[Nick Barnett]

Does anyone have any experiences running CIPC on wireless for UCCE agents?
It sounds like a...um, bad idea to me.  One of my customers is moving to
this "design."

A cursory look at the 10.0 SRND didn't show any hits for "wired" or
"wireless".

thanks,
Nick
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Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

Thanks for everybody's ideas.

Unfortunately, 15.6 is OUT because it is not on the CVP 10.0 compatibility
matrix :(

I'm going to look at using multiple registrars and see if I can trick it
into behaving... if that doesn't work, I guess I'll have to remove my
global binding...


On Wed, May 4, 2016 at 11:35 AM, Sreekanth <sreen...@cisco.com> wrote:

> Yes, sip-ua tells CUBE to send REGISTER messages towards a Registrar
> server globally with the authentication and credential parameters. These
> REGISTER messages will be bound to the interface that is bound under voice
> service voip -> sip. However, in the 15.6(2)T version, the tenant
> configurations under the dial-peers will instruct the CUBE to send out
> REGISTER messages.
>
> I just checked with the router in my lab and actually, option 2 won't be
> possible. It won't instruct the CUBE to send out REGISTER messages. It will
> only instruct the CUBE to add authentication credentials and realm settings
> when sending out the INVITE messages towards the session target configured
> under the dial-peer.
>
> You will have to go with option 1.
> *1. Create the voice class tenant for the SIP trunk to ITSP and bind it
> with the right interface.*
> voice class tenant 1
>   registrar dns:cisco.com expires 3600
>   credentials username cisco password cisco realm cisco.com
>   authentication username cisco123 password 7 cisco123
>   sip-server dns:cisco.com
>   bind control source-interface GigabitEthernet0/2
>   bind media source-interface GigabitEthernet0/2
>   early-offer forced
>
> *2. Apply the voice class tenant to the dial-peer. Create specific
> dial-peers towards ITSP.*
> dial-peer voice X voip
>  voice-class sip tenant 1
>
> When this is done, CUBE will send REGISTER messages as well towards this
> ITSP with the traffic bound to gig0/2.
> This way you can have multiple ITSP trunks on 1 CUBE.
>
> Sreekanth
>
>
>
>
> On Wednesday 04 May 2016 09:29 PM, Nick Barnett wrote:
>
> I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
> upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
> versions kind of scares me, so maybe staging is in order.
>
> *I do have some limited auth commands on the dial peer, if this is what
> you were talking about... but I don't think it applies in this scenario. I
> don't have any options for credentials:*
> CUBE(config-dial-peer)#voice-class sip authenticate ?
>   redirecting-number  Use redirecting number credentials while
> authenticating
>
> CUBE(config-dial-peer)#voice-class sip cred
> CUBE(config-dial-peer)#voice-class sip c?
>   call-route  calltype-video  conn-reuse  copy-list
>
> *There is also the registration commands:*
> CUBE(config-dial-peer)#voice-class sip registration ?
>   passthrough  SIP Registration Passthrough Options
>
> CUBE(config-dial-peer)#voice-class sip registration passthrough ?
>   dynamic  SIP Registration Use dynamic Registrar Details (default)
>   local-fallback   Local Fallback - (e2e)
>   rate-limit   SIP Registration pass-through rate-limit Options
>   reg-sync Registration Sync - send REGISTER when registrar up
> (p2p)
>   registrar-index  Registrar Index(s) used for registration passthrough
>   static   SIP Registration Use static Registrar Details
>   system   Use global registration passthrough CLI setting
>   
>
> *I tried using the system passthrough setting, but it did not work.*
>
> *I need to make sure I understand what is actually happening.*
>
> *I don't think the CUBE is even looking at dial-peers for REGISTER
> messages. Am I correct?  If so, no amount of dial peer settings is going to
> make any difference here... unless there is a way to create a dial-peer
> that will intercept REGISTER messages. I believe it is using the REALM
> settings in the credentials and authentication strings (that I entered into
> sip-ua). And sip-ua is using the global bind settings I set within voice
> service voip -> SIP (which are set to the inside interface).*
>
> *Please set me straight!*
>
> Thanks,
> Nick
>
> On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
> <sreen...@cisco.com>sreen...@cisco.com> wrote:
>
>> What IOS version are you running on the CUBE? I can think of a couple of
>> things.
>> 1. In 15.6(2)T, a new feature has been introduced called multi-tenant
>> where you can configure separate voice class tenants. Each tenant can have
>> separate authentication mutually exclusive to one another and can be bound
>> to different interfaces.
>>
>> 2. In your current IOS, check if you are able to configure the
>> authentication and credential commands at th

Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I'm currently on c3900e-universalk9-mz.SPA.153-3.M6, but can totally
upgrade. Was actually planning on going to 15.4 this weekend. Jumping 3
versions kind of scares me, so maybe staging is in order.

*I do have some limited auth commands on the dial peer, if this is what you
were talking about... but I don't think it applies in this scenario. I
don't have any options for credentials:*
CUBE(config-dial-peer)#voice-class sip authenticate ?
  redirecting-number  Use redirecting number credentials while
authenticating

CUBE(config-dial-peer)#voice-class sip cred
CUBE(config-dial-peer)#voice-class sip c?
  call-route  calltype-video  conn-reuse  copy-list

*There is also the registration commands:*
CUBE(config-dial-peer)#voice-class sip registration ?
  passthrough  SIP Registration Passthrough Options

CUBE(config-dial-peer)#voice-class sip registration passthrough ?
  dynamic  SIP Registration Use dynamic Registrar Details (default)
  local-fallback   Local Fallback - (e2e)
  rate-limit   SIP Registration pass-through rate-limit Options
  reg-sync Registration Sync - send REGISTER when registrar up (p2p)
  registrar-index  Registrar Index(s) used for registration passthrough
  static   SIP Registration Use static Registrar Details
  system   Use global registration passthrough CLI setting
  

*I tried using the system passthrough setting, but it did not work.*

*I need to make sure I understand what is actually happening.*

*I don't think the CUBE is even looking at dial-peers for REGISTER
messages. Am I correct?  If so, no amount of dial peer settings is going to
make any difference here... unless there is a way to create a dial-peer
that will intercept REGISTER messages. I believe it is using the REALM
settings in the credentials and authentication strings (that I entered into
sip-ua). And sip-ua is using the global bind settings I set within voice
service voip -> SIP (which are set to the inside interface).*

*Please set me straight!*

Thanks,
Nick

On Wed, May 4, 2016 at 10:37 AM, Sreekanth Narayanan (sreenara) <
sreen...@cisco.com> wrote:

> What IOS version are you running on the CUBE? I can think of a couple of
> things.
> 1. In 15.6(2)T, a new feature has been introduced called multi-tenant
> where you can configure separate voice class tenants. Each tenant can have
> separate authentication mutually exclusive to one another and can be bound
> to different interfaces.
>
> 2. In your current IOS, check if you are able to configure the
> authentication and credential commands at the dial peer level. I am not
> sure which IOS had this introduced but it is worth a try.
>
>
>
> Sreekanth
>
> Sent from a phone.
>
>
>  Original message 
> From: Nick Barnett <nicksbarn...@gmail.com>
> Date: 5/4/16 8:03 PM (GMT+05:30)
> To: Brian Meade <bmead...@vt.edu>
> Cc: Cisco VoIP Group <cisco-voip@puck.nether.net>
> Subject: Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?
>
> I'm binding control and media to my inside interface:
>
> sip
>
>   bind control source-interface GigabitEthernet0/0
>   bind media source-interface GigabitEthernet0/0
>
> I suspect this is the issue... is there any way to make the REGISTER
> messages come from the outside gi0/1 interface?
>
> The reason I'm binding to inside is that we have a a very fluid internal
> network. I have to make and modify internal dial peers almost daily.  When
> I need to create a dial peer and put the bind statements on the dial peer,
> it won't bind properly since there are active SIP calls on the CUBE... so I
> bound it globally. My external dial peers rarely change, so I bind those
> directly to gi0/1 (on the DP).
>
> I was under the impression that REGISTER events can take place without a
> dial peer... but is there a way to, i dunno, make a dial peer for register
> messages?  Can I use SIP profile magic to get it working as is?
>
> I found this article which is pretty much exactly what I'm dealing with,
> but it doesn't mention REGISTER at all...
>
>https://supportforums.cisco.com/blog/154506
> <https://urldefense.proofpoint.com/v2/url?u=https-3A__supportforums.cisco.com_blog_154506=CwMFAg=M-KQspD_LQogCbR-BWCHOaeDEPOhF8vWqHZTaiwxT3c=T9uVLZucbHG2NKKKzOrp-o5cpdReHj02PkJJsCVkgfwcv7S0R5lDeFJg2VRbiNih=UIAzGDQs8RCZld9kCbExwqpJhTgzpDVwM0k8_I7JRqU=jZN-R2pRsZOWN3r5is-aSivDlf9hqddUzDIoOWRWc3E=>
>
>
>
>
> On Wed, May 4, 2016 at 9:06 AM, Brian Meade <bmead...@vt.edu> wrote:
>
>> Do you already have the SIP bind under voice service voip?
>> voice service voice
>>  sip
>>   bind all source-interface FastEthernet0
>>
>> On Wed, May 4, 2016 at 9:58 AM, Nick Barnett <nicksbarn...@gmail.com>
>> wrote:
>>
>>> I've never dealt with an authen

Re: [cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I'm binding control and media to my inside interface:

sip

  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0

I suspect this is the issue... is there any way to make the REGISTER
messages come from the outside gi0/1 interface?

The reason I'm binding to inside is that we have a a very fluid internal
network. I have to make and modify internal dial peers almost daily.  When
I need to create a dial peer and put the bind statements on the dial peer,
it won't bind properly since there are active SIP calls on the CUBE... so I
bound it globally. My external dial peers rarely change, so I bind those
directly to gi0/1 (on the DP).

I was under the impression that REGISTER events can take place without a
dial peer... but is there a way to, i dunno, make a dial peer for register
messages?  Can I use SIP profile magic to get it working as is?

I found this article which is pretty much exactly what I'm dealing with,
but it doesn't mention REGISTER at all...

   https://supportforums.cisco.com/blog/154506
<https://urldefense.proofpoint.com/v2/url?u=https-3A__supportforums.cisco.com_blog_154506=CwMFAg=M-KQspD_LQogCbR-BWCHOaeDEPOhF8vWqHZTaiwxT3c=T9uVLZucbHG2NKKKzOrp-o5cpdReHj02PkJJsCVkgfwcv7S0R5lDeFJg2VRbiNih=UIAzGDQs8RCZld9kCbExwqpJhTgzpDVwM0k8_I7JRqU=jZN-R2pRsZOWN3r5is-aSivDlf9hqddUzDIoOWRWc3E=>




On Wed, May 4, 2016 at 9:06 AM, Brian Meade <bmead...@vt.edu> wrote:

> Do you already have the SIP bind under voice service voip?
> voice service voice
>  sip
>   bind all source-interface FastEthernet0
>
> On Wed, May 4, 2016 at 9:58 AM, Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> I've never dealt with an authenticated SIP trunk before and I'm having
>> some issues. I was wondering if anyone has had a similar experience. I
>> already have 2 SIP trunks from ITSP-1 that do NOT require authentication.
>> These are working fine and have been for years.
>>
>> We are adding ITSP-2 and their SIP service DOES require auth.  I've
>> followed their integration guide (which left a lot to be desired) and their
>> acceptance team is telling me my auth is coming from our private class A
>> address.
>>
>> Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
>> address. They are seeing REGISTER messages sourcing the inside VIP.
>>
>> I was looking around for an auth BIND statement or something like that,
>> but I haven't had any luck. Any pointers?
>>
>> Thanks,
>> Nick
>>
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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[cisco-voip] Authenticating sip trunk to ITSP from CUBE?

[Nick Barnett]

I've never dealt with an authenticated SIP trunk before and I'm having some
issues. I was wondering if anyone has had a similar experience. I already
have 2 SIP trunks from ITSP-1 that do NOT require authentication. These are
working fine and have been for years.

We are adding ITSP-2 and their SIP service DOES require auth.  I've
followed their integration guide (which left a lot to be desired) and their
acceptance team is telling me my auth is coming from our private class A
address.

Our CUBE is in HA with an inside (10.x.x.x) and outside (public) IP
address. They are seeing REGISTER messages sourcing the inside VIP.

I was looking around for an auth BIND statement or something like that, but
I haven't had any luck. Any pointers?

Thanks,
Nick
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Re: [cisco-voip] Cisco Voice Training

[Nick Barnett]

I've never been happy with any Cisco class I've taken. That's 3 (sunset
learning) and I'm not wasting my time or the company's money any more.

I got to attend a Skype class last month and Microsoft's training just
embarrasses Cisco (unfortunately).


However... I highly suggest Alta3's SIP essentials class.

On Mon, May 2, 2016, 3:07 PM Aaron Jenkins  wrote:

> Meant to mention I’m in the U.S.
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Aaron Jenkins
> *Sent:* Monday, May 02, 2016 4:05 PM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] Cisco Voice Training
>
>
>
> Looking for recommendations on Training Facilities to take Cisco Voice
> classes.
>
>
>
> Thanks.
>
>
>
>
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Re: [cisco-voip] CUCM SIP Early Attended Transfers

[Nick Barnett]

My experience is that it isn't going to give ringback to PSTN phone A (step
7) when going back through a CUBE without forcing MTP.

This was called out as a caveat in the SIP integration guide for AT flex
reach via CUBE.
https://www.cisco.com/c/dam/en/us/solutions/collateral/enterprise/interoperability-portal/flexible-reach.pdf



On Mon, Apr 25, 2016 at 1:59 PM, Anthony Holloway <
avholloway+cisco-v...@gmail.com> wrote:

> All,
>
> Does anyone have any experience with CUCM, SIP Phones, SIP Trunks, and
> early attended transfers (AKA hitting the transfer button a second time
> quickly)?  I'm not hearing ringback on the transferee phone.
>
> Components in play:
>
> CUCM 10.5(2)
> CUBE 15.4 (SIP-SIP)
>
> Scenario:
>
>1. PSTN Phone A calls into enterprise
>2. IP Phone B answers
>3. IP Phone B Presses Transfer Key
>4. PSTN Phone A hears MOH
>5. IP Phone B Dials Transfer Target and hears ringback
>6. IP Phone B Presses Transfer Key Again
>7. PSTN Phone A hears silence until call connects then two way audio
>
> I found this reference in the CUCM System Guide on SIP:
>
>
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/sip_msg/line_std/11_0_1/CUCM_BK_SB03D18E_00_sip-line-messaging-standard-1101/CUCM_BK_SB03D18E_00_sip-line-messaging-standard-1101_chapter_00.html#CUCM_CN_E798E831_00
>
> Specifically these two seemingly contradictory sentences:
>
> "The transferee receives a ringback while the target phone is alerting."
>
> "The transferee will not receive a ringback although the target is
> alerting."
>
> Seems like it's just not going to work.  What do you think?
>
> ___
> cisco-voip mailing list
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] Constantly having db replication issues

[Nick Barnett]

Thanks James

Ok, yes, there's a lot in rhosts. They are all identical, and each of them
has forward and reverse lookups.



On Wed, Apr 20, 2016 at 12:39 PM, James Buchanan <james.buchan...@gmail.com>
wrote:

> Hello,
>
> Even though you are not using DNS, do you have DNS servers and a domain
> name configured? If so, you should have forward and reverse entries
> configured for all servers. When you look in Unified Reporting, do you see
> anything about the rhosts under Database Status?
>
> Thanks,
>
> James
>
> On Wed, Apr 20, 2016 at 1:07 PM, Nick Barnett <nicksbarn...@gmail.com>
> wrote:
>
>> Thanks Ryan.
>>
>> We have 3 CCM and 1 TFTP node in each of our two data centers. The main
>> data center is here, and that is where our DRS sftp server (and publisher)
>> is located. Nothing is using DNS right now, all of the servers are entered
>> into CUCM as IP addresses... this cluster has been around for years. It was
>> upgraded from 7.BeforeMyTime to 8.6 to 10.0.
>>
>>
>>
>> On Wed, Apr 20, 2016 at 11:54 AM, Ryan Huff <ryanh...@outlook.com> wrote:
>>
>>> Hi Nick.
>>>
>>> Let me ask you a few things;
>>>
>>> - How is the cluster laid out (how many nodes in the cluster and what
>>> nodes are in which DC)?
>>>
>>> - Are you using DNS and if so, where is the DNS server located and do
>>> you have redundant DNS in both DCs?
>>>
>>> - Where is your DRS server in relation to the cluster publisher (same DC
>>> or no)?
>>>
>>> Thanks,
>>>
>>> Ryan
>>>
>>> On Apr 20, 2016, at 11:09 AM, Nick Barnett <nicksbarn...@gmail.com>
>>> wrote:
>>>
>>> I'm wondering how many others have had as many issues with db
>>> replication? It seems that any time we lose a connection to our 2nd data
>>> center (even a 2 minute MPLS planned maintenance outage causes the issue),
>>> our database synchronization has errors.  After a WAN blip, within an hour
>>> or so, I get a message from RTMT about a subscriber being in "blocked"
>>> state:
>>>
>>> %[AppID=Cisco Database Layer
>>> Monitor][ClusterID=ProdVoiceCluster][NodeID=XXX1]: A change
>>> notification client is busy (blocked). If the change notification client
>>> continues to be blocked for 10 minutes, the system automatically clears the
>>> block and change notification should resume successfully."
>>>
>>>
>>> After that, if I run utils dbreplication status, it will have errors...
>>> so then I run the "repair all" option and it fixes it. Then I'm good for a
>>> few weeks until something else happens that starts the whole cycle over.
>>>
>>> Something else that happens after a WAN blip is that DRS begins to fail,
>>> so we have to restart the master DRS and the subsequent DRS services on the
>>> subs. Am I doing something wrong? Is this normal?
>>>
>>> I'm on CUCM 10.0.1.12900-2.
>>>
>>> Thanks,
>>> Nick
>>>
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>> ___
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>>
>>
>
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Re: [cisco-voip] Constantly having db replication issues

[Nick Barnett]

Thanks Ryan.

We have 3 CCM and 1 TFTP node in each of our two data centers. The main
data center is here, and that is where our DRS sftp server (and publisher)
is located. Nothing is using DNS right now, all of the servers are entered
into CUCM as IP addresses... this cluster has been around for years. It was
upgraded from 7.BeforeMyTime to 8.6 to 10.0.



On Wed, Apr 20, 2016 at 11:54 AM, Ryan Huff <ryanh...@outlook.com> wrote:

> Hi Nick.
>
> Let me ask you a few things;
>
> - How is the cluster laid out (how many nodes in the cluster and what
> nodes are in which DC)?
>
> - Are you using DNS and if so, where is the DNS server located and do you
> have redundant DNS in both DCs?
>
> - Where is your DRS server in relation to the cluster publisher (same DC
> or no)?
>
> Thanks,
>
> Ryan
>
> On Apr 20, 2016, at 11:09 AM, Nick Barnett <nicksbarn...@gmail.com> wrote:
>
> I'm wondering how many others have had as many issues with db replication?
> It seems that any time we lose a connection to our 2nd data center (even a
> 2 minute MPLS planned maintenance outage causes the issue), our database
> synchronization has errors.  After a WAN blip, within an hour or so, I get
> a message from RTMT about a subscriber being in "blocked" state:
>
> %[AppID=Cisco Database Layer
> Monitor][ClusterID=ProdVoiceCluster][NodeID=XXX1]: A change
> notification client is busy (blocked). If the change notification client
> continues to be blocked for 10 minutes, the system automatically clears the
> block and change notification should resume successfully."
>
>
> After that, if I run utils dbreplication status, it will have errors... so
> then I run the "repair all" option and it fixes it. Then I'm good for a few
> weeks until something else happens that starts the whole cycle over.
>
> Something else that happens after a WAN blip is that DRS begins to fail,
> so we have to restart the master DRS and the subsequent DRS services on the
> subs. Am I doing something wrong? Is this normal?
>
> I'm on CUCM 10.0.1.12900-2.
>
> Thanks,
> Nick
>
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>
>
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[cisco-voip] Constantly having db replication issues

[Nick Barnett]

I'm wondering how many others have had as many issues with db replication?
It seems that any time we lose a connection to our 2nd data center (even a
2 minute MPLS planned maintenance outage causes the issue), our database
synchronization has errors.  After a WAN blip, within an hour or so, I get
a message from RTMT about a subscriber being in "blocked" state:

%[AppID=Cisco Database Layer
Monitor][ClusterID=ProdVoiceCluster][NodeID=XXX1]: A change
notification client is busy (blocked). If the change notification client
continues to be blocked for 10 minutes, the system automatically clears the
block and change notification should resume successfully."


After that, if I run utils dbreplication status, it will have errors... so
then I run the "repair all" option and it fixes it. Then I'm good for a few
weeks until something else happens that starts the whole cycle over.

Something else that happens after a WAN blip is that DRS begins to fail, so
we have to restart the master DRS and the subsequent DRS services on the
subs. Am I doing something wrong? Is this normal?

I'm on CUCM 10.0.1.12900-2.

Thanks,
Nick
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