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

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Call Roll Over

Before I switched to Digital Voice, I had two lines where the 1st line would roll over to the second line if it was busy. How do I do this in Digital Voice?

The closest would be the "Locate Me" feature. You can program this by logging into your account at www.verizon.com/fiosvoice . Also on this login page there is a link for the user manual in the lower right hand side. Once logged in, click on "Calling Features". From there you can click on the "Locate Me" feature to turn it on and assign the list of numbers to call.
Anthony_VZ
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Question about calling batch file by using the System Exec+.vi?

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Question about calling a function.

Im trying to call a function when a button is clicked, im using actionlistner() on the button, which looks as so :-
private void jButton1ActionPerformed(java.awt.event.ActionEvent evt) {
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Doo

So what is your question?
Besides:
- the things are called "methods"Hello There!
Sorry, i'm trying to call a 'method'. I have buttons numbered 0-9 which enter into a text field once clicked using ActionEvent. At the moment each button has its own block of code which is a bad habit as its the same code over with one differance, the number value.
Am i right in thinking i should create one method? Then calling this method on each button.
The problem i have is i am unsure of the code to call the method within each buttons ActionEvent ie:-
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Doo

Question about Model Driven over RTMPS configuration

I am testing model driven development features these days and encountered a problem:
When I used RTMP channel to retrieve data from the server, everything works fine, however when I tried to use RTMPS channel, I got following error:
Could not initialize DataService.
Missing or invalid configuration for destinations: ["CodeModel.Code"]
the CodeModel is the model I created by using the modler under Flash Builder4, and it simply include one table called Code. I can retrieve data, update and delete data with this model over RTMP channel, but not over RTMPS channel.
I set up my RTMPS channel by creating a self-signed certificate and installed it in the trusted area of the browser, then I referenced the keystore file in the service-config.xml, here is the snap of my configuration:
<channel-definition id="my-rtmps" class="mx.messaging.channels.SecureRTMPChannel">
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[LCDS]Deserializing AMF/RTMP request
Version: 0 "connect"
1.0
(Command method=connect (2) trxId=1.0)
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      audioCodecs = 3191.0
      videoCodecs = 252.0
      videoFunction = 1.0
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      objectEncoding = 3.0
    true
    "nil"
    (Typed Object #1 'flex.messaging.messages.CommandMessage')
      operation = 5.0
      correlationId = ""
      clientId = null
      body = (Object #2)
      headers = (Object #3)
        DSMessagingVersion = 1.0
        DSNeedsConfig = true
        DSId = "my-rtmps"
      messageId = "12B87B6D-9372-71E2-3D63-8C680CBEA8EE"
      timestamp = 0.0
      timeToLive = 0.0
      destination = ""
[LCDS]Received command: TCCommand [ Cmd: 2, MethodName: connect, TrxID: 1.0]
[LCDS]FlexSession created with id 'FECA09F0-F71A-F8ED-9E68-30B9D6609791' for a direct RTMP connection. Id value was server generated.
[LCDS]Returning service description for endpoint: my-rtmps config: {default-channels={channel={ref=my-rtmp}}, channels={channel=[{id=my-rtmps, type=mx.messaging.channels.SecureRTMPChannel, endpoint={uri=rtmps://{server.name}:2099}, properties={serialization={enable-small-messages=true}}}, {id=my-rtmp, type=mx.messaging.channels.RTMPChannel, endpoint={uri=rtmp://{server.name}:2039}, properties={serialization={enable-small-messages=true}}}]}}
[LCDS]Serializing AMF/RTMP response
Version: 0
(Command method=_result (0) trxId=1)
    (Object #0)
      id = "FECA09F0-F729-8037-075D-EDD727DDE50E"
      objectEncoding = 3.0
      level = "status"
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        default-channels = (Typed Object #2 'flex.messaging.config.ConfigMap')
          channel = (Typed Object #3 'flex.messaging.config.ConfigMap')
            ref = "my-rtmp"
        channels = (Typed Object #4 'flex.messaging.config.ConfigMap')
          channel = (Typed Object #5 'flex.messaging.io.ArrayCollection')
            source = (Array #6)
              [0] = (Typed Object #7 'flex.messaging.config.ConfigMap')
                id = "my-rtmps"
                type = "mx.messaging.channels.SecureRTMPChannel"
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                  uri = "rtmps://{server.name}:2099"
                properties = (Typed Object #9 'flex.messaging.config.ConfigMap')
                  serialization = (Typed Object #10 'flex.messaging.config.ConfigMap')
                    enable-small-messages = "true"
              [1] = (Typed Object #11 'flex.messaging.config.ConfigMap')
                id = "my-rtmp"
                type = "mx.messaging.channels.RTMPChannel"
                endpoint = (Typed Object #12 'flex.messaging.config.ConfigMap')
                  uri = "rtmp://{server.name}:2039"
                properties = (Typed Object #13 'flex.messaging.config.ConfigMap')
                  serialization = (Typed Object #14 'flex.messaging.config.ConfigMap')
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      details = null
      description = "Connection succeeded."
      DSMessagingVersion = 1.0
      code = "NetConnection.Connect.Success"
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[LCDS]Thread[my-rtmps-SocketServer-WorkerThread-2,5,main] registering write interest for Connection '26991461'.
The server side log did not show any exceptions, I am wondering is there any other settings that I need to pre-config in order to make my app run over the RTMPS channel?
Any help will be appreciated!

The problem is solved. The error message is due to the default channel was set to RTMP in my data-management-config.xml file. That's why modifing the application level default channel does not work. After I changed it to RTMPS, it worked.
One more thing to make sure though, if I want to use both of the RTMP channel and RTMPS channel in my app (not for fail over) , I have to create at least two models right? Also do I need to change the default channel in data-management-config.xml if the model I am about to deploy expects different channel than previous models?

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Hi,
I have a question regarding software design so that each class is independent of other.
public class A
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I am trying to figure out if this can be done using interfaces and abstract classes.
I am waiting for some suggestions and if you have completely different suggestion then I am open for it too.
thanks in advance,
@debug.

interface I {
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Luis,
There is a section (Security Issues to Consider) in the file upload/download how-to that describes how to handle this. Also, Anton Nielsen describes a technique linked to from Re: About file .
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Some very basic questions about calling a DLL in C[VI] code.

Hello everyone!
I'm sorry if this a really simple concept. I've come from a background of embedded systems, so DLLs are something I haven't really played around with much. Regardless. I'm developing an application in CVI for a project, and I generated a DLL that I want to use in Visual Studio 2013. I was able to compile it and it output the expected *.dll file. My question however is how do I properly import that *.dll into the CVI IDE? I did google and search the forums and I didn't find what I needed exactly.... I did find however this link: <http://www.ni.com/white-paper/8503/en/>. I did read through it all, and I think I'm going to try and use Implicit Linking. In the link it does detail 3 steps:
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I was able to easily complete step 1.. Now I'm on step 2. The headerfile that contains the function prototype of my code...I was looking at examples, do I just call the dll file as a .h file now? It seemed that way from what I saw. Do I need to output a header file for the DLL some how? Does the CVI compiler automatically reference that .h file with the lib that was imported?
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Proud to use LW/CVI from 3.1 on.
My contributions to the Developer Zone Community
If I have helped you, why not giving me a kudos?

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I faced a question in written exam.
A CALL statement inside a trigger allow us to call
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c)function
d)another trigger
Can anyone give me answer with reason?
I used CALL statement inside trigger but not allowing to use it. Might be earlier in oracle CALL statement we can use..its only a guess so I am asking in forum..
plz guide me..
rgds,
pc

You can use CALL in a trigger without resorting to EXECUTE IMMEDIATE
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Hello
All<o:p></o:p>
Can
someone please help me with the following question<o:p></o:p>
At the moment I have a couple of Windows 2003 R2 Servers with Microsoft
Certificate Services installed (e.g. domain joined enterprise CA infrastructure). <o:p></o:p>
The CDP extension of issued certs contains both and LDAP path and HTTP path to retrieve the CRL
the HTTP path being the standard <o:p></o:p>
URL=http://<CAServer FQDN>/CertEnroll/ENTRootCA.crl<o:p></o:p>
and LDAP being the usual location in AD<o:p></o:p>
Now I need to turn off the 2003 R2 CA in the near future and want to be sure the
clients requesting a CRL can still obtain the CRL.<o:p></o:p>
The CARoot Cert is already distributed to all the workstations (as is the SubCA cert) to the usual containers.<o:p></o:p>
The LDAP Path comes first in the list of CRL locations in the CDP extension followed by the URL location.<o:p></o:p>
I know (believe) in general a UA (user agent i.e. WEB Browser) with check the list of CRL locations in turn and as long as it can reach one of them will be OK.<o:p></o:p>
I am OK on the Windows side of things but we also have a UNIX/Linux environment (like most companies) that also utilize certificates issued by the MS CA’s therefore their respective UG will
(or should that be might) check the CRL via
the CDP.<o:p></o:p>
If I turn off the CA then the HTTP path in the CDP will no longer be available (LDAP will still be available).<o:p></o:p>
I just want to check if there are any UG out on the network using HTTP to retrieve the CRL, so I was thinking about checking one or more logs on the CA for HTTP traffic regarding CRL retrieve.<o:p></o:p>
Questions<o:p></o:p>
Is CRL retrieval from the CA (via the default HTTP path) logged in any of the CA
Logs?
If so which logs?
If not logged by default can I turn up logging (i.e. I see there is a Debug option) and if so will this then log HTTP CRL retrieval requests (e.g. clients IP address making the http request)?<o:p></o:p>
I just want to check this in case some UNIX/Linux UG are not trying LDAP first or cannot retrieve via LDAP and therefore have to reply on Http before I turn on the CA <o:p></o:p>
Thanks
all in advance<o:p></o:p>
AAnotherUser__<o:p></o:p>
AAnotherUser__

(same text, formatted better)
Hello All
Can someone please help me with the following question
At the moment I have a couple of Windows 2003 R2 Servers with Microsoft
Certificate Services installed (e.g. domain joined enterprise CA infrastructure).
The CDP extension of issued certs contains both and LDAP path and HTTP path to retrieve the CRL
the HTTP path being the standard
URL=http://<CAServer FQDN>/CertEnroll/ENTRootCA.crl
and LDAP being the usual location in AD
Now I need to turn off the 2003 R2 CA in the near future and want to be sure the
clients requesting a CRL can still obtain the CRL.
The CARoot Cert is already distributed to all the workstations (as is the SubCA cert) to the usual containers
The LDAP Path comes first in the list of CRL locations in the CDP extension followed by the URL location.
I know (believe) in general a UA (user agent i.e. WEB Browser) with check the list of CRL locations in turn and as long as it can reach one of them will be OK
I am OK on the Windows side of things but we also have a UNIX/Linux environment (like most companies) that also utilize certificates issued by the MS CA’s therefore their respective UG will
(or should that be might) check the CRL via
the CDP
If I turn off the CA then the HTTP path in the CDP will no longer be available (LDAP will still be available).
I just want to check if there are any UG out on the network using HTTP to retrieve the CRL, so I was thinking about checking one or more logs on the CA for HTTP traffic regarding CRL retrieve.
Questions
Is CRL retrieval from the CA (via the default HTTP path) logged in any of the CA
Logs?
If so which logs?
If not logged by default can I turn up logging (i.e. I see there is a Debug option) and if so will this then log HTTP CRL retrieval requests (e.g. clients IP address making the http request)?
I just want to check this in case some UNIX/Linux UG are not trying LDAP first or cannot retrieve via LDAP and therefore have to reply on Http before I turn on the CA
Thanks all in advance
AAnotherUser__
AAnotherUser__

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Any helps?
Thanks
Message was edited by:
King666

I also tested the webservice from PDB. (its WSDL url is http://www.rcsb.org/pdbws/rcsbWebService?wsdl). Only method getIDStatus works. and others such as getAtomSite do not work. And I got the same error message (invocationTargetException).
The insteresting thing is that I wrote a standalone java program and I could sucessfully call all web services method from there. It looks like something wrong with my JSC configuration.
Can anyone give me some help?
Thanks in advance,

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I tried applying the text in Photoshop and that works just fine until the file gets imported into ID, at which point the text looks terrible, having become somehow degraded in transit. I am fairly new to ID and PS, and would appreciate any helpful input.
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~Artemis

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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) .
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|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

2 Skype #'s, 1 Account: Question About Calling Out

Hi folks - I am needing a 2nd Skype # for business in a 2nd area code. Same biz, just a different location. From my research it seems that having both #'s under 1 account could work. I don't really need to know which # was called. However... I do need to be able to choose which # to call out on. One area code for one region and the other # for the other location. I use Skype on my iPad and Android phone. Will I be able to choose which # to make a call from on both of these devices? Thanks much.
Solved!
Go to Solution.

When two Skype Numbers are associated with the same account, it is possible to control which number will be used for outgoing Caller ID purposes
Toggling between these numbers is possible in your account settings, but it is not particularly convenient to do this on an ongoing basis, especially since the setting is not available in profile settings under Android
It is also possible set up Caller ID so that the number that corresponds to the country code of the number you are calling is displayed, but that won't work when the Skype Numbers are for different area codes in the same country
Another way to address this issue is to use 2 Skype accounts, and get a Skype Number for each Skype Name
It is possible to simultaneously run multiple instances of the Skype client under Windows and Linux (and I think OS X too), but, unfortunately, not on smartphones or tablets
Regards,
Neil

Question about Sharing folders over network

Alright, i have a mac and a PC, on the same network. I have a folder on my PC thats shared, and i can access it via the network tab in finder. The network icon thingy appears on the desktop, and it acts like a folder on my mac. Is there anyway i can move this "folder" to another location? Also, when i restart my mac, i have to conncet to the same folder again via the network tab. Is there anyway to have the folder stay there, even when restart my mac?

Create an alias or a symbolic link would be the normal manner so you can access a folder/directory from another location.
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