and will also ensure that services remain up and running when
the security update becomes mandatory.”
- Daniel Pagan
From: cisco-voip On Behalf Of Daniel Pagan
Sent: Tuesday, February 11, 2020 2:36 PM
To: Lelio Fulgenzi ; Matthew Loraditch
Cc: voyp list, cisco-voip (cisco-voip@puck.nether.net
is a bit behind and needs to be updated. Hopefully
this buys everyone some time, especially for those supporting a number of
environments.
- Daniel Pagan
From: Daniel Pagan
Sent: Tuesday, February 11, 2020 2:33 PM
To: Lelio Fulgenzi ; Matthew Loraditch
Cc: voyp list, cisco-voip (cisco-voip
It does not appear Microsoft will be enforcing LDAP over TLS with this upcoming
patch. While the original plan was indeed to tighten this up, it seems this
requirement is being delayed until after Q2 of the year.
The advisory was updated February 4th and shows:
Windows Updates in March 2020 add
In your example, the SERVER2 certificate in phone-vpn-trust is there because
someone would have placed it there for some reason. Some additional info...
certificates uploaded to the phone-vpn-trust store can be associated with a VPN
gateway in /ccmadmin. When assigned to a VPN-enabled phone
Some of you might be interested to hear that Cisco has announced a new method
of renewing one's expert-level certification. While taking a written exam is
still an option, those needing to renew their CCIE/CCDE will be allowed to
enroll themselves in select classes, online and in-person, which
Just adding my experience to this…
I agree and can attest to the stratum-1 server caveat below. After some time,
the NTP client can get blocked and force you (the general “you”) to update your
entries in the near future. Of course multiple NTP entries can be configured
but if you’re doing
ly pushing for creation
of a new defect and can share that ID if you feel it would help.
Hope this helps.
Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Thursday, February 9, 2017 10:28 AM
To: cisco.voip <cisco.v...@verizon.net>; cisco-voip
I have heard of this but have yet to see it from a trace perspective. For
troubleshooting, my first step would be to determine if the problem is CUCM or
the IP Phones that don't stop ringing. The call is sent to an 8851/61 using
INVITE, which means CUCM should force them to stop ringing on
ll suggest, if possible, to address the main
problem of your session expiration.
Hope this helps.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Friday, September 23, 2016 4:12 PM
To: Norton, Mike <mikenor...@pwsd76.ab.ca>; cisco-voip
I’ve heard of this as well and actually have an open case that will require a
rebuild of CUC in order to add an additional vCPU. From what I recall, it can
be done but Unity won’t utilize the additional core unless it’s detected during
the install process. CUC, from what I know, is the only UC
According to the RFC, the Allow-Events header is specifically used to convey
events that a UA can support using the SUBSCRIBE method. Typically we’ll see
KPML or Presence as a supported Allow-Event value, which makes sense since both
of those events are initiated through SUBSCRIBE/NOTIFY
Personally, I would push back to them and simply provide them with the request
URI your transmitting, which they should be using for routing purposes.
INVITE sip:1343@10.11.0.9:5060 SIP/2.0
I say push back because a 404 is a very straight forward response and sent for
one reason. If their
Had no idea this XML generator was available. I’ve been grabbing sample scripts
off Github and customizing them in the lab for testing purposes so this should
supplement that setup very well. I just tested it for a few minutes and it
seems like it replaces device-specific fields (via, contact
Comments below. Hope this helps.
- Danend attach-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Dave
Goodwin
Sent: Tuesday, December 01, 2015 8:41 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] advice on upgrading large CUCM cluster with CoW from 8.6
Looking at the SIP transactions in a parser that doesn't display header fields
(it doesn't even have the start-line) really limits how much information one
can gather from the output. The REFER should have either a Replaces: header
field or, most likely, a Refer-To: header field, which should
Just a heads up, the Use Forward Settings of Line Group Member should use the
forward no coverage settings of the DN forwarding to the Hunt Pilot. No need to
have an active line member in the LG. Just ran a quick test to confirm - the
only member of a LG was logged out with the device
the Call Forward No Coverage (CFNC) settings for
the original called number that forwarded the call to this hunt pilot.”
- Dan
From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, November 23, 2015 11:38 AM
To: Daniel Pagan <dpa...@fidelus.com>; rschukne...@gmx.de;
cisc
Set the call handler to send voice messages to a specific user with mailbox.
Configure that CUC user to relay voice messages (not accept and relay) to an
SMTP address in the Message Actions config page. This assumes you have an SMTP
server configured in the SMTP Smart Host page. Also the user
Not to mention further impact caused by a “can’t break what’s broken” approach
without having full knowledge of additional dependencies. I would add a third
to your list:
3) Do I understand all the dependencies of this service I’m restarting, trunk
I’m resetting, server I’m rebooting, or
On the topic of NTP and upgrades/migrations, I would advise to also watch out
for CSCur94973.
https://tools.cisco.com/bugsearch/bug/CSCur94973
- Dan
-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Anthony Holloway
Sent: Monday, November 16, 2015 11:20 AM
Sreekanth: Really enjoyed your LUA scripting video.
Ahmed: Adding to this, you can use the Call-ID header for identifying a
specific SIP call flow in the output of debug ccsip messages. The Call-ID
header field is a unique identifier for the SIP dialog, which means the Call-ID
header field for
If you’re certain the hold request is coming from IM, and the hold request is
in reference to the call reference (CI) associated with the call in question,
then my suggestion would be to open a TAC case. At this point, assuming this
finding is applicable to the problem at hand, and end-user
Adding to list of possibilities, it could also be an unaccepted transport
type... E.g. CUCM now sending SIP requests via TCP when UDP inbound is
configured on Exchange.
Also if require MTP is configured but none are available, in which case I’d
check MRM in SDL traces.
But my guess would be
the
local ringer, but if the issue is a delay in *starting* the ringer, then I
agree this 487 certainly shouldn’t be applicable.
Dan
From: avhollo...@gmail.com [mailto:avhollo...@gmail.com] On Behalf Of Anthony
Holloway
Sent: Wednesday, September 30, 2015 11:27 PM
To: Erick Wellnitz
Cc: Daniel Pagan
Interesting… Out of curiosity, there should first be a CANCEL to the IP phones
where the call wasn’t answered. The phones should then 200 that CANCEL request,
and then send the 487 final response to the original INVITE for the call. Do
you see the 487 final response six seconds after the
cucm <-- 487 with CSeq: 101 INVITE <-- phoneB (one response per
phone)
cucm --> ACK --> phoneB
cucm --> ACK --> phoneC
Hope this helps. I’m pretty interested in knowing what you or TAC finds.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf O
I would also suggest debugging the signaling protocol being used for calls
to/from this analog port or reviewing CCM SDI/SDL traces. If these are MGCP
controlled, try to determine if CUCM transmitted a DLCX without first receiving
a NTFY for the on-hook event (O: L/hu). Seeing a NTFY with O:
No problem. The solution you used was described in option #2 of my email.
- Dan
end attach-
From: abbas Wali [mailto:abba...@gmail.com]
Sent: Monday, September 14, 2015 8:36 AM
To: Daniel Pagan <dpa...@fidelus.com>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip]
The request-URI in this case is the PKID of the DN followed by the IP address
of the phone. Check it out:
https://10.10.13.100/ccmadmin/directoryNumberEdit.do?key=ae97ba4b-2397-4b52-5ddc-07589fb380ab
^^^URL of a test DN.
Key= ae97ba4b-2397-4b52-5ddc-07589fb380ab
INVITE
configuration file:
ae97ba4b-2397-4b52-5ddc-07589fb380ab
Hope this helps.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Monday, September 14, 2015 10:44 AM
To: Mark Holloway <m...@markholloway.com>; voip puck
<cisco-voip@puck.nether.net>
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/8x/gui_reference/guide/8xcucgrgx/8xcucgrg060.html#pgfId-1052219
Check out table 6-9, specifically the “Callers Hear” section. This doc talks
about Call Handlers but the “Callers Hear” section is applicable to CUC users
and should
I don’t know why this failed to connected when reading the release notes. Quite
the brain fart… No OS = no SFTP = local mounting of ISO.
- Dan
From: NateCCIE [mailto:natec...@gmail.com]
Sent: Monday, September 14, 2015 4:42 PM
To: Daniel Pagan <dpa...@fidelus.com>
Cc: Justin Steinberg &l
I’m digging up this old thread once again.
With PCD v11 released, it’s great to see that remote SFTP sources can now be
used, but it’s also disappointing to see it restricted to upgrades only. Does
anyone know if there are plans to support migrations using a remote SFTP
server? Is the issue of
Accomplish this with the Accept and Relay option for user message actions for
VM1 and specify VM2’s SMTP address. This should keep the message available for
VM1 while forwarding a copy to VM2. You’ll need to setup CUC with a SMTP smart
host in order to relay messages, and will likely need to
My first step would be to find out the direction of the 500 final response. I
would run a ccsip messages debug on CUBE and recreate the issue for determining
where the 500 final response is being generated. The User-Agent header should
tell you what device is generating it. My second step would
Hey Ryan! Hope you’re well ☺
Just wanted to add here that not supporting early offer should result in one of
two things – either local ring back would be generated on the calling device
and/or there will be no audio in scenarios where audio cut-through is required
before the 200 final response
“If a device pool doesn't have anything setup for Standard Local Route Group,
when someone from that pool calls a route pattern referencing this Route List
will it just smoothly go to option #2 or is there some gotcha i'm not grasping?”
Yes - CUCM attempts to get information on the SLRG once
I personally can't speak to 3rd party FXS gateways, but I've worked with
customers in the past whose VG224s were integrated w/ CUCM via SIP trunk. The
two caveats that immediately come to mind are applicable to a non-Cisco FXS
gateway using SIP as well:
1) Lack of SCCP supplementary
voip
fax-relay ecm disable
For MGCP
no mgcp fax t38 ecm
Hope this helps.
- Dan
From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Wednesday, August 19, 2015 10:53 AM
To: nh...@co.fresno.ca.us; Daniel Pagan dpa...@fidelus.com;
norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net
Subject
.
Hope this helps.
- Dan
On Aug 18, 2015, at 5:43 PM, Daniel Pagan dpa...@fidelus.com wrote:
there’s no way to inject the CNG tone
___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
that you won't get the same
results using any type of fax relay since PCM audio isn't being transmitted,
but the flow remains the same.
Hope this helps.
- Dan
-Original Message-
From: NateCCIE [mailto:natec...@gmail.com]
Sent: Tuesday, August 18, 2015 6:45 PM
To: Daniel Pagan dpa
That “pre-tone” you’re referring to is called the CNG tone, assuming you hear
it every 3 seconds on your calling fax machine. It’s basically what tells the
remote end “I’m a fax machine” but many devices no long wait to hear this
before attempting to start negotiating capabilities. However, it
In 8.6 and 9.x it’s five. Limit of sync agreements was increased to 20 in CUCM
10.5.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Techguy
Sent: Thursday, August 13, 2015 1:27 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Ldap
Is there a cap on the
Use the timestamp from audit logs and review CCM SDI detailed traces. If set to
detailed, there should be an entry for the DB change notification mentioning
what was changed. It'll either give you a before after PKID or a before
after numerical value depending on what was modified.
- Dan
Most if not all of the system default prompts are stored in the local
filesystem and can’t be accessed without root permission. These prompts (two
different files play to form that message) are two of those /PHGreet/ .wav
files and cannot be changed. Unfortunately I’m unware of any method for
On occasion I’ll also review the Tomcat logs for identifying who accessed a
specific portion of CCMAdmin, then use that information to begin digging deeper
in other trace files. You can try collecting Tomcat logs off all nodes (because
we don’t know which node was accessed via HTTP), and search
straight IP address and not a FQDN for
comm.Since I did this patch's I am going to let things bake and see if the
error(s) returns.
I will keep my eyes on RTMT too see how things behave.
-Mike
Michael T. Voity
Network Engineer
University of Vermont
(802) 656-8112
On 7/14/2015 9:29 AM, Daniel
Is your Nortel PBX sending a FQDN in the Contact header? This is important
because in 10.5(2) CUCM performs a SRV and A Record lookup on FQDNs contained
in a SIP Contact header. If this lookup fails, then expect to see a CANCEL
followed by a BYE. I encountered this a few times over the past few
]
Sent: Tuesday, July 14, 2015 9:49 AM
To: Daniel Pagan; voip puck
Subject: Re: [cisco-voip] Nortel 81C / CS1000 SIP Trunk to CUCM 10.5.2
Hi Dan,
Last night I migrated to 10.5.2.SU2 and then this morning applied the
ciscocm.FQDNwithDNS-v1.0.k3.cop.sgn patch.
On both CUCM and CS1000 we are using
LDAP authentication is used by Tomcat and isn’t just restricted to the
Publisher server - Subscriber nodes handle this as well. DirSync is specific to
synchronization of LDAP attributes and only runs on the Pub, so synchronization
would definitely be affected if the Publisher is offline. I
to
7.x as far as I recall.
Hope this helps
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Monday, July 06, 2015 9:45 AM
To: Lelio Fulgenzi; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] LDAP Authentication when CUCM publisher is down
I know it's not an exact match but it's very similar to a dialing forest dump
procedure off a SCCP handset:
https://supportforums.cisco.com/document/97751/how-dump-ccm-memoryimdb-diagnostic-information-traces
I personally haven't used the **##**32 you mentioned though.
- Dan
From: cisco-voip
I'm also attending.
Anthony: Nice! I'll be sure to do my best to stop by and check it out.
Stephen: Looking forward to checking out the applications on demo.
- Dan
Sent from my mobile device.
On Jun 8, 2015, at 12:10 PM, Anthony Holloway
Do you see a StartMediaTransmission being sent to your MOH server? Look for
MohDControl for SCCP signals to the MOH server - one process instance is
created per MOH server. But first, before sending the StartMediaTransmission
signal to MohDControl, CCM will first attempt to get media
/local/cm/moh/ringback.ulaw.wav)
Of course CUCM won’t get to this point if the issue resides at the signaling
protocol and media setup aspect of the call. Regardless, hope this helps.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Friday, May 15
Don't think it does but rather the other way around. In this scenario, the
called phone pressing park is going to need CSS access to a call park range
sitting on the node where the gateway is registered, but I don't think the
gateway CSS gets used to allocate that park DN. That's assuming
I’m digging up an old thread here (perhaps because of tender pain points) but
the point you made about multi-site is spot on, Dennis… especially if the
remote location’s connectivity to the PCD’s site is less than ideal. In this
type of scenario, one can encounter issues with PCD where a
that with a
WinSCP session into PCD (browse through /export sub directories…) and a person
can really get more detailed information than what’s offered in documentation.
- Dan
From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com]
Sent: Saturday, April 11, 2015 11:49 AM
To: Daniel Pagan; Heim
Adding to Ryan’s suggestion re: ACKs - I took a snip of some internal
documentation I wrote re: MGCP debugs that might help. The MGCP transaction ID,
in a CCM SDL trace or MGCP packet debug output, would look similar to this:
http://i.imgur.com/O1YGnUO.png. Just showing the transaction ID
But to answer your specific question, I haven't seen this issue being directly
related to a 10.5 upgrade.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Tuesday, March 31, 2015 9:43 AM
To: Adam Frankel (afrankel); Lisa Notarianni; cisco-voip
and Cc signals
Some more detail on this issue in my previous messages.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Tuesday, January 13, 2015 12:10 PM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ParkingLotD and SNR | Leaked Calls
Adding to Neal’s comments, you might find the following documentation helpful:
Cisco Live: Best Practices for Migrating to CUCM 10.5
https://clnv.s3.amazonaws.com/2014%2Fusa%2Fpdf%2FBRKUCC-2011.pdf?Expires=1427481707AWSAccessKeyId=AKIAJO3XQSJMRXKWDHZQSignature=ZqJonKRBn4Tlw9mXHoxg7d9RUUY%3D
The
Hmmm… Subject line says “from 8.5” but message body says “from 8.6”. Perhaps
he’s referring to 8.5?
If you are then keep in mind the upgrade will require not only the
version3-keys.cop file, but you’ll need to install the refresh_upgrade file as
well. See the ReadMe:
Agreed – To add to Brian’s message, the only benefit to having this powered
down standby IMO would be minimizing time to restore should the Publisher ever
need to be rebuilt in a DR situation. If you had a VM sharing the same IP
address/hostname/cert information as the existing Publisher and
never receives a 200 Final Response, nor is there a 183 w/ SDP for early media,
I would imagine because we never reached the alerting state in CUCM.
Dan
-Original Message-
From: gen...@ucpenguin.com [mailto:gen...@ucpenguin.com]
Sent: Thursday, March 19, 2015 3:18 PM
To: Daniel Pagan
Cc
The main question I have... if CUC is being used simply to hang-up on the
calling party, what's the purpose of needing this migrated to CUCM instead of
simply leaving the number unallocated? Correct me if I'm wrong, but it seems to
me that you're specifically looking for a method in CUCM where
...@ccgs.wa.edu.au]
Sent: Wednesday, March 18, 2015 11:38 AM
To: Daniel Pagan; Ryan Huff; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Extension that hangs up on the user?
Hi Dan,
This is to transfer to from an Exchange UM AA where we want to hang up on the
caller after playing a message (it's summer
Ahmed -
Not sure if I completely follow but are you concerned about the order of
operations for digit manipulation in IOS? If you are then I would say your
expectation would be correct with digit manipulation in IOS occurring in the
following steps and in the following order:
incoming
For clarity, by “higher bandwidth codec” I meant to say higher bit-rate codec,
or codec of higher bandwidth consumption.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Monday, March 16, 2015 9:08 PM
To: Ryan Huff; cisco-voip@puck.nether.net
Hey Ryan how’s it going? Transcoder allocated by CUCM comes from the side using
a higher bandwidth codec, regardless if it’s the calling or called party, with
the intention to avoid streaming a high bandwidth consuming codec over a WAN
connection – keeping it local to the LAN. Of course, this
:01 AM
To: Daniel Pagan; Dave Goodwin
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] vMotion w/o Shared Storage
Daniel,
Could you give a little more detail about your experience with this process?
The confusion you faced is likely the same confusion many of us would face.
Which document
...@outlook.com]
Sent: Tuesday, March 10, 2015 5:08 PM
To: Daniel Pagan; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] vMotion w/o Shared Storage
Shutdown, yes. Running, no.
Thanks,
Ryan
Original Message
From: Daniel Pagan dpa...@fidelus.commailto:dpa...@fidelus.com
Sent
, which is definitely Cisco supported.
Some confusion on my part between the two migration methods but all is clear.
Thanks!
- Dan
From: Dave Goodwin [mailto:dave.good...@december.net]
Sent: Tuesday, March 10, 2015 8:47 PM
To: Daniel Pagan
Cc: Ryan Huff; cisco-voip@puck.nether.net
Subject: Re
continue
In CUCM... DN 56789 - CFwdAll to Voicemail - Unity CH or User w/ mailbox
Add the partition of 56789 to the CSS of the UCCX CTI ports performing the
redirect.
- Dan
From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Tuesday, March 10, 2015 12:23 PM
To: Daniel Pagan; norm.nichol
user based on
first redirecting number, which might use the CTI port performing the
redirect... Not sure... would need to test... but it's probably the cleanest
solution IMO.
From: Daniel Pagan
Sent: Tuesday, March 10, 2015 1:11 PM
To: 'Ryan Huff'; norm.nichol...@kitchener.ca; cisco-voip
For routing based on ANI, Ryan Huff's suggestion will certainly help from the
perspective of CUCM. You can use this alongside CUC routing rules - route calls
from the 519 area code to a specific mailbox. If you know these callers from
area code 519 are dialing the same DNIS, then routing by ANI
Quick question...
vMotion of a shut down UCM between two hosts without shared storage using
vCenter. Is this supported? The virtualization document for UC platforms says
vMotion is supported on shared storage, so I figured to ask. Is this migration
method supported?
- Dan
But this doesn’t provide the logic required for the call acceptance rule of
“accept only the first (regardless of ToD), reject all others, reset after 24
hours”. Aside from using some form of UCCX scripting or using a custom app
integrated through the JTAPI or Routing Rules API, I too don’t see
Adding to Dennis's question below, is CUCM v11 going to be discussed at Cisco
Live in June? I see no sessions or breakouts about this. There's a few sessions
on CUC and UCCX roadmaps but nothing similar with regards to UCM.
Thanks!
- Dan
-Original Message-
From: cisco-voip
I’d also say a similar defect would be CSCul71689 – a defect I came across
quite often on 8.6(2). Like Brian said, hunt lists use RouteListControl which
creates HuntListCdrc for call distribution to your line group members. With
either defect, I’d say another possible workaround would be to
Jonathan:
I’ll zip and email a copy to you directly. What I don’t have however is
documentation on the tool. I was given a wiki URL that doesn’t work.
- Dan
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Jonathan Charles
Sent: Thursday, March 05, 2015 12:07 PM
To:
, 2015, at 10:18 PM, Daniel Pagan
dpa...@fidelus.commailto:dpa...@fidelus.com wrote:
I’d also say a similar defect would be CSCul71689 – a defect I came across
quite often on 8.6(2). Like Brian said, hunt lists use RouteListControl which
creates HuntListCdrc for call distribution to your line group
Just to add to Anthony’s email, I suggest making sure to add to the vCPU socket
count, not number of vCPU cores per socket. Also note that you can increase
your vCPU socket count but should *not* decrease it – this is not supported.
- Dan
From: cisco-voip
After re-reading my email, perhaps my words were a bit harsh on that section of
the SRND :) but I do think the timer statement might need to be revisited and
possibly rewritten.
Good weekend to all!
- Dan
On Feb 6, 2015, at 5:38 PM, Daniel Pagan
dpa...@fidelus.commailto:dpa...@fidelus.com
://andrewjprokop.wordpress.com/2013/07/02/understanding-sip-timers-part-i/
Hope this helps.
- Dan
-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Daniel Pagan
Sent: Thursday, February 05, 2015 2:33 PM
To: gen...@ucpenguin.com; cisco-voip
If we're talking about transport level timeout, it looks like the command is
available in CUBE SP Edition:
In addition to the SIP protocol-level timers, Cisco Unified Border Element (SP
Edition) also allows modification of transport-related timer commands:
tcp-connect-timeout (how long TCP SYN
The DNS assignments make up part of the hash used to generate the license MAC
address in UCM 8.x. If you need to update the DNS entries, my recommendation
would be to simply:
1. Document the current license MAC address. Would also be helpful to to
have your Cisco TAC support contract
Quick correction - I should have said the primary DNS server assignment and
not simply DNS assignments, which after reading my message a 2nd time might
imply that any DNS entry change would regenerate a new license MAC.
Hope this helps.
- Dan
From: Daniel Pagan
Sent: Monday, January 26, 2015
or set the voicemail policy to Timer Control.
- Dan
From: Daniel Pagan
Sent: Tuesday, January 13, 2015 11:53 AM
To: cisco-voip@puck.nether.net
Subject: ParkingLotD and SNR | Leaked Calls on StationD
Folks:
Hoping someone can tell me if and how ParkingLotD and ParkingLotCdpc are
related
Folks:
Hoping to get some insight on SDL process creation for H245...
Scenario is three CUCM clusters communicating over ICTs. Call is routed from
Cluster-1 to Cluster-2... then Cluster-2 to Cluster-3. Cluster-3 sends the H245
address port info via H225 ALERTING to Cluster-2, which then sends
Thanks Ryan - that's what I was hoping to hear. I'll try to set this up in a
lab to confirm with some simple ACLs.
- Dan
From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: Monday, December 01, 2014 5:33 PM
To: Daniel Pagan
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip
] On Behalf Of Brian Meade
Sent: Monday, December 01, 2014 5:51 PM
To: Ryan Ratliff (rratliff)
Cc: Daniel Pagan; cisco-voip voyp list
Subject: Re: [cisco-voip] H245Interface Related Processes
Also should be fairly easy to capture via a packet capture on Cluster 1 if this
is easily reproducible
Funny it certainly is. Reminds me of a quirky issue we found a while back. Had
a customer who opened a case saying his phone was a vampire. The customer said
his phone calls were being disconnected by sunlight. His phone was positioned
on a desk where direct sunlight would hit the phone at
To: Daniel Pagan; Jason Aarons (AM); Jeffrey Girard; Ryan Ratliff (rratliff)
Cc: cisco-voip voyp list
Subject: RE: [cisco-voip] 7945 more button does nothing under ITL (can't delete
the ITL)
So the sunlight only happens with CUCM 8.5(2).
Dennis Heim | Collaboration Solutions Architect
World Wide
If you absolutely can't have any log files older than seven days on disk, one
option would be to configure and schedule trace archiving for all services and
applications, but make sure the delete log files from the server option is
enabled.
This would provide you with two things:
1.
Adding to Wes's comments for managing consumed disk space on an archive
server... For a Windows OS server, the forfiles command with a few options can
help with the disk space cleanup as well:
http://stackoverflow.com/questions/51054/batch-file-to-delete-files-older-than-n-days
But to the
Few questions I would try to answer here:
Is CUCM negotiating Early Media successfully on these problem calls?
Do I see a StationD event for AlertingTone to the IP Phone?
CUCM should instruct the SCCP phone to play local ring back via AlertingTone
event when early media cannot be negotiated.
This isn’t bulk export, but Message Archiver allows you to export voice
messages for a Unity Connection user/subscriber to a compressed file.
Tool:
http://www.ciscounitytools.com/Applications/CxN/MessageArchiver/MessageArchiver.html
Tool Help:
Folks:
I'm hoping someone can share their experience with the Cisco recommended method
for removing EWS limits on Exchange 2010 SP2 RU4 and higher. In earlier
releases of Ex2010 the process of setting a throttling policy applied only to
the UM service account, and any throttling performed
[mailto:jsteinb...@gmail.com]
Sent: Thursday, September 11, 2014 11:08 AM
To: Daniel Pagan
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] EWS Limits Throttling Policy
I've successfully used the 'paged view functionality' on the later versions of
Connections that works around this issue
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