Trace This: Call Manager RTMT

Cisco Forum:
By any chance, might you know what the RTMT trace files reasons are for the below? I'm chasing a called Directory Number for which the IP Phone does not ring and the call goes directly to Unity Connection voicemail. The end user says the phone is not forwarded, etc. It just does not ring then they get the MWI of a new message.
The RTMT trace shows the call seemingly pointed to the DN but immediately forwarding to voicemail. Between the two occurrences are the line items below. I've yet to produce a Google search that defines what "2" (or another other digits) might be regarding the reason for the call redirection.
Thank you for your comments.
Dan
lastRedirectingReason=2
originalCdpnRedirectReason=2

Hi Dan,
Here are the Reason Codes for Busy, CFNA & CFWDALL. It looks like the user was on the phone
when a call came in and went directly to voicemail. I just tested these out and captured them as I went
along. Sometimes the MWI can be slightly delayed so this is a pretty common question from users.
"Hey...I was sitting here at my desk and my RED message lamp went on....
I missed the call and my phone didn't ring at all"
This used to happed to us on our old Octel as well
Call to a Busy DN - CFB
09:24:34, New Call, CalledId=5126, RedirectingId=5126, Origin=16, Reason=2, CallGuid=A8D82689ADB947708076173B9675D535, CallerName=Rob Huffman, LastRedirectingId=5126, LastRedirectingReason=2, PortDisplayName=CallManager-1-005,[Origin=Invalid],[Reason=Invalid]
09:24:35, AttemptForward
09:24:35, State - AttemptForward.cde!Dummy
09:24:35, Event is [NULL]
09:24:35, PHTransfer
09:24:35, State - PHTransfer.cde!LoadInfo
09:24:35, Event is [PCTREnabled]
09:24:35, State - PHTransfer.cde!RunRoutingRules
09:24:35, Event is [NULL]
09:24:35, ConvRoutingRule
09:24:35, State - ConvRoutingRule.cde!ConvRoutingRules_LoadInfo
09:24:35, Event is [TrueEvent]
09:24:35, PHGreeting
09:24:35, State - PHGreeting.cde!PlayGreeting
09:24:35, Call answered if needed
09:24:35, Playing greeting for Subscriber: bob
09:24:40, No DTMF received
09:24:40, Playing greeting for Subscriber: bob
09:24:40, Event is [RecordMsgEvent]
09:24:40, State - PHGreeting.cde!RecordMsg
09:24:49, Event is [NULL]
09:24:49, State - PHGreeting.cde!RunEditMsg
09:24:49, -->MessageEditing
09:24:49,         State - MessageEditing.cde!CheckMsgMenuOpt
09:24:49,         Event is [EditMessageMenuEvent]
09:24:49,         State - MessageEditing.cde!PlayEditMenu
09:24:49,         Event is [HangupEvent]
09:24:49,         State - MessageEditing.cde!CheckMsgLength
09:24:49,         Event is [ManyEvent]
09:24:49,         State - MessageEditing.cde!SendMsg
09:24:49,         Event is [TrueEvent]
09:24:49,         State - MessageEditing.cde!ConfirmSend
09:24:49,         Event is [HangupEvent]
09:24:49, <--MessageEditing
09:24:49, Event is [HangupEvent]
09:24:49, Idle
This is a Forward No Answer - CFNA
09:26:40, New Call, CalledId=5126, RedirectingId=5126, Origin=16, Reason=4, CallGuid=F65164E144E047FFB75F6343A68D6D3D, CallerName=Rob Huffman, LastRedirectingId=5126, LastRedirectingReason=4, PortDisplayName=CallManager-1-004,[Origin=Invalid],[Reason=Invalid]
09:26:40, AttemptForward
09:26:40, State - AttemptForward.cde!Dummy
09:26:40, Event is [NULL]
09:26:40, PHTransfer
09:26:40, State - PHTransfer.cde!LoadInfo
09:26:40, Event is [PCTREnabled]
09:26:40, State - PHTransfer.cde!RunRoutingRules
09:26:40, Event is [NULL]
09:26:40, ConvRoutingRule
09:26:40, State - ConvRoutingRule.cde!ConvRoutingRules_LoadInfo
09:26:40, Event is [TrueEvent]
09:26:40, PHGreeting
09:26:40, State - PHGreeting.cde!PlayGreeting
09:26:40, Call answered if needed
09:26:40, Playing greeting for Subscriber: bob
09:26:45, No DTMF received
09:26:45, Playing greeting for Subscriber: bob
09:26:45, Event is [RecordMsgEvent]
09:26:45, State - PHGreeting.cde!RecordMsg
09:26:55, Event is [NULL]
09:26:55, State - PHGreeting.cde!RunEditMsg
09:26:55, -->MessageEditing
09:26:55,         State - MessageEditing.cde!CheckMsgMenuOpt
09:26:55,         Event is [EditMessageMenuEvent]
09:26:55,         State - MessageEditing.cde!PlayEditMenu
09:26:55,         Event is [HangupEvent]
09:26:55,         State - MessageEditing.cde!CheckMsgLength
09:26:55,         Event is [ManyEvent]
09:26:55,         State - MessageEditing.cde!SendMsg
09:26:55,         Event is [TrueEvent]
09:26:55,         State - MessageEditing.cde!ConfirmSend
09:26:55,         Event is [HangupEvent]
09:26:55, <--MessageEditing
09:26:55, Event is [HangupEvent]
09:26:55, Idle
Call Forward All to VM - CFWDALL
09:29:49, New Call, CalledId=5126, RedirectingId=5126, Origin=16, Reason=8, CallGuid=3691A86D05BA4A08A3FFD009E416DFA4, CallerName=Rob Huffman, LastRedirectingId=5126, LastRedirectingReason=8, PortDisplayName=CallManager-1-002,[Origin=Invalid],[Reason=Invalid]
09:29:49, AttemptForward
09:29:49, State - AttemptForward.cde!Dummy
09:29:49, Event is [NULL]
09:29:49, PHTransfer
09:29:49, State - PHTransfer.cde!LoadInfo
09:29:49, Event is [PCTREnabled]
09:29:49, State - PHTransfer.cde!RunRoutingRules
09:29:49, Event is [NULL]
09:29:49, ConvRoutingRule
09:29:49, State - ConvRoutingRule.cde!ConvRoutingRules_LoadInfo
09:29:49, Event is [TrueEvent]
09:29:49, PHGreeting
09:29:49, State - PHGreeting.cde!PlayGreeting
09:29:50, Call answered if needed
09:29:50, Playing greeting for Subscriber: bob
09:29:54, No DTMF received
09:29:55, Playing greeting for Subscriber: bob
09:29:55, Event is [RecordMsgEvent]
09:29:55, State - PHGreeting.cde!RecordMsg
09:30:06, Event is [NULL]
09:30:06, State - PHGreeting.cde!RunEditMsg
09:30:06, -->MessageEditing
09:30:06,         State - MessageEditing.cde!CheckMsgMenuOpt
09:30:06,         Event is [EditMessageMenuEvent]
09:30:06,         State - MessageEditing.cde!PlayEditMenu
09:30:06,         Event is [HangupEvent]
09:30:06,         State - MessageEditing.cde!CheckMsgLength
09:30:06,         Event is [ManyEvent]
09:30:06,         State - MessageEditing.cde!SendMsg
09:30:06,         Event is [TrueEvent]
09:30:06,         State - MessageEditing.cde!ConfirmSend
09:30:06,         Event is [HangupEvent]
09:30:06, <--MessageEditing
09:30:06, Event is [HangupEvent]
09:30:06, Idle
Cheers!
Rob
"Every fool's got a reason to feelin' sorry for himself" - Springsteen

Similar Messages

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 Cdfc

Here 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

CDR FIles in Call Manager version 7.1.5-30000

Good Morning:
I have a cluster of servers Cisco Call Manager version 7.1.5.30000; 01 Publisher and 02 subscriber. The publisher server passes CDR files to a call accounting application PC-SYSTEL on another server. At one point the CDR files are no longer spend the Call Manager to the application PC-SYSTEL
Call manager services are reviewed, the part of CDR Management in Cisco Unified Serviciability right. The only solution is doing a RESTART to PUBLISHER server, and then again go CDR files to the application PC-SYSTEL
Please anyone has any solution other than to give a RESTART PUBLISHER server, or is this the only way to restore this service.
Expect your news.
Thanks in advance for any ideas.
Regards.
Fernando.

Hi Fernando,
Please check the following.
(1) Confirm that the billing server is configured correctly (verify the IP address). From the Cisco Unified Serviceability page, navigate to Tools > CDR Management to view the billing server parameters.
(2) Check if you can ping the billing server from the Publisher. From the Publisher CLI, you can use the following command:
utils network ping a.b.c.d
where a.b.c.d is the IP address of the billing server
(3) Check if the FTP (or SFTP) service/software running on the billing server.
(4) Check if the billing server is running out of disk-space.
(5) Check if the CDR Respository Manager service is running on the Publisher. If not, start the CDR Repository Manager service.
(6) On the Publisher, check the 'preserve' and 'destinationX' (X can be 1, 2 or 3) directories for CDR flat files. For example: if transfer of CDR files to billing server is broken from Jan 6th, then use the following commands:
file list activelog /cm/cdr_repository/preserve/20110106
file list activelog /cm/cdr_repository/destination1/20110106
(if multiple billing servers are configuration, the location will be similar to above, but the corresponding directories will be named destination2, destination3). This location contains symbolic links to files under preserve. CDR Repository Manager service uses these soft-links to determine what files need to be transfered to billing server.
If you don't observe files in either the 'preserve' or the 'destinationX' locations for the specific date, then it means that problem does not reside on the Publisher. Either the CDR records were not generated by the server that handled the call or it could mean that CDR Agent service on that server could not transfer flat files to the Publisher. Was the 'CDR Enabled Flag is enabled in CallManager service' enabled on that particular date ? Is the CDR Agent service running on the node that handled the call ?
If you observe files in the 'preserve' directory for the specific date, try restarting the CDR Repository Manager service on the Publisher. If the file transfer to the billing server does not commence, then set the trace level for the CDR Repository Manager service to Debug, wait 10 minutes, download CDR Repository Manager logs via RTMT and engage Cisco-TAC for further troubleshooting.
HTH
Manish

Cisco Call manager down

Suddenly morning we can't access the call manager via https or ssh ,then we did hard reboot the server then its work fine.
After the system come up then we take the RTMT logs to find out the what root causes of server was hung.
As per the log file, I didn't found any thing related to ccm hung or error.
Attached the log file, Please can you help me.

Hi,
If you want help can you precise some points :
What node on your cluster was unreacheable ?
When it was unreacheable did you have try to ping it ? If yes did you have a response of this test or not ?
In the same time of this issue do you had another equipement unreacheable (switch, router or another server) ?
Your problem can be network congestion not only a problem on your CUCM.
Regards,
Fabien.

Cisco Call Manager TFTP requests.

I was looking to see if there was a way to look at what devices request config files from a Cisco Call Manager TFTP server.
I know that you can use RTMT to look at the total number of requests made, but is there a way to know what device make the request.
Thanks in advance,
Robert

You could do this by doing a packet capture on the server as described in the link below
https://supportforums.cisco.com/docs/DOC-11599

Error on Call Manager 7

Hi all we have 2 Call Manager 7.1.2.31900-1 boxes. Recently we had a power outage and the servers were not shut down properly. One remaining issue we are having trouble resolving is this error below. It appears in the logs every 15 seconds or so. Any ideas on how to resolve it? I think it might be a corrupt file or file permissions but I can't access the real unix shell so how does one fix this sort of thing?
Cisco CallManager: Not able to create Trace path /var/log/active/cm/trace/dbl/sdi/ccm/. Ensure that the path is valid and the application user has appropriate security settings for this path"
Thanks much in advance

Hi,
It looks like a rebuild is needed. Please check the following
https://tools.cisco.com/bugsearch/bug/CSCth53322/?reffering_site=dumpcr
Symptom:
1)System had a non-graceful shutdown.
2)System is in a running state but part of the file system is corrupt causing certain features to fail.
Conditions:
Server goes down non-gracefully for any reason.
A non-graceful shutdown is any time the server is powered down by means other than through the GUI or CLI command "utils system shutdown"
Cisco recommends a UPS in the release notes already to avoid this.
Workaround:
Rebuild the server if their is a non-graceful shutdown regardless if the recovery disk was required. As any non-graceful shutdown can cause file system corruption even if no symptoms are seen immediately.
HTH
Manish

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 ?
Thanks

Wrong, there is still app log in any linux appliance.
You access it thru RTMT
HTH
java
if this helps, please rate

Issue with User web page into Call Manager

Hello
I have an issue where all my users can log into their User page on UCM but when they make a change to speed dial or anything else, the change does not reflect on the phone in UCM. Doing a reset from their web page doesn't reset the phone. This is UCM 7.1.5 and the standard ccm end user group is enable for all end users.
thanks

Hi Bill -
Have you verified your User Management Roles and effective permissions? Go to User Management - Roles - select Standard CCMUSER Administration and for the permissions you want, ensure both Read and Update are selected.
Role Information
Application
Cisco Call Manager End User
Name
Description
Resource Access Information
">Resource Description Privilege
read                                               update
CCMUser: Access List
read                       update
CCMUser: Device
read                       update
CCMUser: Directory
read                       update
CCMUser: Fast Dials
read                       update
CCMUser: IP Phone Services
read                       update
CCMUser: Line Settings
read                       update
CCMUser: Personal Address Book
read                       update
CCMUser: Plugins
read                       update
CCMUser: RemoteDestination
read                       update
CCMUser: Service URL
read                       update
CCMUser: Speed Dial User
read                       update
CCMUser: User Settings
read                       update
You can also check an individual's effective permissions by using these procedures:
Viewing a User's Roles, User Groups, and Permissions
This section describes how to view the roles, user groups, and permissions that are assigned to a user that belongs to a specified user group. Use the next procedure to view the roles, user groups, and permissions that are assigned to a user in a user group.
Note: You can also view user roles by using User Management > Application User (for application users) or User Management >End User (for end users) to view a particular user and then display the user roles.
Choose User Management > User Group.
The Find and List User Groups window displays.
Find the user group that has the users for which you want to display assigned roles.
Click the name of the user group for which you want to view the roles that are assigned to the users.
The User Group Configuration window displays for the user group that you chose. The Users in Group pane shows the users that belong to the user group.
For a particular user, click the i icon in the Permission column for the user.
The User Privilege window displays. For the user that you chose, this information displays:
User groups to which the user belongs
Roles that are assigned to the user
Resources to which the user has access. For each resource, this information displays:
Application
Resource
Permission (read and/or update)
Now if both items above look OK, you might check your DB replication status. I assume you have a Publisher and one or more Subscribers? User phones registered to Subscriber? You can check replication several ways:
CLI
RTMT
Unified Reporting on CUCM administrator web page - select "Unified CM Database Replication Debug" report. This is the easiest.
The desired Replication State is 2.
Here is some further information:
Check the DB replication status on all the Cisco Unified Communications Manager nodes in the cluster to ensure that all servers are replicating database changes successfully. You can check by using either RTMT or a CLI command.
• To check by using RTMT, access the Database Summary and inspect the replication status.
• To check by using the CLI, enter the command that is shown in the following example: admin: show perf query class "Number of Replicates Created and State of Replication"
==>query class :
- Perf class (Number of Replicates Created and State of Replication) has instances and values:
ReplicateCount -> Number of Replicates Created = 344 ReplicateCount -> Replicate_State = 2
Be aware that the Replicate_State object shows a value of 2 in this case. The following list shows the possible values for Replicate_State:
0—Replication Not Started. Either no subscribers exist, or the Database Layer Monitor service is not running and has not been running since the subscriber was installed.
1—Replicates have been created, but their count is incorrect.
2—Replication is good.
3—Replication is bad in the cluster.
4—Replication setup did not succeed.
I'm thinking it's more your Roles/permissions and not replication, but I included just in case. Hope this helps!
Ginger

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

Call Manager 8.0 to 9.1 upgrade

We are currently running Call Manager 8, UCCE 8, and CVP 8. ICM/CTI 8.
We would like to upgrade Call Manager 8 to 9.1 first before upgrading UCCE, CVP, etc., it could be months before these are upgraded.
Does anyone know or foresee any issues if UCCE, CVP, etc., are not upgraded right away after CM is upgraded?
Any comment is appreciated. Thanks.

Look at the UCCE compatibility matrix as you need the exact versions to find out whether they will work together, of ir you'll need a single window on which to upgrade all.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Upgrade Call Manager from 8.6.2 to 10.5

Hello All,
We are planning to upgrade a Cisco Call Manager Publisher node from 8.6.2 to 10.5. We want to install a new publisher on a new environment, but we are struggling with some questions.
If we install the Pub node 10.5 can we migrate the configuration from the 8.6.2 to 10.5? Do we need a special tool for this?
If above doesn't work, we can still migrate the existing call manager towards the new servers and start the upgrade, the only problem we face there is that we have to change the publisher's IP because we want to use a new ip addressing scheme.
Kr,
Yannick Vranckx

Your best bet is to use the new tool: cisco prime collaboration deployment. This fits perfectly into what you want to do here and can easily help you with all aspect of the migration including the ip address change. You can learn how to use the tool here..
https://www.youtube.com/watch?v=JzG4kz1_hL4

Call Manager Migration 7.1.5 to 9.1 / Trunking : ICT vs. SIP

Hello All,
Currently studying for CCNA Voice and have been asked by my current employer to upgrade CUCM 7.1.5 to 9.1.1. There has been a time frame put on the deployment however I want to dig pretty deep into the deployment to learn as much as possible. I am going to start by configuring a trunk between the 2 call manager clusters. from there I will add users to the new call manager and try to convert slowly to the new call manager instead of one big cutover. I will place a route pattern over the trunk so that cucm 9.1 will route through cucm 7.1 while I slowly convert users to the new system. I will then deploy directory numbers in a Staging partition that is not associated with any CSS. Once I have all users and phones configured on the new 9.1 call manager I want to use the bulk admin tool to change the staging partition to the working internal_PT production partition. The users will be imported from LDAP. My questions are, does this sound like a feasible plan? Is there any other difference between a SIP trunk and ICT besides the SIP and H.323 protocol? ( I prefer SIP from what I have read) and will having the same users on both boxes interfere with call processing, if these users and phones are not active yet?

Just for the record. I found the solution to my problem. Checking more logs I read this:
The installation has encountered a unrecoverable internal error. For further assistance report the following information to your support provider.
"/usr/local/cm/script/cm-dbl-ontape_backup-install RU PostInstall 9.1.2.11900-12 7.1.5.30000-1 /usr/local/cm/ /common/component/database /common/log/install/capture.txt " terminated. Exceeded max time (240)
The system will now halt.
So I accessed the Dissaster Recovery Section on CUCM and deleted the tape backup device that was configured there. After deleting it the upgrade went well.

Can't remove registered ephone in call-manager-fallback

This ephone and dn keeps registering so long as call-manager-fallback is not shutdown.
RTR001#show ephone registered
ephone-1[0] Mac:0FD4.9DA0.D415 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 6/5 max_streams=1
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.32.21.183 * 26602 SCCP Gateway (AN) keepalive 4 max_line 2 available_line 1 dual-line
port 2/0/21
button 1: cw:1 ccw:(0)
dn 2 number 4851 CM Fallback CH1 IDLE
Preferred Codec: g711ulaw
Lpcor Type: none
The MAC 0FD4.9DA0.D415 identifies port 21 on a Cisco VG224. After shutting down that voice-port, the ephone doesn't register when call-manager-fallback is enabled.

Well, that is one idea that I've already had, Linc, but I'm reluctant to use the "nuclear option" for obvious reasons. I'm actually wondering now if the Secure Cert / OD problem is affecting Profile Manager. See this thread: https://discussions.apple.com/message/23686348#23686348

Aging out user profiles in Call Manager

If a user profile is deleted in AD what is the length of time before that user is “aged out” / removed from the CUCM and CUC user DB?
Is there a difference between how it works in 4.1 with Cisco Customer Directory Configuration Plug In vs. LDAP Synchronization in later version?
My understanding is that in 4.1 the Plug In extends AD database to Call Manager and it is not embedded in Call Manager as it is in 5.0 and later. Not sure how that plays on removing users from CUCM through synchronization.
Additionally, my understanding is when a user profile is removed from AD, it gets flagged during Synchronization and that could take a minimum of 6 hours after sync for the user profile to be removed in CUCM or whatever the synchronization is scheduled to.

Oli,
It all depends on your sync frequency. If you remove users from AD, CUCM will mark them as "delete pending".
CUCM garbage collector will rip these out 24 hrs later.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080a57c4c.shtml
Every night at 3.15 am, an internal process called the Garbage Collector service runs. This process permanently deletes any account that has been in the
Inactive – Delete Pending state for over 24 hours. The Cisco Unified Communications Manager does not sync Active Directory passwords. Cisco Unified Communications Manager has no knowledge of Microsoft Active Directory encryption mechanism. Instead, in Cisco Unified Communications Manager 5.0, a default password of ciscociscoand a default PIN of 12345are assigned.
cheers
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Call Manager will not activate in CUCM 7

i have installed cucm 7 and can get to it with windows IE, but when i try to activate call manager in the cisco unified servicability it give me this error:
Update Failed for the Service(s): Cisco CallManager Request unsuccessful to license manager(Please check the Licensing logs for further details)
i can not add any phones or do anything.
can someone help.

Hello Sanil,
You have to check your services on your publisher and subscriber , or you can use ssh and login to your publisher and check if the replication between your pubisher and subscriber is perfect or not. Most likely services not started yet.
Br,
Nadeem

Trace This: Call Manager RTMT

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