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Today's Topics:
1. Re: Is CUPS required to get phone status / user status for
Attendant Console 9.1?
(Jamie Gale -X (jamgale - Arc Solutions at Cisco))
2. CME and abbreviated dialling issue (Andy)
3. Cisco ELM license clarification (Mauro Celli)
4. Re: H.323 DTMF to SIP 2833 (Eric Pedersen)
----------------------------------------------------------------------
Message: 1
Date: Thu, 18 Jul 2013 17:09:30 +0000
From: "Jamie Gale -X (jamgale - Arc Solutions at Cisco)"
<[email protected]>
To: Robert Kulagowski <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] Is CUPS required to get phone status / user
status for Attendant Console 9.1?
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Robert,
In order to get simple phone status (on hook/off hook/DND/CFW) you do not need
to have CUPS installed and running.
If you want to get presence information for Cisco Jabber then you need the CUP
component, you can also get presence information for Microsoft OCS/Lync and
this uses the local client on the operator's PC to obtain the information.
Is CUPS required for CUEAC to know whether I'm at my PC, if we're in a
Microsoft environment and we're running MOC / Lync?
Yes is you need the information from Cisco Jabber, no if you want it from
MOC/Lync
Will CUEAC be able to tell the operator that I've set my calendar in
Outlook to "Out of Office"?
Yes if CUPS/MOC/Lync is integrated with Microsoft Exchange
Will CUEAC be able to tell the operator that I'm on the phone, or that
I've set it to "Call Forward All"?
Yes, without having CUPS or MOC/Lync
Also, feel free to watch our other videos at the following link, these are more
up to date than the ones you found:
https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs?
Kind Regards
Jamie Gale
Technical Marketing Engineer, Cisco Unified Attendant Consoles
Arc Solutions, onsite at Cisco
[email protected]
D +1 919 392 4671
M +1 919 699 4910
Find our new Cisco Unified Attendant Console End User training videos at
https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs%3D
Join the Cisco Unified Attendant Console Forum at Arc Solutions!
http://forum.arcsolutions.com/forumdisplay.php?f=4
On Jul 18, 2013, at 11:51 AM, Robert Kulagowski <[email protected]>
wrote:
I watched the video over at the Arcsoft website
(http://www.arcsolutions.com/North_America/Solutions/Products/video/CUEAC_Overview.aspx)
and must have been too distracted by the presenter and may have missed
it.
Hint: do not do a presentation / recording when you have a cold.
Sniffling into the microphone every 5 seconds is pretty annoying.
If I deploy CUEAC 9.1 Enterprise (to replace our old attendant console
software which we've dragged forward all the way from 4.1) what do I
need to give me:
1- phone presence.
2- user status.
Is CUPS required for CUEAC to know whether I'm at my PC, if we're in a
Microsoft environment and we're running MOC / Lync?
Will CUEAC be able to tell the operator that I've set my calendar in
Outlook to "Out of Office"?
Will CUEAC be able to tell the operator that I'm on the phone, or that
I've set it to "Call Forward All"?
Any other products that people like other than CUEAC?
Thanks.
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Message: 2
Date: Thu, 18 Jul 2013 20:06:44 +0100
From: Andy <[email protected]>
To: Cisco VoIP List <[email protected]>
Subject: [cisco-voip] CME and abbreviated dialling issue
Message-ID: <[email protected]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Ok, I have an issue thats driving me mad.
I have a CME its v8.6 and there seems to be an issue with abbreviated
dialling using the Bulk-Speed-dial
here is the cme config
telephony-service
no auto-reg-ephone
authentication credential admin cisco
max-ephones 20
max-dn 60
ip source-address 172.31.102.3 port 2000
service phone webaccess 0
url services http://172.31.102.4/voiceview/common/login.do
url authentication http://172.31.102.3/CCMCIP/authenticate.asp
load 7970 SCCP70.9-1-1SR1S.loads
time-zone 22
time-format 24
date-format dd-mm-yy
bulk-speed-dial list 0 flash:List0.txt
bulk-speed-dial list 1 flash:List1.txt
bulk-speed-dial list 2 flash:List2.txt
bulk-speed-dial list 3 flash:List3.txt
bulk-speed-dial list 4 flash:List4.txt
bulk-speed-dial list 5 flash:List5.txt
bulk-speed-dial list 6 flash:List6.txt
bulk-speed-dial list 7 flash:List7.txt
bulk-speed-dial list 8 flash:List8.txt
bulk-speed-dial list 9 flash:List9.txt
bulk-speed-dial prefix *
voicemail 5600
max-conferences 4 gain -6
moh music-on-hold.au
multicast moh 239.1.1.10 port 16384
web admin system name admin password cisco
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern ....
transfer-pattern 9T
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone 2
mac-address 0019.AA96.D6C1
ephone-template 1
type 7970
keep-conference
button 1:5
!
!
!
ephone 3
mac-address 0019.AA96.D6BF
ephone-template 1
type 7970
keep-conference
button 1:6
ephone-dn 5 dual-line
number 7811
pickup-group 1
description 7811
name D Cosham
!
!
ephone-dn 6 dual-line
number 7812
pickup-group 2
description 7812
name S Christmas
Only some of the lists work and I can't for the life of me work out why,
the lists have exactly the same numbers in them.
so *202, *502, *702, *802, *902 work the others dont.
this is the contents of the ListX.txt file
02,7811,some data,,append
03,7811
04,7811,,,,
any pointers would be gratefully received.
--
Regards
Andy
------------------------------
Message: 3
Date: Fri, 19 Jul 2013 11:09:35 +0000
From: Mauro Celli <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Cisco ELM license clarification
Message-ID:
<7e5f77f37136a9448d89426ccacb3f7605a3e...@hme-mbx01.hme.2000net.it>
Content-Type: text/plain; charset="us-ascii"
Hi,
if I build cucm in my vmware lab, install license in ELM and after I move all
to my customer vcenter (I maintain ip address hostname and mac address),
ELM goes in demo mode and I need to rehome my license.
Why? How it calculate uuid?
Thanks
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Message: 4
Date: Fri, 19 Jul 2013 16:28:25 +0000
From: Eric Pedersen <[email protected]>
To: "Divin John (dijohn)" <[email protected]>, "Chris Ward (chrward)"
<[email protected]>, Mark Holloway <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Does the other end of the SIP trunk support KPML? I'm pretty sure that if you
use alphanumeric on the h323 trunk and kpml on the SIP trunk you won't need an
MTP.
From: cisco-voip [mailto:[email protected]] On Behalf Of Divin
John (dijohn)
Sent: 18 July 2013 2:44 AM
To: Chris Ward (chrward); Mark Holloway
Cc: [email protected]
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
SCCP and SIP Phones support OOB and in band.
CUCM does rtp-nte only out a SIP trunk.
H323 Trunks don't support RFc2833 (inband)
H323 to SIP would need a MTP if you need to convert OOB (h323 side) to
rtp-nte/inband (SIP side). CUCM would automatically pick up a MTP for DTMF
inter-networking.
Regards,
Divin
From: "Chris Ward (chrward)" <[email protected]<mailto:[email protected]>>
Date: Wednesday, 17 July 2013 8:48 PM
To: Mark Holloway <[email protected]<mailto:[email protected]>>
Cc: Divin John <[email protected]<mailto:[email protected]>>,
"[email protected]<mailto:[email protected]>"
<[email protected]<mailto:[email protected]>>
Subject: RE: [cisco-voip] H.323 DTMF to SIP 2833
Since its in-band, it wouldn't require an MTP... The 2833 would flow as part of
the RTP stream.
Adding rtp-nte to the dial peer should remove the need for an MTP if the other
leg (from CUCM's perspective at least) is SIP.
+Chris
Unity Connection TME
From: Mark Holloway [mailto:[email protected]]
Sent: Wednesday, July 17, 2013 11:08 AM
To: Chris Ward (chrward)
Cc: Divin John (dijohn);
[email protected]<mailto:[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Wouldn't this still require MTP? It looks like he is trying to avoid using MTP
for the conversion.
On Jul 17, 2013, at 9:17 AM, Chris Ward (chrward)
<[email protected]<mailto:[email protected]>> wrote:
CUCM support RFC2833 so long as one of the legs is SIP. An H323 to SIP call
through CUCM should allow RFC2833.
+Chris
Unity Connection TME
From: cisco-voip
[mailto:[email protected]<mailto:[email protected]>]
On Behalf Of Mark Holloway
Sent: Wednesday, July 17, 2013 8:13 AM
To: Divin John (dijohn)
Cc: [email protected]<mailto:[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
John is correct, if you're using Alphanumeric on H323 you should be able to use
SIP INFO without the need of an MTP. The question is what's on the other end of
the SIP Trunk, and will they support SIP INFO?
DTMF interworking between SIP and H.323 in the signaling plane, using the
Alphanumeric method
of UserInputIndication
On Jul 16, 2013, at 1:29 PM, Divin John (dijohn)
<[email protected]<mailto:[email protected]>> wrote:
CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
So, to stop CUCM from invoking an MTP, the SIP side should support Out of band
(notify maybe)
Regards,
Divin
From: Nate VanMaren <[email protected]<mailto:[email protected]>>
Date: Tuesday, 11 June 2013 1:26 AM
To: "[email protected]<mailto:[email protected]>"
<[email protected]<mailto:[email protected]>>
Subject: [cisco-voip] H.323 DTMF to SIP 2833
What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
invoke an MTP when the outbound side of CUCM is a SIP Trunk that is configured
to use 2833.
I am currently using h245-alphanumeric, which seems to be invoking an MTP.
Thanks,
-Nate
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