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

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My Question would be is there a limit of time which you can play any song with out paying for it and use it in a podcast legally.
Thanks Mike

Legally you cannot use copyright music without written permission or a licence from a recognized authority, no matter how short the clip. Many people do this and get away with it (and for full tracks too) and it's debatable how much notice anyone is likely to take, or whether they would be bothered about a short clip. But the legal position is, it's not legal.

SIP- h323 in a AS5850 - Not able to send h323 calls coming from a SIP Phone

Dear All!
I have an AS5850 configured as a SIP Gateway and as a H323 Gateway. I'm planning to use this equipment as an interconnection point between PSTN,SIP and H323.
I already have a functional H323 Network with ISDN trunks to the pstn and it is working fine. I added SIP configuration to the AS5850 in order to be able to route calls out to the PSTN or H323 remote ends coming from a SIP Phone registered with a third-party SIP Proxy.
When the calls coming from the SIP Phone goes to a PSTN destination the calls completes properly, but i am having problems trying to send calls coming from the SIP phone to a remote h323 gateway(also cisco)
Attached is my configuration and the error i'm getting in my cdr. It seems that the "ext" number of the phone is being used as destination string in the last call leg, but i'm not sure.
Please Help!
dial-peer voice 100 pots
application session
destination-pattern 5T
port 2/6:D
forward-digits all
dial-peer voice 102 pots
application session
destination-pattern 044T
port 2/6:D
forward-digits all
dial-peer voice 103 voip
application session
incoming called-number 001T
destination-pattern 001T
session protocol sipv2
session target ipv4:20X.21X.17X.1X
tech-prefix 10511
sip-ua
sip-server ipv4:20X.6X.14X.18X
CDR ERROR:
.Mar 24 2004 18:31:42.620 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9F74CE17 7D2A11D8 82A09B41 D2C3D418, SetupTime .18:31:42.470 GMT Wed Mar 24 2004, ***PeerAddress 2006***, PeerSubAddress , DisconnectCause 3 , DisconnectText no route to destination (3), ConnectTime .18:31:42.620 GMT Wed Mar 24 2004, DisconnectTime .18:31:42.620 GMT Wed Mar 24 2004, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Thanks.
Attached you can find the debug ccsip messages output.

There are 2 solutions here.
1. Use of SIP/H.323 Signalling Gateway as the protocol convertor. Search google will yield heaps of hits on this subject. Product available both commercial and open source, trial, etc. Using this method means that the SIP End Point will communicate with H.323 End Point without going out the PSTN. I believe this is what you want to achieve in the long term. You are trying the AS5xxx as the protocol convertor for you, which it will not work. A call flow will be something like SIP IP Phone->SIP Server->SIP-to-H.323 Gateway->H.323 Gatekeeper->H.323 End Point. Of couse there is a SIP server that do the protocol convertor in the same box but the functionality is the still the same. Performance and concurrent call setup differ from products to products. Going for this solution would require you to find such products and test it on the your network.
2. If you do not wish to try on Soluton 1, this solution is a workaround way by not getting device but using the existing equipment that you have right now. Onto whether this good long term solution for depends on what you want to achieve both in term of commercially and technically. A call flow will be SIP End Point->SIP Server->Voice Gateway (AS5xxx)->PSTN Switch(ISDN/PRI)->Voice Gateway->H.323 Gatekeeper>H.323 End Point. The key is the Voice session must traverse the ISDN link. In other words your dial pattern must be setup is such as way that will go out thru the dial peer pots to pstn switch then come back to another dial-peer pots. I am not saying this is the most efficient way of doing it, I merely suggesting a workable way to achieve your desired goal without soluton 1.
Hopes you get better understanding now.
Thanks
SSng

Some questions about Skype Manager and Skype Conne...

Hallo,
I found one of your post:
That, if I have some questions I may contact with Contact our solutions team
Sadly that link wont work...
I have few questions and also I contacted with Microsoft support, but sadly they wasn’t able to help me. Only what they suggests, that I may contact with Skype Team trough to Forum, or Facebook...
So, I choose first one and maybe you can help me, or redirect my question !?
About my questions:
As I understand Skype for business is just name what contain two free software, like Skype manager and Skype connect. Also those are free, without charge ?
Also I read that there are two different accounts, like Premium and regular one. Premium contains some benefit. So, If I use that Skype for business manager system to I may or can add some premium accounts for some my staff and leave others with regular account ? I don’t have that manager systems installed jet, but what I read, I understand that I maybe just split credit - or is there way how I may also share regular and premium conots ?
Also, Right now my customers have 25 workers, but what - if soon my another customer also want to use Skype Service, and he have 50 - 100 workers ? Will be there special price for different amount of users on one firm ?
And at last, firm who I represent right now are using Elion VoiP (Elion, is Estonian ISP) service, trough to xLite program.
Sadly that one wont work properly, it will hack, or you cant hear other people and so on. With, Skype everything works fine. So, if he will starting to use Skype manger and will make account for every workers (sips) can he bound those sips with same VoiP numbers what he will use right now with Elion ? Or if not, is there some way how to redirect those numbers (to lose those numbers, is sadly out of question) ?
I hope, you can help me guy´s.
Best regards
Rainer

Tere Rainer,
indeed we don't have the solutions team offer any longer. Let me try to answer some of your questions here.
Skype Manager and Skype Connect itself are free of charge. But obviously you can purchase and distribute paid products via Skype Manager.
Yes, you can freely distribute Premium subscription to Manager group members and leave others without. You can do the same with all kind of Skype products, e.g. Skype numbers, calling subscriptions etc.
There are no discounts with Skype Manager purchases. It's a tool to simplify administration.
Unfortunately you cannot transfer numbers to Skype connect. For more details on which features are supported and how they work please have a look here: https://support.skype.com/en/faq/FA10549/what-is-skype-connect-and-how-does-it-work
I hope my answers could help a bit already.
Tervitustega Mustamäelt.
Follow the latest Skype Community News
↓ Did my reply answer your question? Accept it as a solution to help others, Thanks. ↓

Questions about Access Manager tutorials available in netbeans site

Hi
Thank you for reading my post
I have some questions about two tutoral which i find in :
http://www.netbeans.org/kb/55/amsecurity.html and
http://www.netbeans.org/kb/55/amsecurity-liberty.html
here is my problem :
we have some web services, now we want to have authentication applied for consumer who try to access our web services.
we need to have most possible flexibility because we may deploy the server for a customer with an already established Identity database ( Database Table with user details)
Also we need to have Transport level security using SSL.
I read and studied both of them and now i have some questions :
-I think Securing Web Services Using the SAML or UserNameToken is what we need for authentication and autorization of web service consumers?
is that right?
-Does Sun Java System Access Manager provide flexibility to authenticate user/password with a database table content?
-How we can apply roles in Sun Java System Access Manager when we authenticate users ?
Thanks

Imagine that we want to have an end to end security for our web services
we thought that we could use message level encryption to protect the soap message and also we should protect our web services from un-authenticated acess,
we will use userName token for this.
Our customer has large database which contains many user/password and role of those users.
some of web services should be available to higher role (manager) and not for all users.
so we should check a user role before we allows him/her to access a web service.
my question is whether Sun Access manager can help us with this? or there are other configuration or packages that we should apply to have this feature.
to explain more :
our client side is a swing application, users enter username/password to login into system. after they loged in, we send user/pass every time user want to request some data from some services. (is it good to send user/pass every time?)
We want Sun Access Manager to handle users authentication .
We also need to handle role related authorization, can Sun access manager handle this?
Thanks

Some Question about solution manager

Question about Trusted/Trusting system solution manager
1. What is the different between:- thease RFC SAP created automatically ? can we create ?
SAPNET_RFC
SAPNET_RTCC
SAPOSS
SM_NSMCLINT100_BACK
SM_NSMCLINT100_TRUSTED
TRUSTING_SYSTEM@NSM
2. How EarlyWatch Report works, Is that Job run first in R/3 system or solution manager, and what job would be?
3. In Tr: SDCCN "ToDo" where this data comes from ? what posible resons to check if data is not there in R3 system
4. SM_CLNT100_BACK & SM_CLNT100_TRUSTED has login user: solman do we need to put password for that ?
5. In R3 NQA system SMT1 there is no Trusted systems show: when create: messages "No authorization to logon as trusted system"
Thanks in Advanced

ok

Question about calls rolling over

Sorry for the noob question but here goes. I have Call Manager v7. My question is if I have 3 phones with one ext 0009 and I want the calls to go to the first phone and if that person isn't there go to the next phone and then if that person isn't there go to the last phone. If the last person isn't there, have it go to vm. Hopefully that made sense. These are all 7941 phones.

Hi yulook,
You would need to take the Shared Support Line/DN 0009 and make
it a new Hunt Pilot DN. Then on the Level 1,2 & 3 phones you would put
a new DN on Line button 1. Shared lines like 0009 will always ring all
phones simultaneously in your current config.
In the Line Group that is associated with the new Shared Support Hunt
Pilot you can then set a Top Down algorithm that routes to Level 1, Level 2
and Level 3 DN Line Group members in the chosen order
Line Groups
Line groups contain one or more directory numbers. A distribution algorithm, such as Top Down, Circular, Longest Idle Time, or Broadcast, associates with a line group. Line groups also have an associated Ring No Answer reversion timeout value.
The following descriptions apply to the members of a line group:
An idle member designates one that is not serving any call.
An available member designates one that is serving an active call but can accept a new call(s).
A busy member cannot accept any calls.
For information on configuring line groups,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_2_3/ccmcfg/b03lngrp.html
Hunt Lists
Hunt lists comprise ordered groupings of line groups. A line group may belong to more than one hunt list. Hunt pilots associate with hunt lists. A hunt list may associate with more than one hunt pilot.
For information on configuring hunt lists,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_2_3/ccmcfg/b03htlst.html
Hunt Pilots ..this is what 0009 would become.
Hunt pilots are sets of digits. They comprise lists of route patterns that are used for hunting. A hunt pilot can specify a partition, numbering plan, route filter, and hunt forward settings. A hunt pilot must specify a hunt list.
For information on configuring hunt pilots,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_2_3/ccmcfg/b03htpil.html
Cheers!
Rob

Question about required workshop classes for OCP

Im working on finishing up my OCP in a few months. Quick question about the workshop class. Is there a test at the end of the week?
Also how do most people pay for these classes? They are pretty steep and my company doesnt pay for stuff like this for me. Im paying totally out of my own pocket. Also why is the online class the same amount as the instructor lead class? Its not like im in a room at some building that Oracle has to lease for use or something.

RedDeuce wrote:
Im working on finishing up my OCP in a few months.What Certificate exactly are you working towards?
Also how do most people pay for these classes? They are pretty steep and my company doesnt pay for stuff like this for me. Im paying totally out of my own pocket.Then talk to your employer again or maybe look for other jobs that support your plan better.
Also why is the online class the same amount as the instructor lead class? Its not like im in a room at some building that Oracle has to lease for use or something.They know that you might save on expenses for travel and hotel etc.

Question about source system copy for BW

Dear all,
I have a quick question about copying a source system for BW.
We have two clients in the source ERP:
100 Customizing
200 Testing
Our BW is only linked to client 200, so when we now delete client 200 and make a copy from 100 to 200, are we then still able to restore the source system connection so that our extraction will work, or will it fail, as in client 100 I have no BW related (except the cross-client-objects like datasource) information, that can be used for a possible restore of this connection.
What is the procedure in this case?
Thanks,
Andreas

Unfortunately, you will lose your source connection and need to create a new one in R/3. You can use the same source system name in BW, but check the source system connection after the copy (SM59). Also, I would "Restore" the source system (in BW, right-click on Source system in RSA1, and click "Restore") to insure the datasources get replicated from R/3 to BW.
Depending on how much you want your BW and R/3 systems to be in synch, you will need to rebuild all your data providers (cubes/ODS's) from your R/3 system. This means deleting all the data and running the initializations from scratch. At the very least, you will need to rebuild the delta queues.
Curious, if 100 is just for customizing, is there any data in it? Are you just creating a new blank client 200?

Some question about Toshiab reocvery disk for Satellite P300-133

Hello guys ,
I got some questions about unpleasant recovery surprises :
*1)* I just made recovery DVD using Toshiba Recovery Disk Creator and it initially asked for 2 DVDs - but when the first one was done it didn`t require a second blanc DVD . Is it possible all system files of Windows Vista to be written on 1 DVD ?
*2)* Could I reinstall Windows Vista using the HDD Recovery option ( by pressing F8 ) without using recovery DVD ?
*3)* Which HDD partitions`ll be formatted ( just system C: or both C: and E: ) using HDD recovery ?
*4)* What is recovery DVD - Image of the hard drives or Vista installation files ?
Its pretty bad idea not to include OS on DVD when I have paid for it.
Thanks for all the problems to Toshiba :(

1. I think the recovery disk creator means 2 CDs. I have a Portege M700 with Windows Vista and one DVD too. I recovered it and after this I have a clean Vista installation with all drivers and tools from Toshiba. Nothing is missing.
2. Yes, you can reinstall Vista using the HDD Recovery option when you press F8 at startup. You dont need the recovery DVD but you should create one when your HDD get some errors or something else. If you need a new Recovery DVD, you have pay for one.
3. The complete HDD and partitions will be formatted, so backup your data first on a DVD, external HDD, USB stick or something else.
After the reinstallation you have the factory settings.
4. The recovery DVD is an image of the HDD about the factory settings. If you make a recovery, there are all drivers and tools already installed. Of course on the disk are the installations files of Vista, but you can only use this DVD on your Satellite P300. The disk can not be used on an other Satellite or on a PC.
When you have more questions, have a look in the user manual. There are a many in formations about the recovery. Its in chapter 3.

Question about calling batch file by using the System Exec+.vi?

Hi
I have a problem about calling batch file. I know that the system exec is equivalent to "run" in Windows. I called the batch file c:\rtxdos\bs\ch0.bat successfully in the "run" but it didn't work in the LabVIEW program. The dos prompt had an error message "Bad command or file name" and it just happen when I call this batch file in LabVIEW. Why?
Bill.

Hi,
Try to set the "working directory" parameter of System exec.vi to the directory where the batch file is located. It may help.
Good luck.
Oleg Chutko.

Which Ports do MCU and Call Manager use for Video?

Hi, Does anyone know which ports the MCU and call Manager Use when using the 3511 MCU box with Call Manager 4.1.3. I am looking to open the appropriate ports on the firewall.
Regards
Dominic

Dominic,
If your MCU ports are Callmanger controlled, they will use Skinny (SCCP) to communicate on TCP port 2000. Please see the following link.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801a62b9.shtml
Hope this helps. If so, please rate the post.
Brandon

Call manager and Cisco IP phones

I would like to know if it's possible to use Cisco IP phones in small environments, without having Call manager, or it's mandatory to have always CallManager if one wants to use the IP phones.
Thank you

You can use Call Manager Express, which runs on cisco 1751/60, 2600 and above routers. it can support up to 120 users. Cisco Unity Express will provide voice mail. this is a network module in 2600 and above routers. for more info, see www.cisco.com/go/ccme

A question about call manager traces for Sip phones.

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