CUCM Timezone
hello dear,
why do we have two parameters for the timezone on CUCM?
you can set timezone through CLI command line, and you can have your time zone on GUI interface, system / date and time where you use it in device pool.
and for the note they can be different values.
so what is the difference between the two?
as long as i use NTP Server to synchronize with, we cannot check time zone through GUI interface os administration.
what it looks for me is that the CLI parameter is the one that take effect, while the GUI parameter affects nothing and is useless.
please can you confirm?
one other issue is that how may i configure Daylight time saving through CLI Command line? and the offset
regards,
Hi
You can set timezones no the date/time groups, and you can also set NTP servers. Some phones will use the NTP servers you specify, and others will use the time of the CallManager server.
If your CCM has the wrong time zone (or has the wrong UTC, and a timezone that is set to make it appear right) then phones will have incorrect time if they offset from the CCM UTC rather than using NTP.
For the server:
1) First do a 'show timezone list', find your timezone, and note the number
2) Then do a 'set timezone x' where x is your timezone number from step 1
On the date/time groups, I would set always set a correct timezone, and NTP servers.
Regards
Aaron
Please rate helpful posts..
Similar Messages
-
Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6
I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running 15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error. I think I’m close.. but who knows. Any assistance would be greatly appreciated
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
CUCM (single/pub)- 192.168.1.200
2811 acting as cube - 192.168.1.203
Calling Number - 18165297500
Called Number - 18452055544
vrtr1#show sip register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
17772253754 -1 20 yes
vrtr1#
The Call Setup Information is:
Call Control Block (CCB) : 0x49646C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 18165297500
Called Number : 17772253754 (my customer number not called number)
Source IP Address (Sig ): 192.168.1.203 (my 2811 router)
Destn SIP Req Addr:Port : 204.11.192.159:5080
Destn SIP Resp Addr:Port : 204.11.192.159:5080
Destination Name : 204.11.192.159
Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m: <sip:[email protected]:5080;transport=udp>
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
s=sip call
c=IN IP4 204.11.192.159
t=0 0
m=audio 61094 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>;tag=35399D8-63
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0
Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
u all
Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0
************************** Running Config **************************
sh run
vrtr1#sh running-config
Building configuration...
Current configuration : 4189 bytes
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
hostname vrtr1
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
dot11 syslog
ip source-route
ip cef
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
voice service voip
ip address trusted list
ipv4 192.168.1.0 255.255.255.0
ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 1800 min 1800
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FTX1133A4QR
controller T1 0/0/0
cablelength long 0db
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.1.1
snmp mib persist circuit
control-plane
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200
ccm-manager config
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end
exitThank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
dial-peer voice 6 voip
description ## INBOUND CALL from ITSP ##
session protocol sipv2
session target sip-server
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description ## INBOUND DID to CUCM ##
destination-pattern 17772253754
session protocol sipv2
session target ipv4:192.168.1.200
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m:
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 61782 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
l: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 102 INVITE
l: 0
Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>;tag=8408714-B60
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
vrtr1#u al
Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0 -
When first login in via the web page. When going to Configure menu and choosing CUCME to enter it manually, I get:
Error: Login to CUCME failed with the new values. Check the new CUCME configuration and enter the correct values.
hostname: 172.23.0.1
web user name: admin
web password: cisco
Sip gateway hostname: 172.23.0.1
ccn reporting historical
database local
description "se-172-23-0-2"
end reporting
ccn subsystem sip
gateway address "172.23.0.1"
mwi sip unsolicited
end subsystem
BR2-ROUTER#sh run
Building configuration...
Current configuration : 5264 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname BR2-ROUTER
boot-start-marker
boot-end-marker
card type t1 0 3
logging message-counter syslog
logging buffered 51200 warnings
no aaa new-model
clock timezone MST -7
clock summer-time MDT recurring
network-clock-participate wic 3
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.21.0.1 172.21.0.49
ip dhcp excluded-address 172.21.0.59 172.21.0.254
ip dhcp excluded-address 172.20.0.1 172.20.0.10
ip dhcp pool CME
network 172.21.0.0 255.255.255.0
option 150 ip 172.21.0.1
default-router 172.21.0.1
ip dhcp pool LAPTOPS
network 172.20.0.0 255.255.255.0
default-router 172.20.0.2
dns-server 10.10.10.1
no ip domain lookup
ip domain name wilson.com
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h225-notify cid-update
sip
bind control source-interface GigabitEthernet0/0.20
bind media source-interface GigabitEthernet0/0.20
registrar server expires max 600 min 60
voice register global
mode cme
source-address 172.21.0.1 port 5060
max-dn 4
max-pool 4
authenticate register
timezone 12
time-format 24
date-format YY-M-D
voicemail 3600
tftp-path flash:
create profile sync 0021447056000116
ntp-server 174.137.67.50 mode directedbroadcast
voice register dn 1
number 3006
call-forward b2bua busy 3600
call-forward b2bua mailbox 3006
call-forward b2bua noan 3600 timeout 12
name rp-sip-1-16
label SIP 511-5016
mwi
voice register pool 1
id mac FCFB.FBCA.30CE
type 7965
number 1 dn 1
dtmf-relay rtp-nte
username 3006 password cisco
description 687-3006
codec g711ulaw
voice-card 0
username admin privilege 15 secret 5 $1$..D.$orbTsqgPSvNkMpfjjkg5q.
archive
log config
hidekeys
controller T1 0/3/0
cablelength long 0db
controller T1 0/3/1
cablelength long 0db
interface Loopback0
ip address 172.23.0.1 255.255.255.252
ip ospf network point-to-point
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10 native
ip address 172.20.0.2 255.255.255.0
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 172.21.0.1 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 172.22.0.1 255.255.255.0
interface GigabitEthernet0/1
ip address 192.168.1.138 255.255.252.0
duplex auto
speed auto
interface Integrated-Service-Engine1/0
ip unnumbered Loopback0
service-module ip address 172.23.0.2 255.255.255.252
service-module ip default-gateway 172.23.0.1
no keepalive
ip forward-protocol nd
ip route 172.23.0.2 255.255.255.255 Integrated-Service-Engine1/0
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:/gui
access-list 23 permit 10.10.10.0 0.0.0.7
nls resp-timeout 1
cpd cr-id 1
control-plane
ccm-manager fax protocol cisco
mgcp fax t38 ecm
dial-peer voice 3600 voip
destination-pattern 36..
session protocol sipv2
session target ipv4:192.168.1.144
dtmf-relay sip-notify
codec g711ulaw
no vad
sip-ua
retry invite 3
timers trying 400
mwi-server ipv4:192.168.1.144 expires 3600 port 5060 transport udp
gatekeeper
shutdown
telephony-service
no auto-reg-ephone
em logout 0:0 0:0 0:0
max-ephones 10
max-dn 10 no-reg both
ip source-address 172.23.0.1 port 2000
voicemail 3600
max-conferences 8 gain -6
call-forward pattern .T
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 3007
description 687-9898-3007
name Vatos locos
call-forward busy 3600
call-forward noan 3600 timeout 12
ephone-dn 2
number 3008
description 687-9898-3008
name Vatos locos2
call-forward busy 3600
call-forward noan 3600 timeout 12
ephone-dn 3 octo-line
number 3009
huntstop channel 6
ephone-dn 4
number 7999....
mwi on
ephone-dn 5
number 7998....
mwi off
ephone 1
device-security-mode none
description TESTTTTT
mac-address FCFB.FBCA.3406
max-calls-per-button 5
busy-trigger-per-button 4
type 7965
button 1:1 2:3
ephone 2
device-security-mode none
description TESTTTTT
mac-address FCFB.FBCA.3030
max-calls-per-button 4
busy-trigger-per-button 3
type 7965
button 1:2 2:3
line con 0
exec-timeout 0 0
logging synchronous
login local
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
access-class 23 in
privilege level 15
login local
transport input telnet
line vty 5 15
access-class 23 in
privilege level 15
login local
transport input telnet
scheduler allocate 20000 1000
ntp server 174.137.67.50
end
BR2-ROUTER#
Apr 12 2011 16:23:12 gui/admin_user.js
122 585532 Mar 30 2011 05:48:46 phone/7975/cnu75.8-3-2-27.sbn
123 2453636 Mar 30 2011 05:48:56 phone/7975/cvm75sccp.8-3-2-27.sbn
124 326315 Mar 30 2011 05:48:58 phone/7975/dsp75.8-3-2-27.sbn
125 557786 Mar 30 2011 05:49:00 phone/7975/jar75sccp.8-3-2-27.sbn
126 638 Mar 30 2011 05:49:02 phone/7975/SCCP75.8-3-3S.loads
127 642 Mar 30 2011 05:49:02 phone/7975/term75.default.loads
128 0 Mar 30 2011 05:49:02 phone/7941-7961
129 2494499 Mar 30 2011 05:49:12 phone/7941-7961/apps41.8-3-2-27.sbn
130 547146 Mar 30 2011 05:49:16 phone/7941-7961/cnu41.8-3-2-27.sbn
131 2340 Apr 02 2011 03:55:02 April012011.txt
132 3579 Apr 12 2011 03:52:42 softkeyDefault_kpml.xml
133 69 Apr 12 2011 03:52:40 syncinfo.xml
134 2682 Apr 12 2011 03:52:42 SEPFCFBFBCA30CE.cnf.xml
135 1882 Apr 12 2011 03:52:42 SIPDefault.cnf
136 3613 Apr 12 2011 03:52:42 softkeyDefault.xml
137 3987 Apr 12 2011 16:23:10 gui/admin_user.html
138 1029 Apr 12 2011 16:23:14 gui/CiscoLogo.gif
139 617 Apr 12 2011 16:23:14 gui/CME_GUI_README.TXT
140 953 Apr 12 2011 16:23:14 gui/Delete.gif
141 16344 Apr 12 2011 16:23:14 gui/dom.js
142 864 Apr 12 2011 16:23:16 gui/downarrow.gif
143 6146 Apr 12 2011 16:23:16 gui/ephone_admin.html
144 4558 Apr 12 2011 16:23:16 gui/logohome.gif
145 3866 Apr 12 2011 16:23:16 gui/normal_user.html
146 78428 Apr 12 2011 16:23:18 gui/normal_user.js
147 1347 Apr 12 2011 16:23:18 gui/Plus.gif
148 843 Apr 12 2011 16:23:18 gui/sxiconad.gif
149 174 Apr 12 2011 16:23:18 gui/Tab.gif
150 2431 Apr 12 2011 16:23:20 gui/telephony_service.html
151 870 Apr 12 2011 16:23:20 gui/uparrow.gif
152 9968 Apr 12 2011 16:23:20 gui/xml-test.html
153 3412 Apr 12 2011 16:23:20 gui/xml.templateFixed. Routing issue:
Routing issue:
ip http access-class 23 !!!!!! Preconfigured from Factory
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:/gui
access-list 23 permit 10.10.10.0 0.0.0.7 !!!!!! Preconfigured from Factory
To fix
No ip http access-class 23 -
DST issue with 7940/7960 on CUCM 8.6.2
Hi Guys,
In the last 19 October, in any city of Brazil started of date summer time (DST), and the IP Phone models 7940/7960 not changed the time automatically. After restart physical or logical the IP Phone stay the correct time.
7940
Firmware de aplicação P00308010200
Firmware de inicialização PC0303010200
Versão 8.1(2.0)
I read any posts in this forum, but, not is clear for me if this problem is related any bug or no. Any have idea about this?
Thank You,
WilsonHi Wilson,
There are a number of DST bugs that affect CUCM 8.6. I would have a look
at the attached bugs . Did you install any of the DST/TZ Timezone updates for CUCM?
Cisco Unified Communications Manager / Cisco Unity Connection
Timezone Update 2013d
Release Notes Version 1
September 19, 2013
Introduction:
These release notes contain important information about installation procedures for the 8.6(2x) Timezone
Update for Cisco Unified Communications Manager or Cisco Unity Connection.
Note
Before you install this Timezone Update, Cisco recommends that you review the
Important Notes section for information about issues that may affect your system.
Updates in This Release
DST updates are cumulative, so installing this patch will provide all of the fixes in the New Updates section plus all of the
fixes in the Previous Updates section
New Updates
CSCui46884
DST/TZ data update with 2013d
Previous Updates
CSCue96910
DST/TZ data update with 2013b
CSCue06553
DST/TZ data update with 2012j
CSCtz20987
DST: Update CM to Olson TZ version 2012c
CSCtn29406
DST: Update CM to Olson TZ version 2011h
CSCtk55066
DST: Update CM for Olson TZ ver 2010o
CSCtj75860
Inconsistent changeover
for DST spring and fall in US, Canada
CSCtg50448
DST: Update CM for Olson TZ ver 2010i
Known Caveats
CSCuj30440
Renaming DST
cop file in logs
CSCuj26553
69xx phones will not display correct time for Chile/Santiago Timezone
CSCue81856
dst
updater.2012j DST cop file does not match Gaza in timeanddate.com
From
http://www.cisco.com/web/software/282074298/106084/dst-862-2013d-cop-readme.pdf
Cheers!
Rob
"Why do the best things always disappear "
- The Band -
Cisco 9951 and cucme registration problem
Hi Everybody
I am trying to register a cisco 9951 phone to a cucme. The phone was previosly connected and registered to a cucm 9.1, now it must be registered in the cucme. The problem is that the status messages of the phone show that the phone has 9.3(2) firmware version and the cucme has 9.2(1) sip9951.9-2-2SR1-9.loads. It means that the phone has a higher version than the version in the router flash (cucme).
I tryed tod download a newer cucme file-set but the last version available in cisco.com is the 10.0 and it has the sip9951.9-2-2SR1-9.loads file which is still older than the version in the phone. So the phone doesnt register.
I follow step by step the instructions to configure and register ip sip phones to cucme. but I am stock.
here is the configuration:voice register global
mode cme
source-address X.X.6.54 port 5060
max-dn 10
max-pool 5
load 9951 sip9951.9-2-2SR1-9
timezone 42
tftp-path flash:
create profile sync 0001184550824421
camera
video
voice register dn 1
number 4002
name SIPPhone 2
voice register pool 1
id mac 501C.BFFC.DD85
type 9951
number 1 dn 1
username nico password 12345
description +571344002
camera
video
tftp-server flash:Phones/9951/dkern9951.100609R2-9-2-2SR1-9.sebn alias dkern9951.100609R2-9-2-2SR1-9.sebn
tftp-server flash:Phones/9951/kern9951.9-2-2SR1-9.sebn alias kern9951.9-2-2SR1-9.sebn
tftp-server flash:Phones/9951/rootfs9951.9-2-2SR1-9.sebn alias rootfs9951.9-2-2SR1-9.sebn
tftp-server flash:Phones/9951/sboot9951.031610R1-9-2-2SR1-9.sebn alias sboot9951.031610R1-9-2-2SR1-9.sebn
tftp-server flash:Phones/9951/sip9951.9-2-2SR1-9.loads alias sip9951.9-2-2SR1-9.loads
tftp-server flash:Phones/9951/skern9951.022809R2-9-2-2SR1-9.sebn alias skern9951.022809R2-9-2-2SR1-9.sebn
the status mesages on the phone say:
Upgrade rejected: HW compat failure. Must use 9.3(2) or later release on this phone
Error updating locale
It still shows:
File not found United_States/g4tones.xml
File not found SIP_English_United_States/ik-sip.jar
Those files tha the phone doesnt find are in the cucm 9.1. I tryed to put them in the flash and configured the tftp-server command and I also downloaded the 9.3(2) from the cucm and copied to the router flash, but it still has registrations problems.
I also tryed to reset the phone to the factory defaults but this sequence with # key and 123456789*0# doesnt aply to sip phones and the procedure of the following link still doesnt work:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/9971_9951_8961/10_0/english/adminguide/P567_BK_A27DADFD_00_adminguide-8961-9951-9971-10_0/P567_BK_A27DADFD_00_adminguide-8961-9951-9971-10_0_chapter_010100.html
Please help me with this issue.
Best regardsHi Sayeed,
Can you please send accross SIP messages when phone is trying to register with CME ?
Thanks
Manish -
RTMT Timezone data version mismatch
When opening the RTMT tool I get an error "There is a mismatch between timezone versions on this RTMT and the server you are trying to connect." Both my workstation and CUCM server are pointing to the same NTP so the time is the same. If I click Yes to update it errors and the tool closes. If I click NO to just let the mismatch be the tools opens up. Any suggestions on how to resolve this problem?
Thanks,
johnHey John,
We experienced the same issue after we upgraded the callmanager.
Our fix was to uninstall RTMT and download it again from CCM and reinstall it.
Hope this helps,
if so, please rate.
Thanks. -
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
registrar server expires max 3600 min 1800
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice register global
mode cme
source-address 172.22.72.1 port 5060
max-dn 30
max-pool 30
timezone 31
time-format 24
create profile sync 000242525906824A
dial-peer voice 1 voip
description VOICE CALLS TO STIMA PLAZA
destination-pattern 70....
session target ipv4:172.16.200.5
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 voip
description VOICE CALLS TO NAKURU E_HSE
destination-pattern 741..
session target ipv4:172.31.40.1
codec g711ulaw
no vad
dial-peer voice 3 voip
description VOICE CALLS TO STIMA PLAZA
destination-pattern 70....
session target ipv4:172.16.200.5
dtmf-relay h245-alphanumeric
codec g711ulaw
no vadwhen calling from CUCM i can hear to CME but when they call back i cannot hear.
-
VG-204 SIP - PSTN outbound not happening from CUCM
I have VG204 configured as SIP , SIP trunk is established to CUCM 7.x , Outbound to PSTN calls is not happening but Extension dialing within CUCM DB is happening. Inbound from PSTN works.
Call manager is not forwarding the calls initiated from vg204 extn to PSTN gateway .
Pls assist if i am missing something on vg204 or CUCM for outbound calls to happen .
, the error log on cucm is
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
version 15.1
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
no service dhcp
boot-start-marker
boot system flash:vg20x-advipservicesk9-mz.151-3.T3.bin
boot system flash:
boot-end-marker
aaa session-id common
clock timezone GMT 0 0
crypto pki token default removal timeout 0
no ip source-route
ip cef
no ip bootp server
no ip domain lookup
no ipv6 cef
voice call send-alert
voice rtp send-recv
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711alaw
modem passthrough nse codec g711alaw
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
redirect contact order best-match
voice class codec 2
codec preference 4 g711alaw
codec preference 5 g711ulaw
voice class sip-profiles 1000
request ANY sip-header User-Agent modify "User-Agent: Cisco-SIPGateway.*" "User-Agent: Cisco/v11 r1/alg01"
request ANY sip-header Server modify "Server: Cisco-SIPGateway.*" "Server: Cisco/v11 r1/alg01"
response ANY sip-header Server modify "Server: Cisco-SIPGateway.*" "Server: Cisco/v11 r1/h2p-vg-alg01"
response ANY sip-header User-Agent modify "User-Agent: Cisco-SIPGateway.*" "User-Agent: Cisco/v11 r1/alg01"
response ANY sip-header Remote-Party-ID remove
request ANY sdp-header Attribute modify "a=rtr" "a=X-nortr"
response ANY sdp-header Attribute modify "a=rtr" "a=X-nortr"
voice vad-time 1000
voice-card 0
application
package callfeature
param long-dur-disc-cause 21
param long-dur-duration 1440
param long-dur-action disconnect
param long-dur-call-mon enable
service dsapp
param callWaiting FALSE
param blind-xfer-wait-time 2
param callTransfer TRUE
global
service default dsapp
archive
log config
hidekeys
dial-control-mib retain-timer 1440
dial-control-mib max-size 500
ip tftp source-interface FastEthernet0/0
interface FastEthernet0/0
ip address 10.10.10.13 255.255.255.224
ip access-group al_trusted_SIP in
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
load-interval 60
speed 100
full-duplex
arp timeout 420
control-plane
voice-port 0/0
no snmp trap link-status
timeouts interdigit 4
description h2p/01
station-id name Analog phone
station-id number 26999
caller-id enable
voice-port 0/1
no snmp trap link-status
timeouts interdigit 4
description h2p/02
station-id name Analog phone
station-id number 26998
caller-id enable
voice-port 0/2
no snmp trap link-status
timeouts interdigit 4
description h2p/03
station-id name Analog phone
station-id number 26997
caller-id enable
voice-port 0/3
no snmp trap link-status
timeouts interdigit 4
description h2p/04
station-id name Analog phone
station-id number 26996
caller-id enable
mgcp profile default
dial-peer voice 1100 voip
permission orig
description Incoming from IP network
huntstop
session protocol sipv2
incoming called-number .
voice-class codec 2
voice-class sip profiles 1000
dtmf-relay rtp-nte
dial-peer voice 1210 voip
description SIP registrar server
huntstop
preference 4
destination-pattern .T
session protocol sipv2
session target ipv4:10.10.10.15
voice-class codec 2
voice-class sip profiles 1000
dtmf-relay sip-notify
dial-peer voice 99900 pots
description Analog phone
huntstop
preference 4
destination-pattern 26999
progress_ind alert strip
port 0/0
dial-peer voice 99901 pots
description Analog phone
huntstop
preference 4
destination-pattern 26998
progress_ind alert strip
port 0/1
dial-peer voice 99902 pots
description Analog phone
huntstop
preference 4
destination-pattern 26997
progress_ind alert strip
port 0/2
dial-peer voice 99903 pots
description Analog phone
huntstop
preference 4
destination-pattern 26996
progress_ind alert strip
no digit-strip
port 0/3
dial-peer hunt 2
no dial-peer outbound status-check pots
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
sip-ua
retry invite 3
registrar 1 ipv4:10.10.10.15 expires 3600
g729-annexb overrideCan someone assist me pls.
-
CUE 3.2.3 timezone table
Good afternoon,
Guys is there a table with all the timezones included in CUE 3.2.3 with the beggining and the end of every one?
For example, CUCM 6 has a table like showed below especifing when it begins and when it ends:
TIMEZONE_NEWFOUNDLAND 210 0/11/0/1,0:1:0:0 0 0/3/0/2,0:1:0:0 -60 NST 100
17 E. South America Standard/Daylight Time (GMT-03:00) Brasilia TIMEZONE_AMERICA 180 0/2/0/3,0:0:0:0 0 0/10/0/2,0:0:0:0 -60 BST 101
18 SA Eastern Standard Time (GMT-03:00) Georgetown TIMEZONE_SA_EASTERN 180 0/0/0/0,0:0:0:0 0 0/0/0/0,0:0:0:0 0 BST 102
19 Mid-Atlantic Standard/Daylight Time (GMT-02:00) Mid-Atlantic TIMEZONE_MID_ATLANTIC 120 0/9/0/5,2:0:0:0 0 0/3/0/5,2:0:0:0 -60 AT 110
20 Azores Standard/Daylight Time (GMT-01:00) Azores TIMEZONE_AZORES 60 0/10/0/5,3:0:0:0 0 0/3/0/5,2:0:0:0 -60 WAT 120
21 GMT Standard/Daylight Time (GMT) Greenwich Mean Time; Dublin, Edinburgh, London, Lisbon TIMEZONE_BRITISH_SUMMER 0 0/10/0/5,2:0:0:0 0 0/3/0/5,1:0:0:0 -60 GMT 130
22 Greenwich Standard Time (GMT) Monrovia, Casablanca TIMEZONE_STANDARD 0 0/0/0/0,0:0:0:0 0 0/0/0/0,0:0:0:0 0 GMT 131
23 W. Europe Standard/Daylight Time (GMT+01:00) Amsterdam, Berlin, Stockholm, Rome, Bern, Vienna TIMEZONE_WEST_EUROPE -60 0/10/0/5,3:0:0:0 0 0/3/0/5,2:0:0:0 -60 CET 140
24 GTB Standard/Daylight Time (GMT+02:00) Athens, Helsinki, Istanbul,Minsk TIMEZONE_GFT -120 0/10/0/5,4:0:0:0 0 0/3/0/5,3:0:0:0 -60 EET 150
25 Egypt Standard/Daylight Time (GMT+02:00) Cairo TIMEZONE_EGYPT -120 0/9/4/4,2:0:0:0 0 0/4/5/4,2:0:0:0 -60 EET 151
26 E. Europe Standard/Daylight Time (GMT+02:00) Eastern Europe TIMEZONE_EAST_EUROPE -120 0/10/0/5,3:0:0:0 0 0/3/0/5,2:0:0:0 -60 EET 152
27 Romance Standard/Daylight Time (GMT+01:00) Brussels, Paris, Madrid, Copenhagen TIMEZONE_ROMANCE -60 0/10/0/5,3:0:0:0 0 0/3/0/5,2:0:0:0 -60 CET 141
In Latin America the governments are changing the Summer time every year, in the case of Argentina, there is not a Summer Time anymore.
Because of that we`re having to change the DST from CUCM and CUE from time to time for each country manually.
Thanks in advance.
FelipeHello Ivan,
Just for being more prcise i wanna ask about the mode of CME. It is supposed to work in SRST mode (CME+SRST). It doesn't matter in our case, right?
WHat do you mean by this , it is not clear ???
If you mean that the CME over SRST , the normal call comes from the PSTN and then the dial-peer will point it to the route point on the call manager which is linked to the CTI Ports .
In case of a CME-SRST mode (no link to the CCM) the first dial-peer which points to the CCM will fail , the configuration for a second dial-peer with a lower preference will take place so that it open directly but the session target ip will not be in the second case the CCM as the first dial-peer , it will point directly to the CUE which is normally located on the same router.
And concerning the settings. Are the deal-peer settings that you mentioned in your answer made on CUE side?
The dial-peer on the gateway that the PSTN is terminated on , you create the dial-peers there .
Amer -
Jabber 9.1 Registration To CUCM 8.6.2
Good evening,
I have a problem I am hoping someone can answer for me. I have a CUCM/ Unity Connection/ CUP platform which are all 8.6x
I have configured all the components and I am currently trying to get Jabber to work, I have 1 user who has 2 clients he wants to use. One client is is home Mac and the other his work Windows based laptop.
Jabber 8.6.4 on the Mac works 100% the client can call, access voicemail and use IM with no issues. The same user when he tries to login with his Windows based client gets IM and authenticates to Unity Connection OK but the client does not register with CUCM and therefore he cannot dial out and so all telephony functionality is unavailable. The options in the windows client only show Phone accounts for the voicemail server which all look OK.
It looks to me that the windows client is not getting the correct paramters from the jabber-xml file (infact I cant find one in the tftp server which is CUCM) or maybe I need to have 2 softphones detailed in CUCM, i.e one for each client (MAC & PC). I have all the CCMCIP and trunking etc working fine, along with the relevant AXL, LDAP and CTI users set up ok.
Anyone have an idea of where I may be going wrong ?
Thanks
SHi Jonathan
I have tried your suggestion but while the symptoms are slightly different I am still not able to get the Windows client to register with CUCM.
If I disable the softphone on the mac client then exit Jabber on the mac, then I open the Windows client initially the results are exactly the same. However if I then sign out of the windows client and back in I get the option on the bottom right of the client to choose the softphone source (i.e computer or deskphone) but then no matter which one I choose the client sits at the setting phone prompt and still does not register.
I guess one option for me may be to run a WIndows client on the home Mac using VMware fusion and use the same client, hopefully then the graceful softphone registration may work between devices running the same client?
Many thanks for your reply, very much appreciated.
Best
Steve -
How to Hide Timezone in Month View in Calendar App?
I live in two timezones and go back and forth often so I use the timezone support in Calendar app for Mavericks and on my iOS devices and it's quite useful. However, one thing about it really stinks, imho.
I cannot find a way to hide the timezone information associated with individual events in the month view of the Calendar desktop application. This redundant and superfluous information results in the truncation of the event information, which *is* important to me.
For example, instead of an event in month view being displayed as "check amex payment", as it was before I enabled timezone support, it now says "check a... 3AM GMT+7". Since every event has this appended timezone information displayed next to it the month view of my calendar has become much less useful than it was previously. I don't really need to know how many hours ahead of GMT an event is, I pretty much just want to know what the event is and on what day, in the month view. Now I cannot tell what any event is unless I open it in in week or day view. Very frustrating.
If anyone knows of a way to hide that information in the month view it would be greatly appreciated to post it here.
thanks Apple Community!No one else has this problem or knows of a solution? *weeps quietly*
-
Error while updating phone button template in CUCM 8.6
Experts,
I'm getting following error while updating phone button template in CUCM 8.6;
Update failed. java.sql.SQLException: System catalog (sysprocbody) corrupted.
Please check the screen shot attached here with.
What could be the reason?
Thanks
VivekI'm not a Cisco employee, so I can't do anything with your backup.!
You need to take a backup as a precaution (You should have been doing this anyway) Then you need to call Cisco TAC to get the underlying problem fixed. CalManager is a locked-down environment, and only TAC can get the low-level access needed to fix database problems.
GTG -
Cisco phone proxy will support on cucm 8.6 or not
Hi as per document i can see that Cisco phone proxy . Without using vpn connect . Customer want to keep configured Cucm on there Jabber or cisco mobile on the iphone and need to get connected when ever they have an internet access .
And also by keeping a small end router at branches havin only one cisco phone needs to connect to the corprate office using proxy.
Is that possible and also is it possible on 8.6 cucm version . Here am just attaching a document info which says version supported
Supported Cisco UCM and IP Phones for the Phone Proxy
Cisco Unified Communications Manager
The following release of the Cisco Unified Communications Manager are supported with the phone proxy:
•Cisco Unified CallManager Version 4.x
•Cisco Unified CallManager Version 5.0
•Cisco Unified CallManager Version 5.1
•Cisco Unified Communications Manager 6.1
•Cisco Unified Communications Manager 7.0I understand the VPN solution is a good one, but we have already 100 proxy phones deployed and from what I read I'd need to first set up a VPN phone on the cluster before deploying it in the field. This would be difficult to do with users all over the country. Or am I missing something?
I have a question about our ASA though if anyone can answer it. We have to move from an ASA 5510 to a 5520 due to the 100 UC proxy-phone license limitation (even though we have 70 phones it appears that a UC license is taken up for each TFTP server configured on the phone - primary and secondary). When I configured the 5520 I can register new phones no problem. None of the existing phones will register until the CTL file is manually deleted on each phone. Is there a way to seamlessly migrate to the 5520? Such as using the same certs, CTL file etc.. on the 5520? When I configured it it generated it's own self-signed key and CTL file. I did import the certificates from CUCM... -
Current State:
• I have a customer running CUCM 6.1 and UCCX 7.01SR5. Currently their CUCM is *NOT* LDAP integrated and using local accounts only. UCCX is AXL integrated to CUCM as usual and is pulling users from CUCM and using CUCM for login validation for CAD.
• The local user accounts in CUCM currently match the naming format in active directory (John Smith in CUCM is jsmith and John Smith is jsmith in AD)
Goal:
• Upgrade software versions and migrate to new hardware for UCCX
• LDAP integrate the CUCM users
Desired Future State and Proposed Upgrade Method
Using the UCCX Pre Upgrade Tool (PUT), backup the current UCCX 7.01 server.
Then during a weekend maintenance window……
• Upgrade the CUCM cluster from 6.1 to 8.0 in 2 step process
• Integrate the CUCM cluster to corporate active directory (LDAP) - sync the same users that were present before, associate with physical phones, select the same ACD/UCCX line under the users settings as before
• Then build UCCX 8.0 server on new hardware and stop at the initial setup stage
• Restore the data from the UCCX PUT tool
• Continue setup per documentation
At this point does UCCX see these agents as the same as they were before?
Is the historical reporting data the same with regards to agent John Smith (local CUCM user) from last week and agent John Smith (LDAP imported CUCM user) from this week ?
I have the feeling that UCCX will see the agents as different almost as if there is a unique identifier that's used in addition to the simple user name.
We can simplify this question along these lines
Starting at the beginning with CUCM 6.1 (local users) and UCCX 7.01. Let's say the customer decided to LDAP integrate the CUCM users and not upgrade any software.
If I follow the same steps with re-associating the users to devices and selecting the ACD/UCCX extension, what happens?
I would guess that UCCX would see all the users it knew about get deleted (making them inactive agents) and the see a whole group of new agents get created.
What would historical reporting show in this case? A set of old agents and a set of new agents treated differently?
Has anyone run into this before?
Is my goal possible while keeping the agent configuration and HR data as it was before?I was doing some more research looking at the DB schema for UCCX 8.
Looking at the Resource table in UCCX, it looks like there is primary key that represents each user.
My question, is this key replicated from CUCM or created locally when the user is imported into UCCX?
How does UCCX determine if user account jsmith in CUCM, when it’s a local account, is different than user account jsmith in CUCM that is LDAP imported?
Would it be possible (with TAC's help most likely) to edit this field back to the previous values so that AQM and historical reporting would think the user accounts are the same?
Database table name: Resource
The Unified CCX system creates a new record in the Resource table when the Unified CCX system retrieves agent information from the Unified CM.
A Resource record contains information about the resource (agent). One such record exists for each active and inactive resource. When a resource is deleted, the old record is flagged as inactive; when a resource is updated, a new record is created and the old one is flagged as inactive. -
CUCM Upgrade 8.0 to 8.6 Question
Hi guys,
i have a 8.0.2 CUCM and i want to go to 8.6
I was wondering if rebooting the server after the installion of "ciscocm.refresh_upgrade_v1.0.cop.sgn" can compromise the update because i did install it a week ago and meanwhile the servers had an outage hence rebooting. If so is there a workaround ?
**I havent started the actual update yet just install the above mentioned file**
Thanks,
Eric D.Hi Edu,
It's ok to reboot the server after has been installed. Just one advcie, be sure to install the right COP file because it cannot be uninstalled (Cisco TAC is the only want able to do it).
Regards
Maybe you are looking for
-
The company is operating in 3 states, say andhra pradesh, tamilnadu, delhi. There are several branches in each state. Tamilnadu has 2 branches and andhra pradesh has 5 branches. I've defined the locations, and added the branches as warehouses. My
-
Problem in displaying o/p in Table Control
Hi , while displaying in table control. its is displaying line twice. I have some contains in iti itab. which i am displaying itf itab through table control.. i have coded like this... PROCESS BEFORE OUTPUT. MODULE STATUS_0112. LOOP at itf WITH CO
-
OLTP data to view in pre-built reports
Hi Forum, I have created few records in Siebel CRM eCommunications application to view the data in OBI Presentation services. Using DAC, I want to populate into W_ tables to view in OBI eCommunications pre-built reports. So can you please guide me in
-
Where is the iCloud icon on my MAC?
I just updated to Lion and it automatically set up icloud, but I can't find the icloud icon. Can anyone help me? Thanks!
-
Hi, while debugging a simple application I consistently get a debugger error: dbx: internal error: byteoff too large in 'set_window' If I ignore this error the debugger crushes a step later. I am running Sun Studio 9 on Sun Solaris. Any idea why this