Time Difference In Between Call Manager & IP Phone
Hello,
We remove the NTP server from the Call Manager 6.5 and configure manual TIME setup on the CUCM but the IP Phone showing 7 minutes difference.
We reset and restart the Phones not no effect. What other thing we can do to accomplish this issue.
Thanks,
Hello,
I remove the NTP Server IP Address information from the OS Administration NTP configuration.
Output for the Command
admin:utils ntp status
ntpd (pid 32278) is running...
remote refid st t when poll reach delay offset jitter
==============================================================================
*127.127.1.0 127.127.1.0 10 l 54 64 377 0.000 0.000 0.015
172.16.1.51 0.0.0.0 16 u - 1024 0 0.000 0.000 4000.00
synchronised to local net at stratum 11
time correct to within 11 ms
polling server every 64 s
Current time in UTC is : Mon Nov 4 07:23:48 UTC 2013 Current time in Mideast/Riyadh89 is : Mon Nov 4 10:30:52 zzz 2013
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A question about call manager traces for Sip phones.
So today I create a sip based ip communicator and pressed the new call button and heard a dial tone. I started typing my telephone number. Half way through, I heard another secondary dial tone (which indicates mis-configured route pattern somewhere) .
However, When I look at the call manager logs, I do not actually see the digits that I was typing. With SCCP, I can see the keypad button press messages in the traces, but here, I cannot see the pressed buttons in my CUCM traces. Can anyone help with telling me how I can see button presses going to call manager . All I can see are the logs below which came up as soon as I got the dial tone and the final sip invite messages. I see nothing in-between.
|SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
[6387070,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.x.x.66:56714;branch=z9hG4bK00005b1e
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
Call-ID: [email protected]
Date: Sat, 14 Feb 2015 14:17:40 GMT
CSeq: 19 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:56714;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:[email protected]">
<dialog id="12" call-id="[email protected]" local-tag="00ffb00bc50a003300006390-00002d4f"><state>trying</state></dialog>
</dialog-info>
SIPStationD(12991) - processCommonDialogNotifyInd: Did 12 Sending Notified SIPOffHook to new CdfcHere is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
+++++ Analysis of SIP Phone making a call +++++++++
The user picks up the phone and the IP Phone sends a NOTIFY to CUCM to indicate the start of a new dialog. This dialog begings by an offhook event
00869539.002 |14:58:13.837 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
[46240,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:13 GMT
CSeq: 11 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
<dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
</dialog-info>
++++ CUCM SIP stack processes the new connection for the phone+++++++
00869540.001 |14:58:13.837 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
00869540.002 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
00869540.003 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
++++ Next CUCM allocates a call id for this call +++++
00869546.002 |14:58:13.838 |AppInfo |LineControl(66) - Get call instance=1 for CI=24419584
+++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
00869555.007 |14:58:13.839 |AppInfo |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
00869555.008 |14:58:13.839 |AppInfo |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
00869556.000 |14:58:13.839 |SdlSig |SIPSPISignal |wait |SIPTcp(1,100,71,1) |SIPHandler(1,100,79,1) |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
00869556.001 |14:58:13.839 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46241,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
To: <sip:[email protected]>;tag=1822746380
Date: Mon, 16 Feb 2015 12:58:13 GMT
Call-ID: [email protected]
CSeq: 11 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
00869541.001 |14:58:13.838 |AppInfo |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
00869542.003 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
00869542.004 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Did 6 Sending Notified SIPOffHook to new Cdfc
00869542.010 |14:58:13.838 |AppInfo |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
00869543.000 |14:58:13.838 |SdlSig |SIPOffHookInd
+++ The next thing is the USER dials a digit on the phone ++++++
This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
In this example the user dialled "4" first so we see an "INVITE sip:4@host-IP"
00869559.002 |14:58:14.064 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
[46242,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 INVITE
User-Agent: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 373
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
s=SIP Call
t=0 0
m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
c=IN IP4 10.50.16.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
+++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
00869562.001 |14:58:14.065 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46243,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
From the INVITE cucm concludes that KPML and rtp-nte is supported
00869566.009 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: KPML Supported.
00869566.010 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: Detected inband DTMF support
Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
00869590.001 |14:58:14.067 |AppInfo |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
+++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
00869594.001 |14:58:14.068 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46244,NET]
SUBSCRIBE sip:[email protected]:52910 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 SUBSCRIBE
Date: Mon, 16 Feb 2015 12:58:14 GMT
User-Agent: Cisco-CUCM10.5
Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
Expires: 7200
Contact: <sip:[email protected]:5060;transport=tcp>
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
<regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern>
</kpml-request>
+++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
00869595.002 |14:58:14.118 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
[46245,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=TCP>
Expires: 7200
Content-Length: 0
+++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
00869603.002 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
[46247,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
00869608.001 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46248,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1000 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
At this point I will insert a paragraph from the RFC 4730 for SIP KPML
+++++++++++++
The event package uses SUBSCRIBE
messages and allows for XML documents that define and describe filter
specifications for capturing key presses (DTMF Tones) entered at a
presentation-free User Interface SIP User Agent (UA). The event
package uses NOTIFY messages and allows for XML documents to report
the captured key presses (DTMF tones), consistent with the filter
specifications, to an Application Server +++++++++++++++++++++++++++
00869609.002 |14:58:14.209 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46249,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
00869622.001 |14:58:14.210 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46250,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1001 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Again we get the next digit ++++
00869624.002 |14:58:14.262 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46251,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
00869637.001 |14:58:14.263 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46252,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1002 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Finally we get the last digit ++++
00869638.002 |14:58:14.390 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46253,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1003 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
00869648.003 |14:58:14.391 |AppInfo |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
00869648.004 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
00869648.005 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
00869648.012 |14:58:14.391 |AppInfo |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
00869648.013 |14:58:14.391 |AppInfo |Digit analysis: analysis results
00869648.014 |14:58:14.391 |AppInfo ||PretransformCallingPartyNumber=9106
|CallingPartyNumber=9106
|DialingPartition=
|DialingPattern=4XXX
|FullyQualifiedCalledPartyNumber=4080
|DialingPatternRegularExpression=(4[0-9][0-9][0-9])
|DialingWhere=
+++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
00869701.001 |14:58:14.435 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
[46256,NET]
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
To: <sip:[email protected]>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces -
How to calculate the % of time difference between different state
Hi there,
The below query returns the follwoign outptut. I need to calculate the % of time period that a specific state ( state ex: open or closed or all report )exists.
I have to find the time difference between the 2 states . How could I do that ?
The first row shows null for the previous state . How could get actual previous state for the first row?
SELECT si.station_name,
vppstation.avi_control_state_code.STATE_SHORT_NAME,
ash.state_id ,
lag(ash.state_id) over (order by ash.STATE_EFF_DATE desc) previous_state,
TO_NUMBER(TO_CHAR(ash.state_eff_date, 'SSSSS')) "periods in sec",
TO_CHAR(ash.state_eff_date, 'dd-mon-yyyy hh24:mi:ss am') "Date"
from vppstation.avi_state_history ash
left outer join vppstation.avi_control_state_code
on ash.state_id = vppstation.avi_control_state_code.state_id
LEFT OUTER JOIN vpproadside.stations_installed si
ON ash.station_id = si.station_id
where ash.state_eff_date >= to_date('28/02/2010', 'dd/mm/yyyy')
and ash.state_eff_date <= to_date('03/03/2010', 'dd/mm/yyyy') ;
group by si.station_name
ash.state_id
order by ash.state_eff_date asc ;
STATION_NAME STATE_SHORT_NAME STATE_ID PREVIOUS_STATE periods in sec Date
IRDNCST02 Open 1 NULL 85166 01-mar-2010 23:39:26 pm
IRDNCST02 All Report 3 1 85159 01-mar-2010 23:39:19 pm
IRDNCMT01 Closed 2 3 81376 01-mar-2010 22:36:16 pm
IRDNCST02 Open 1 2 78723 01-mar-2010 21:52:03 pm
IRDNCST02 All Report 3 1 76023 01-mar-2010 21:07:03 pm
IRDNCMT01 Open 1 3 55922 01-mar-2010 15:32:02 pm
IRDNCMT01 Closed 2 1 54931 01-mar-2010 15:15:31 pm
IRDNCHA01 Closed 2 2 41291 01-mar-2010 11:28:11 am
IRDNCAS01 Open 1 2 38847 01-mar-2010 10:47:27 am
IRDNCAS01 All Report 3 1 37947 01-mar-2010 10:32:27 am
IRDNCST02 Open 1 3 35332 01-mar-2010 09:48:52 am
IRDNCST02 All Report 3 1 32632 01-mar-2010 09:03:52 am
IRDNCST02 Open 1 3 31502 01-mar-2010 08:45:02 am
IRDNCST02 All Report 3 1 28802 01-mar-2010 08:00:02 am
IRDNCHI01 Open 1 3 25368 01-mar-2010 07:02:48 am
IRDNCHI02 Open 1 1 23939 01-mar-2010 06:38:59 am
IRDNCMT01 Open 1 1 20696 01-mar-2010 05:44:56 am
IRDNCCH02 Open 1 1 13452 01-mar-2010 03:44:12 am
Edited by: Indhu Ram on Mar 11, 2010 1:34 PM
Edited by: Indhu Ram on Mar 11, 2010 2:20 PMIf you look at the table which is output of the given query , there is column called "STATE_ID" It shows the current state ie open or closed or all report.
The column 'PREVIOUS_STATE' shows the state before the current state. .In the table in 2nd row in previous_state column there is state_id = 1 (open)which is the state before 'all report state' -
I have a Time Capsule of the 4th generation and I am pretty impressed since I just had a fried harddrive on my 3 month old Imac and Time Capsule restored it fully including all programs etc. without having me to put in all serial numbers etc. for that I give it an A+. However what I find very frustrating is the very short and incomplete user manual and setup instructions. I give those a D-. It does not explain the consequences of certain choices to a normel non computer gig user like me:
I do not understand what the consequences and difference is between the security setting WPA/WPA2 and WPA2 when setting it up. It would be helpful if the manual would explain something like: If you set it onto WPA2 then computers older than XXX years will not able to hook up to the internet through your router...or the likes, since I slowly supect this may be the reason for some of my troubles.
I also have a 5 year old PC with vista connected to the Time Capsule. It can wireless connect to the Time Capsule and access the printer (I have managed that after some painful hours...) but it cannot access the internet with that PC through the timecapsule wireless, only if I hook it up with a cable. Why? Does that have to do with the fact that I have set the TIme Capsule to WPA2?
Our PS3 can connect wireless, no problem. However, I friend of mine tried it with a quite new Toshiba Notebook and he could see the Time Capsule but again it shows the line from the Time Capsule to the internet blocked. All other computers linked with data cable to the Time Capsule are connected to the internet no problem. What have we done wrong? Should I change the security setting to WPA/WPA2?
I tried the latter, but then another problem does come up, which was there from beginning on: When I change the settings for my Time Capsule my Imac suddenly cannot find the Timne Capsule anymore, even so I am only 3feet away from it. When I set it up for the first time 3 month ago it drove me crazy. The very first time it found the Time Capsule right away. But when I discuverd a mistake in my initial setup and I went through the settings again to change stuff it suddenly could not find the Time Capsule anymore. Only after frustrating 20 times or so with turning wi-fi on and off several times inbetween did it suddenly find the Time Capsule in the set up menu. However, when I tried changing my security setting to WPA/WPA2 today to see whether this solves my other problems again it could not find the Time Capsule when I was finished with the settings and I had to give up/cancel. Why is that???
I would really appreciate some help here. But please I am not a computer gig and normally always had assistance with these things. On the other hand I am not all stupid either (even so these manuals assuming we all grew up sleeping on a computer make me feel stupid).....
Thank you all in advanceHello Schimi. The Time Capsule is a kind of product that offer you two possibility:
- To use the TC as a router or as an external HD.
The difference between WPA and WPA 2 are:
- They are two kinds of internet connection.
WPA uses dynamic key encryption , which means that the key is constantly changing and makes the invasion of a wireless network more difficult than WEP . WPA is considered one of the highest levels of wireless security for your network and is recommended if your devices support this kind of encryption . Newer routers offer WPA2 security . WPA2 is compatible with WPA , but offers higher security level . In fact, it meets the high standards of many government agencies . If the router and computer support WPA2 , this should be your choice.
If you need any help or if you find some difficult setting up it you can contact Apple Support.
I am sure that they can help you with that.
I am sory for the delay. I holpe that you solve your issue. -
Like my header says.. I have a I Phone 4S.. with recently installed OS 7 software.. One problem has occured.. when someone calls me thier phone rings
3 to 4 times before my phone rings.. I have tested this with several family and friends calling me..
> I have a business this is unacceptable.. does anyone have a solution thanks...If this were happening to me I would try resetting the System Management Controller. Next step would be to take it to an authorized service place.
-
after the basement remodel i will have a server and computer repair area for my testing. I have a call manager server and a couple of ip phones and would like to setup a phone at each of my families houses. what would i need and how would i set it up.
there might be a way to do it using SIP but I have not done it and I am just giving you an idea of how to do it with SCCP default and in a secure way
Forget about the router model I was talking about.
You would have to assume that your home where the CCM is as the Main site and pretty much have all othere families as remote sites, you would need to setup VPN between the houses and if you want everyone to dial each other you would need DMVPN(Dynamic Multipoint).
You would need a good router at the main site, depending on how many sites you have you may be able to use just a 2801. For the remote sites, I would think you can use an 871 or 831 router which does VPN and also has a few PoE ports switch.
You would have to create tunnels between each other house.
JoeL -
How to get the time difference (in seconds) between two timestamp fields?
Hi All,
I'm exploring validation expressions in SAP MDM Data Manager version 7.1.07.245
I have two timestamp fields, suppose Update Date and Update-GDS.
What I'm trying to do is prompt a warning when:
(Update Date minus Update-GDS) is greater than or equal to zero seconds
AND
(Update Date minus Update-GDS) is less than or equal to 900 seconds
the bottomline is, Update Date must be equal or greater than to Update-GDS but Update Date must not exceed 900 seconds difference.
my initial try was:
(Update Date - Update-GDS)>=0 s AND (Update Date - Update-GDS)<=900 s
But it always returns true even the time difference between update date and update-gds is beyond the condition.
Can you please guide me through this?
Thank you very much in advance.
Regards,
~erwinHi Abhishek,
I tried your code but it always returns true.
It seems subtraction of timestamp fields do not actually get the time difference... (in seconds)
Is this possible anyway? Or is there any other better approach on this?
Thanks,
~erwin
Edited by: erwin.j.rivera on Dec 14, 2011 8:40 AM -
Call Manager - Phone register & unregister
How can we be informed if a phone unregisters from call manager? Can we see it in the Real Time Monitoring Tool or snmp ?
ThanksWrong, there is still app log in any linux appliance.
You access it thru RTMT
HTH
java
if this helps, please rate -
View Call Manager Call Real-Time?
Cisco Forum Members:
Is there a way to view Call Manager calls real-time?
The application is that someone picks up an IP Phone, dials an access code to overhead paging, then fails to properly hang up the call as which point their subsequent conversation is broadcast over the paging speakers. Can it be determined which Directory Number is connected to the paging port (a router FXO/FXS port)?
Thank you.
DanThe easiest way would be to look in the GW that houses the paging port with the show call active voice command. For more info look at this link, http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008019ab88.shtml
Please remember to rate all useful posts.
Sent from Cisco Technical Support iPhone App -
Difference between case management and complaint management
Hi ,
Can any one explain the difference between case management and complaint management with some simple example .
Regards,
RahulHi
complaint mngt is for creating, maintaing, processing of complaint documents - when customer calls you and tells about problem with product he bought, you create complaint document in CRM.
case mngt is for managing more documents , it could be documents for many customers. Case is usually about complex problem.
http://help.sap.com/saphelp_crm50/helpdata/en/4c/d7fb717f7b3e4cba12953ec1b9b8c3/frameset.htm
Regards
Radek
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