LMS 4.1 with CUCM 8.0.2 Phones only detected as End Hosts
Hi,
looks like I still have an issue with LMS to recognize the IP Phones in UT as IP Phones.
SNMP RO on Call Manager is enabled and is green in CM (e.g. topology) - so SNMP get is basically fine.
The Phones are recognised as End Devices in UT.
As far as I understand, now if I start a Phone Aquisition, the CUCM is polled by LMS to gather additional information about the phones.
So it seems there is a problem with the SNMP polling of the Callmanager?
thanks for any help!
br.herwig
Yes, its in DCR
below the device status of the Callmanager:
20.
192.168.2.10
Callmanager
192.168.2.10
1.3.6.1.4.1.9.1.583
Cisco 7825H Media Convergence Server
The phones are collected by UT as end hosts (it is seen through the voice vlan called "UC"):
But this phones are not recognized as phones in UT:
thanks,
br.herwig
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I have some problems but not with all the users, is only working with 3 users and the rest of them don't work, I attached the errors that I get when I try to log in Telephony Accounts in the Jabber, I used the Jabber of one User, and from here I introduce my credentials (user and password) and I can get the voice mails (only from de 3 users that is woriking fine), but when I introduce the credentials of the other user I get the error and Can't get the voicemails.
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Hi,
we have this problem from many days...
we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
our setup is like below
HO and BR2 having SIP lines and BR1 has PSTN Lines.
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when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
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Thankyou for the Response...
here is the Runn config for BR1 Connected to PSTN lines....
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice class h323 1
h225 timeout tcp establish 3
interface Tunnel100
description " Tunnel JED-RYD "
bandwidth 2048
ip address 10.10.0.1 255.255.255.252
tunnel source 172.31.217.202
tunnel destination 172.31.3.18
interface FastEthernet0/0
description DAMMAM Local LAN
no ip address
duplex auto
speed auto
interface FastEthernet0/0.20
description JEDDAH Local LAN
encapsulation dot1Q 20
ip address 192.168.20.5 255.255.255.0
interface FastEthernet0/0.21
description JEDDAH VOICE VLAN
encapsulation dot1Q 21
ip address 192.168.21.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.21.5
interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
router eigrp 200
network 10.10.0.0 0.0.0.3
network 192.168.20.0
network 192.168.21.0
no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
neighbor 172.31.217.201 remote-as 65000
no auto-summary
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.20.1
ip route 192.168.20.50 255.255.255.255 192.168.20.1
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
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timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
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impedance complex2
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caller-id alerting dsp-pre-allocate
voice-port 0/2/0
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
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shutdown
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voice-port 0/2/1
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
shutdown
impedance complex2
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input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
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timing hookflash-out 500
timing guard-out 300
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timeouts interdigit 3
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timeouts ringing 5
timeouts wait-release 1
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timing guard-out 300
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caller-id alerting dsp-pre-allocate
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no battery-reversal
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output attenuation -3
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timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
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supervisory disconnect dualtone mid-call
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output attenuation -3
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timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
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sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
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associate ccm 2 priority 2
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associate profile 2 register TRNJED
associate profile 3 register MTPJED
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switchback method immediate
switchback interval 15
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
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codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
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description To CallManager - SBWPMPUB
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voice-class h323 1
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destination-pattern [^2].T
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dial-peer voice 9002 pots
description ** 02-6140295(outgoing) **
destination-pattern [^2].T
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dial-peer voice 9003 pots
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destination-pattern [^2].T
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dial-peer voice 9004 pots
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destination-pattern [^2].T
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destination-pattern [^0].T
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dial-peer voice 9010 pots
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Please rate helpful posts! -
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Hi
I have CUCM and CUC 8.6.2 running and MOC with CUCI-Lync 8.5 (with visual Voicemail) running OK with full registry configuration (see below). We are moving to Lync 2013 and want to use CUCI-Lync 9.2.
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OLD REG Configuration:
Windows Registry Editor Version 5.00
[HKEY_CURRENT_USER\Software\Cisco Systems, Inc.\Unified Communications\CUCIMOC]
"RememberMe"=dword:00000001
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[HKEY_CURRENT_USER\Software\Cisco Systems, Inc.\Client Services Framework\AdminData]
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http://www.cisco.com/en/US/docs/net_mgmt/ciscoworks_health_and_utilization_monitor/1.2/user/guide_12/HUMrprts.html#wp1134146
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The Compatibility Tool does show support for MPE 2.1.1 with CUCM 7.1 (x)
Cisco Unified Communications Compatibility Tool
http://tools.cisco.com/ITDIT/vtgsca/VTGServlet
Here are 3 groups running this similar combo;
https://supportforums.cisco.com/message/3145002#3145002
https://supportforums.cisco.com/message/1330041#1330041
https://supportforums.cisco.com/message/3150066#3150066
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Hello,
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Thank youhttp://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/directry.html#pgfId-1133454
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Is unity 7 with domino supported with cucm 8 ?
I checked the following link and it seemed confusing
http://tools.cisco.com/ITDIT/vtgsca/VTGServlet
Kindly confirmHello -
I think this is a more clear document - http://www.cisco.com/en/US/docs/voice_ip_comm/unity/compatibility/matrix/cutspmtx.html
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Ginger -
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ABAB,
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I cannot really help you with specifc requirements as it largely depends on exactly how you intend to deploy it and what features you intend to enable. Clustering, Digital Networking, Unified Inbox, etc.. all have their own additional requirements the whole of which would not fit into a message board post.
However, specific bandwidth and latency requirements are listed in the "System Requirements for Cisco Unity Connection Release 8.X. http://www.cisco.com/en/US/partner/docs/voice_ip_comm/connection/8x/requirements/8xcucsysreqs.html
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The system is Cisco UCCE 9.0.
Thanks in advance.But I am actually running several more PGs on the same system but they all VRU PGs.
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