fied-contact-center-express/118823-technote-uccx-00.html
>
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RTCP) due to access list not permitting it. Is there a doc
somewhere that shows different/expanded range of ports that Cisco phones
will use?
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> Email: ryanthomash...@outlook.com
>
> Spark: ryanthomash...@outlook.com
>
> Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff>
>
> LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff>
>
> Web ryanthomashuff.com
>
>
> -
d a phone model
> firmware that thought 32768 was the upper bound. I personally, have always
> gone with 32766 as being the upper bound, but then again, I've never
> created an ACL for this range either, so it hasn't presented itself as a
> problem thus far.
>
> O
gateway was receiving RTP on?
>
> On Tue, Jan 19, 2016 at 12:55 PM, Ed Leatherman
> wrote:
>
>> Come to think of it, the session was between an IOS MGCP gateway and the
>> 8945 - perhaps something the gateway is trying to send to the phone that I
>> dont reali
xtend
the persistent call. When a call is presented, it tries to hit my remote
number but doesn't actually connect. UCCX is send to enable the persistent
connection.
Only difference I can tell is I don't have IMP running - seems odd if that
is required, what does it use that for?
UCCX 10.6
be comical, in light of your reply.
>
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ork as advertised. IM&P wasn't necessary.
On Tue, Jan 26, 2016 at 7:04 PM, Brian V wrote:
> In my experience you don't need IM&P to make extend and connect work.
>
> On Tue, Jan 26, 2016 at 2:44 PM, Ed Leatherman
> wrote:
>
>> Trying to get this feature workin
g a little cautious about unintended consequences.
IPMA
Cisco Web Dialer
Jabber - Click to call, what other cases?
Remote Destinations
Is there a rule that gets checked to see when these rules get applied?
--
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___
cisco-voi
tocol pass-through g711ulaw
sip
pass-thru content sdp
sip-profiles 100
!
dialpeers all have
!
dtmf-relay rtp-nte
codec g711ulaw
no vad
Thanks!
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of the chatter
> CUCM will send to CUBE that really has no business going out to the ITSP,
> but keeping the important messages, such as mid-call media changes.
>
> voice service voip
> sip
> midcall-signaling passthru media-change
> !
>
> As far as your 4K router and t
similar command you might want to run is to suppress all of the chatter
> CUCM will send to CUBE that really has no business going out to the ITSP,
> but keeping the important messages, such as mid-call media changes.
>
> voice service voip
> sip
> midcall-signaling passthru m
nd connect to a live agent.
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experience, turning up SIP services is like peeing an onion ...*".
>
> All in good fun Anthony ;)
>
> Sent from my iPhone
>
> On Feb 9, 2016, at 12:12 PM, Anthony Holloway <
> avholloway+cisco-v...@gmail.com> wrote:
>
> I
Diego) - 2 Hours
> <https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=83773&backBtn=true>
>
> On Tue, Feb 9, 2016 at 11:28 AM, Ed Leatherman
> wrote:
>
>> When I first read it I thought he just meant he hated it that much :)
>>
>> A
ributing or using the
> information. Please contact the sender immediately by return email and
> delete the original message.
>
>
>
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y run out of Memory and Disk
> Space soon.
>
> Hate to report this here, but this one is a doozy.
>
>
> --
> Baha
>
>
> On Feb 25, 2016, at 6:46 AM, Ed Leatherman wrote:
>
> We just patched vsphere this past weekend and ran into the selinux bug
> with vmtoo
a problem.
My first thought is to script something that will do a sql query against
the end user table and send back user ids that are inactive - on a schedule
so it just shows up in our inboxes in the morning. Anyone ever had to do
this, is there a better way?
--
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___
s. In the above example the agents time is queue is showing blank
> and when I debug the Time in Queue is set at “-1”
>
>
>
> TAI
>
>
>
> Matt
>
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I haven't touched this in 2 years - here's notes I took when I was playing
around with this before in case it will help:
http://www.evernote.com/l/AAVCJ8P2UbJG5o0yqjxIO6-9p7edydMHCfE/
On Fri, Apr 8, 2016 at 8:27 AM, Ed Leatherman
wrote:
> Bill,
>
> Does your SP use tempor
ss it.
Thanks!
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I was able to execute a stored
> procedure which wasn't creating a temp table. Thanks again for sharing Ed.
>
> On Fri, Apr 8, 2016 at 7:27 AM, Ed Leatherman
> wrote:
>
>> Bill,
>>
>> Does your SP use temporary tables? look at CSCul06940 - I dont know that
&
"Non-Stock" for the
>>report I just imported
>>21. Within that new folder, which is the name of the report "Reason
>>Code Report by Agent Grouping" the report itself is in there, with the
>> same
>>name "Reason Code Report by A
CX to see if that helped, and now I cannot run
>>> the Reason Code Report by Agent Grouping report, as it fails with the same
>>> database error. So, it got worse.
>>>
>>> On Fri, Apr 8, 2016 at 9:44 AM, Ed Leatherman
>>> wrote:
>>>
>>&g
es for the reports will be uploaded soon. Please let me
> know if you have any questions.
>
> Regards,
> Abhiram Kramadhati
> Technical Solutions Manager, CCBU
> CCIE Collaboration # 40065
>
> From: cisco-voip on behalf of
> akramadh
> Date: Monday, 11 April 2016 at 10:41 AM
>
04 PM, Aaron Jenkins
wrote:
> Looking for recommendations on Training Facilities to take Cisco Voice
> classes.
>
>
>
> Thanks.
>
>
>
>
>
>
>
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t;> _______
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hanks!
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> CloudPOP/InvictaCloud
> www.cloudpop.com
>
> *“Go beyond your limits, push yourself, be the best you can be.*
> *Experience new cultures, broaden your horizons, stay connected.”*
>
>
>
> ___
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; World Wide Technology, Inc
>
>
>
>
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Ed Leatherman
> *Sent:* Tuesday, May 17, 2016 8:53 AM
> *To:* Cisco VOIP
> *Subject:* [cisco-voip] SIP door phone with video camer
n jabber's name cache. It's a VP level so I don't have
immediate access to just go wipe it out and try it.
I can't reproduce the issue calling the same number myself, so it appears
to be local to him.
Thanks!
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ther.net/mailman/listinfo/cisco-voip
>>
>>
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es-cbc,blowfish-cbc". Just to err on the side of caution I
> would add a few of the ciphers that UC Manager is looking for.
>
>
> Hope this saves some pain,
>
> = Ryan =
>
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> <http://www.dimensiondata.com/Global/Policies/Pages/Email-Disclaimer.aspx>
>
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>
>
O
> UnifiedFX
>
> Come along and meet the UnifiedFX team
> World of Solutions stand T2
>
>
>
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>
>
t;>>
>>>
>>> Any comments, suggestions, mild criticism appreciated.
>>>
>>>
>>> Aaron
>>>
>>> _______
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>&g
gt; *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
>> Behalf Of *Anthony Holloway
>> *Sent:* Thursday, June 16, 2016 1:08 PM
>> *To:* Cisco VoIP Group
>> *Subject:* [cisco-voip] CUCM 11.5 Upgrade from 11.0 Requires PUT?
>>
>>
>>
&g
ng Carrier B every day to figure it out, what magic is
involved behind the scenes here? I miss BGP.
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find the route and remove it.
>
>
> = Ryan =
>
>
>
> Email: ryanh...@outlook.com
>
> Spark: ryanh...@outlook.com
>
> Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff>
>
> LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff>
> something outside of the provider's provisioning tools and it was never
> documented.
>
> -Adam
>
>
> On Friday, June 24, 2016, Ed Leatherman wrote:
>
>> Good morning,
>>
>> Is anyone familiar with the process of porting telephone numbers between
>
>>>> rebooted after as well.
>>>>>>
>>>>>> Seems like with the standard requirement of rfc2833 on SIP trunks to
>>>>>> the ITPS, and CTI apps in the network (I'm looking at you UCCX), MTPs
>>>>>> play
>>>>>> a large role in the success of SIP trunking for customers, and yet I
>>>>>> cannot
>>>>>> even register them locally with the LTI.
>>>>>>
>>>>>> I do have a fallback plan, so I'm not stuck. I'm just looking for
>>>>>> the optimal design scenario. In my order of preference I would like to
>>>>>> go:
>>>>>>
>>>>>> 1. LTI
>>>>>> 2. SCCP via Telephony Service
>>>>>> 3. SCCP via CUCM
>>>>>>
>>>>>> Would you rank them differently?
>>>>>>
>>>>>> Thanks for your input in advance.
>>>>>>
>>>>>> ___
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>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>
>>>>>>
>>>>>
>>>
>>
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r the call around and I'm trying
to nail down everything I find that i'm not sure of... in one of the
transactions the telephone-event is missing out of the allow-events in the
invite.
Thanks!
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I know RTP-NTE
> is negotiated and have found multiple instances where DTMF was negotiated
> successfully (to RTP-NTE) without including telephone-events in the
> Allow-Events header.
>
>
>
> What type of issues are you seeing with ReINVITEs?
>
>
>
> *From:* cisco-voi
North Carolina Public Records Law and may be disclosed to third
> parties.
>
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ll wrote:
> Anyone seen feature is unavailable when trying to forward all calls?
>
> 8861 on CM 10.5.2
>
> TIA
>
> Scott
>
>
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> https://puc
r too much) and I'm
not getting what exactly this command does.
Anyone know how that asymmeteric command works?
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l legs for our NTE PT.
>
>
>
> Note that CUBE has no issues receiving one dynamic PT for NTE and sending
> another (ex: receiving PT 100 and transmitting 101 for RTP-NTE) on the same
> call leg.
>
>
>
> Hope this helps
>
>
>
> - Dan
>
> end atta
in the middle.
>
> On Mon, Jul 18, 2016 at 5:08 PM, Ed Leatherman
> wrote:
>
>> Thanks Daniel, that helps a lot in understanding the feature. I'm curious
>> if CUBE will also translate digits to KPML in this case if the leg to CUCM
>> has that negotiated. Wish I
go back to the original call and it comes
> back and just keeps ringing.
>
>
>
> Anyone seen anything like this? CM 10.5. expressways 8.7
>
>
>
> TIA
>
>
>
> Scott
>
>
>
> ___
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> ci
supported yet. Fix for us was to
> disable recording config for just the line appearance on MRA Phone and
> Mobile Jabber - TCT, BOT, TAB. Internal desktop Jabber should work - CSF.
>
>
> Patrick Robitaille
> AQR Capital Management
> O: (203) 742-3797 | C: (203) 914-9572
>
>= '2016-01-27 00:00:01'
>> (COUNT(*))
>> --
>> 20519
>>
>> But CUIC is only pulling in the new calls. Any ideas how to fix this?
>>
>> Thanks,
>>
>> Brian
>>
>
>
>
row
> REPLID: 131076 / 0x20004
> REPLMODE: PRIMARY ON:g_uccx106pub_uccx
> APPLY-AS: INFORMIX ON:g_uccx106pub_uccx
> REPLTYPE: Master
>
>
> On Wed, Sep 28, 2016 at 7:58 AM, Ed Leatherman
> wrote:
>
>> I'm troubleshooting UCCX/CUIC
_
>> 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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the
> 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 materi
gt; Phone: 216-426-3535 x1104
>
> 1457 East 40th Street | Cleveland, OH 44103
>
> bam...@humanarc.com | www.humanarc.com
> <http://www.humanarc.com%20>
>
> [image: cid:image001.gif@01CEF0F8.9227CA80]
>
>
>
> *From:* Ed Leatherman [mailto:ealeather...@gmail.com]
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le...@uoguelph.ca
> www.uoguelph.ca/ccs
> Room 037, Animal Science and Nutrition Building
> Guelph, Ontario, N1G 2W1
>
>
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p
> *Subject:* [cisco-voip] OT: WAN Technology
>
>
>
> Kind of off topic but I've been wondering...
>
>
>
> Does anyone still use frame relay or has MPLS type of connectivity pretty
> much replaced it?
>
>
u 1.1)
> - how many users presses 1 and next 2 (menu 1.2)
> - how many users presses 2 and next 2 (menu 2.2)
> etc.
>
> Is above reports are supported on CCX ?
>
> Rob
>
>
>
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rried over, or if I'm barking up the wrong tree?
I had read that there was some issues importing PINs into version 7.0, but
exporting from 7.0 TO a version 7.1.3+ was not cited as a problem.
Thanks!
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an older version of connection that used a different
>> encryption type, that is not supported by v9.
>> On Mar 3, 2014 8:57 PM, "Bill Talley" wrote:
>>
>>> I interpreted the issue as being from or to version 7.0 as it also
>>> suggests upgrading to 7
is is the hash issue then where the PIN was actually restored
but versions use a different hash function? OR the users that reported the
issues were doing something wrong?
On Tue, Mar 4, 2014 at 8:09 AM, Ed Leatherman wrote:
> Justin,
>
> Thanks for the insight. Was this a straight-up
not supported by v9.
>
> On Mar 3, 2014 8:57 PM, "Bill Talley" wrote:
>
> I interpreted the issue as being from or to version 7.0 as it also
> suggests upgrading to 7.1.3 prior (IIRC) to using COBRAS.
>
> Sent from an Apple iOS device with very tiny touchscreen input
or is that the users in question last
> changed their pin on an older version of connection that used a different
> encryption type, that is not supported by v9.
>
> On Mar 3, 2014 8:57 PM, "Bill Talley" wrote:
>
> I interpreted the issue as being from or to version 7.0 as
sort out
all its tcp sessions cleanly if I just swap network connectivity?
I've got matching firmware on both clusters and the database appears to be
in good working order on the new cluster, just want to minimize any
surprises.
Thanks!
Ed
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; fax. 410.252.9284
>
> Twitter <http://twitter.com/heliontech> |
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> | Website <http://www.heliontechnologies.com/> | Email
> Support
>
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>
>
>
>
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> *F
nologies.com/> | Email
> Support
>
> Support Phone. 410.252.8830
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Ed Leatherman
> *Sent:* Thursday, March 06, 2014 3:48 PM
> *To:* Cisco VOIP
> *Subject:* [cisco-voip]
tion and
explain why the configs are all different :) I think a lot of the initial
configuration was done via GUI so I suspect something in there got a little
flaky.
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cial-cert-guide-9781587143854
>
>
> Is this the best to get to study/prepare for the test? Any suggestions
> are appreciated.
>
>
> Thank you
>
>
>
>
>
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Solutions.
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t; Timeout Design Estimation".
>
> If that formula gives you more than 300 seconds, increase the repltimeout.
>
> Brian
>
>
> On Tue, Apr 22, 2014 at 11:26 AM, Ed Leatherman wrote:
>
>> Reviewing some docs for upgrading my primary production cluster from 8 to
>
stage ;)
Appreciate any tips!
Thanks,
Ed
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>
>
> Support Phone. 410.252.8830
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...
Hello!
I'm trying to help chase down a intermittent issue where Unity needs to
transfer a caller off-site to an answering service, and sometimes the
transfer doesn't complete and the caller gets left on-hold. I was hoping
someone could explain a message i'm seeing in the traces during a failure.
ke an inbound call to the CUCM from PSTN, and direct this call to an
> sccp phone (this is replacing the Unity), and then do a blind, consult
> transfer (2 different tests) to the answering service and see if you can
> reproduce the problem.
>
> Thanks
> Sreekanth
>
>
> On 1 Se
onds and find the ISDN Call
> Proceeding message that triggers CUCM’s attempt at connecting media between
> the two call-legs.
>
>
>
> - Dan
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Sreekanth Narayanan
&
to replicate this with an SCCP phone is marginal value - SCCP
> phones implement the softkey set received from UCM. Thus the phone won’t
> allow you to send transfer when UCM has not activated that soft key.
>
> Why is the egress call stopped in the progress state?
>
> -W
00 response – all immediately after ISDN Call
>>> Proceeding comes in. If you notice one of these missing then it’s likely an
>>> MX timeout issue. I’ve recently seen an issue where StationD doesn’t ACK an
>>> OpenReceiveChannel signal, resulting in a MX timeout. Doubt it’s
eeding comes in. If you notice one of these missing then it’s likely an
> MX timeout issue. I’ve recently seen an issue where StationD doesn’t ACK an
> OpenReceiveChannel signal, resulting in a MX timeout. Doubt it’s related to
> this problem though… my issue was related to CTI ports.
>
TAC engineer with the SR also concluded it was CSCun15967 and pointed
us to 9.1.2.12900.002
or 9.1.2.12035.001
Workaround with Unity Connection using supervised transfer seems to work
fine in the meantime.
Thanks again!
Ed
On Tue, Sep 2, 2014 at 2:06 PM, Ed Leatherman
wrote:
> We're
https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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s on the subject, thanks!
Ed
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lay_service.php
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Ed Leatherman
> *Sent:* Tuesday, October 07, 2014 7:50 AM
> *To:* Cisco VOIP
> *Subject:* [cisco-voip] Endpoint accessibility options for deaf
>
>
>
>
>
> G
ords - at some point
we will probably have this as part of our real billing software though.
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partition. Wish there was a cleaner way to report on these
>> that included the remote destination number.
>>
>> Does anyone know of a better way to do this? CUCM 9.1.
>>
>> BTW I am using splunk right now to sift through the records - at some
>> po
ledPartyNumber) & finalCalledPartyNumber =
> PSTN number would work. If you can combine an AXL query to do a lookup of
> a user's remote destinations then that would make it even easier.
>
> -Ryan
>
> On Oct 28, 2014, at 2:19 PM, Ed Leatherman
> wrote:
>
> Thanks for the sugges
UCM_BK_C268EF1B_00_cucm-cdr-admin-guide-91_chapter_011.html#CUCM_RF_ME367C12_00
>
> -Ryan
>
> On Oct 23, 2014, at 11:49 AM, Wes Sisk (wsisk) wrote:
>
> I see suggestions that the last redirecting number is the desk number.
>
> -Wes
>
>
>
> On Oct 23, 2014, at 10:04
I agree w Lelio
—
Sent from Mailbox
On Thu, Oct 30, 2014 at 1:46 PM, Lelio Fulgenzi wrote:
> I think shared lines are gonna cause you more grief. Especially if they're
> centrally located, in quadrants so to speak. Guessing out loud, an emergency
> is probably going to affect multiple peopl
d
> make it easier. Maybe create a special CSS for this one function?
>
> On Thu, Oct 23, 2014 at 5:04 PM, Ed Leatherman
> wrote:
>
>> Hello!
>>
>> I'm trying to figure out how we can identify call legs from Mobile
>> Connect that are going out to long
.
>
>
>
> [image: LNsignatureFile]
>
>
>
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ent of this information is
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> us immediately by e-mail and delete the original message and any attachment
> without reading or saving in any manner.
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://en.wikipedia.org/wiki/Magic_8-Ball#Possible_answers>," etc.), why
> not go with something original and creative and defend your answer with a
> well thought out explanation? Who knows, you might provoke the next big
> revolution in QoS. Oh wait, that's MediaNet, isn't it?
>
> So, now that you have my answer, I'd be really interested in reading
> your's.
>
> Have a great Saturday everyone.
>
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UCM/IM&P through expressway is ok
>>>
>>>
>>>
>>> when i type the correct password, i receive the following error: "Cannot
>>> communicate with the server".
>>>
>>>
>>>
>>> I downloaded the debug "Jabber for iphone" generate but couldn't find
>>> any clue about what is happening.
>>>
>>>
>>>
>>> Diagnose logging at Expressway-C neither.
>>>
>>>
>>>
>>> What should i look first?
>>>
>>>
>>>
>>> Does my "Jabber" devices at CUCM needs to use a secure phone profile or
>>> i can use a non-secure phone profile (default)?
>>>
>>>
>>>
>>> Thanks in advance,
>>>
>>>
>>> José Paulo de Oliveira Petry
>>> petr...@gmail.com
>>>
>>
>>
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t;
>> Adam Pawlowski
>> SUNYAB
>>
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configuring the client - I'm just
looking for ease-of-use here for the students and staff that will be using
it.
Thanks for any clarification, most of the documents are centered around
IM&P which isn't what i'm after right now.
Happy New Year!
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