Call Manager 4.13 Multi-Level Access Question

I need to configure MLA and have read the CCO documentation. My question is - I'm not familiar with the ccmadministrator account and want to make sure that when I'm prompted to reset the password to this account immediately after enabling MLA - that I'm not going to muck up any underlying service that uses this account for something other than ccmadmin web access.

Hi Pklos,
I enabled the MLA and configured two functional groups and everything is working like a charm!
Thanks!
Amir

Similar Messages

Factory Default from Call Manager

I have a client which is private. They have many type phones that we need to change with new one. I want erase all information like call history, Itl files, tftp servers etc.
My first question is : What can I do to delete all these kind of information. My first idea is factory default? Can we make differently?
Second question is : If we can success that with anyone's answer , Can we send this command to all 7841 ip phones from Call Manager 10.5.1?
Second question is important part of work. We must do approximately 500 phones which I need to change with new phones. I want to send this command from Call Manager?
Regards..

Just go the phones erase call history/network config/ITL files etc the settings you need to delete. I dont think there will be a bulk way to do some of the items.
But If you must erase everything from the phone, you will need to reset the phone to factory defaults.
There is no easy/bulk/remote way to factory reset the phones. You will need to do it from the phones itself by pressing the sequence of keys.
Still only possible way would be to setup a DEMO cucm and connect the CUCM/phones to an isolated 24/48 port switch/switches. Factory reset them and let them download the new firmware files from the demo system.
They will still have TFTP server defined in the end (of the DEMO server) and without which you cant get the phone to download the firmware.
-Terry
Please rate all helpful posts

ESYU: R12 - Order Management를 위한 Multi Org Access Control(MOAC) setup 방법

Purpose
Oracle Order Management - Version: 12.0 to 12.0
Information in this document applies to any platform.
R12의 Order Management에 대핸 Multi Org Access Control(MOAC) setup 방법에 대해 알아본다.
Solution
일반적인 MOAC Setup:
1. HRMS에서 Security Profile을 정의:
a. HRMS Management responsibility 선택
b. HRMS Manager> Security> Profile로 이동
c. Security Profile이 정의되어 있는지 확인 (OM responsibility 혹은 Site level로)
d. 만일 아직 setup 되어져 있지 않다면 Operating Units를 입력
e. 저장
Note: 만일 위 d step과 같이 새로운 security profile을 생성하였다면 concurrent program 'Security List Maintenance'를 꼭 실행해야 한다.
그렇지 않으면 multiple operating units가 OM forms의 LOV에 나타나지 않을 것이다.
이 program은 multi-org access를 validating 하기 위해 사용하는 table에 data를 생성한다.
Navigation: HRMS Management> HRMS Manager> Processes & Reports> Submit Process & Report> Security List Maintenance
2. MO Profile Options setup:
a. MO: Security Profile - 이 profile setting은 MOAC functionality를 활성화 한다.
b. MO: Default Operating Unit - 이 Operating Unit는 OM forms과 report에서 default가 될 것이며, 이를 clear 하거나 변경하기 위해 LOV를 사용할 수 있다.
Keep the MO profiles in sync:
MO: Security Profile은 site와 responsibility level로 setting 할 수 있다.
MO: Default Operating Unit은 site, responsibility, user level로 setting 할 수 있다.
Application이 원하는대로 동작되지 않는것을 발견하면 이 profile options의 setting 값을 확인한다.
3. OM setup:
R12 upgrade 시 OM Profile에서 migrate 된 새로운 OM System Parameters를 확인:
Order Management Super User> Setup> System Parameters> Values
(See <<NOTE 393646.1>>-R12 Readiness Cheat Sheet: Migrated OM Profile Options)
4. Form에서 hidden field 'Operatin Unit'를 활성화시키고 default folder로 저장:
Sales Order and Order Organizer forms
Quick Sales order and Organizer forms
Sales Agreement forms
Pricing and Availability form
Other forms
Note: Sales Order form에서 hidden field 'Operating Unit'를 'Show' 하기 전에 fotm안에 이 field를 위한 공간을 만들어 놓아야 한다.
예를 들면 Customer Number field를 짧게 하거나 Operating Unit field로 이 field를 덮어씌울수 있다.
Reference
Note 393634.1

Hi Larry,
Have you considered adding the exec apps.mo_global.set_policy_context call to your connection's start-up script?
Tools -> Preferences -> Database -> Filename for connection startup scriptNot the most flexible approach, so I'm not sure if it is appropriate for your application, but just a thought. You might create distinct connection names with different start-up scripts for each org_id.
Regards,
Gary
SQL Developer Team

How to handle and manage a multi Database access in runtime with LCDS?

Hello there
I got several customer working with the same application and I wonder how, with LCDS, to manage in a runtime a multi dataBase access; without creating a configuration "mxl" file in
the folder catalina for each database.
Indeed, each customer have their own dataBase, and so far, I did not find out how to avoid creating a config xml file in catalina for every single database; which force me to create as well for each customer a folder application, since the name of the config file in catalina require a folder application to be ran under tomcat....
Thus, my question is :
Is there anyway to create only one configuration mxl file in catalina (in the server side) and then from the client side (application) let the user select its environment (meaning its database) to run the application.... this technic can be also used for multi database environment such as : Dev / Test / Prod environment (or database) where the same application can access to.
Please if any one have an idea or already delt with; just let me know, because I'm entering in a bootle neck and the situation is getting serioulsy critical....
Regards

Hello Ulrich,
with compact and repair I mean the MSAccess function "Compact and Repair".
Please follow the link below for more details:
http://office.microsoft.com/en-us/access-help/compact-and-repair-an-access-file-HP005187449.aspx
Normally you can execute this function directly in Access or with the Windows ODBC Data Sources Administrator => "Control Panel" => "Administrative Tools" => "Data Sources (ODBC)"...
I want to execute this function via cvi code and not by hand ;-).
Thank you for your support.
Frank

How to handle and Manage Multi DataBase access with LCDS in runtime ?

Hello there
I got several customer working with the same application and I wonder how, with LCDS, to manage in a runtime a multi dataBase access; without creating a configuration "mxl" file in
the folder catalina for each database.
Indeed, each customer have their own dataBase, and so far, I did not find out how to avoid creating a config xml file in catalina for every single database; which force me to create as well for each customer a folder application, since the name of the config file in catalina require a folder application to be ran under tomcat....
Thus, my question is :
Is there anyway to create only one configuration mxl file in catalina (in the server side) and then from the client side (application) let the user select its environment (meaning its database) to run the application.... this technic can be also used for multi database environment such as : Dev / Test / Prod environment (or database) where the same application can access to.
Please if any one have an idea or already delt with; just let me know, because I'm entering in a bootle neck and the situation is getting serioulsy critical....
Regards

Hello Ulrich,
with compact and repair I mean the MSAccess function "Compact and Repair".
Please follow the link below for more details:
http://office.microsoft.com/en-us/access-help/compact-and-repair-an-access-file-HP005187449.aspx
Normally you can execute this function directly in Access or with the Windows ODBC Data Sources Administrator => "Control Panel" => "Administrative Tools" => "Data Sources (ODBC)"...
I want to execute this function via cvi code and not by hand ;-).
Thank you for your support.
Frank

Call Manager 9.1 Active Directory Question(s)

Hello All!
Firstly let me establish that I am not an administrator of our VoIP system however I do manage the Server side of our network. We are in the process of planning an Active Directory upgrade and I'm having some difficulty getting a question answered about the requirements for Call Manager. We are at version 9.1 of Call Manager currently with our Active Directory version at 2003 R2. We are planning to upgrade to Active Directory version 2008 R2 (functional level) however we would like to use Server 2012 R2 as the OS for our AD servers. From a Microsoft standpoint this is a valid solution, it's built into Active Directory that you can run at different "functional levels" of AD on higher server operating systems. Any Call Manager applications that require a Windows operating system would run on whatever works for that (2003 or 2008 etc). Can we use Server 2012 R2 as the Domain Controller operating system while running at 2008 R2 functional level for Active Directory and still retain our Cisco support?

Hi Allen,
This is from Cisco site (you already may have seen this), though it talk about the directory services but it is specifically mentioned 2008. 2012 may work and specially as you are saying with functional level set to 2008 shouldn't have any issues. But Cisco have not tested that and you may get into support issues (if any).
Its completely tested and supported with CUCM 10.X
Version 9:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_010011.html#CUCM_TK_C4E65231_00
Configure LDAP directory
If you want to do so, you can add users from your corporate directory to the Cisco Unified Communications Manager database by synchronizing the user data to the database. Cisco Unified Communications Manager allows synchronization from the following directories to the database:
Microsoft Active Directory 2000
Microsoft Active Directory 2003
Microsoft Active Directory 2008
Microsoft Active Directory Application Mode 2003
Microsoft Lightweight Directory Services 2008
iPlanet Directory Server 5.1
Sun ONE Directory Server 5.2
Sun ONE Directory Server 6.x
OpenLDAP 2.3.39
OpenLDAP 2.4
Terry

About multi process access BDB question

I am designing one system using BDB.I have some questions about multi process access BDB.
1.If there are two process, they are shared BDB cache or every one has self BDB cache? (My understanding is every process has cache by itself.)
2.If one process write BDB and at the same time one process read BDB,how to make read processs can read data which write by write process just now?

1.If there are two process, they are shared BDB cache
or every one has self BDB cache? (My understanding
is every process has cache by itself.)You can configure it either way. The cache is maintained as part of the so-called environment. The usual thing is to configure a shared cache. You need to read about environments in the BDB and BDB/XML documentation.
http://www.oracle.com/technology/documentation/berkeley-db/xml/index.html
2.If one process write BDB and at the same time one
process read BDB,how to make read processs can read
data which write by write process just now?Use transactions, set up deadlock handling, and think about what transaction isolation guarantees you need. Note that there are four different fundamental setups for BDB: DS, CDS, TDS and HA, which stand for Data Store, Concurrent DS, Transactional DS and High Availability. For read-write concurrency, you need TDS.
http://www.oracle.com/technology/documentation/berkeley-db/xml/ref/intro/products.html
For high concurrency at the expense of memory, consider using multi-version concurrency control (MVCC). You can read about it in the following thread, especially in George Feinberg's replies:
Deadlock handling for beginners
In addition, there is ample documentation included in the BDB/XML distribution. It's pretty complex, as BDB can be configured in many different ways.
Michael Ludwig

Questions about TANDBERG MXP 550 with Call manager 4.2

Hello,
I have an TANDBERG MXP550 configured as an H.323 client in call manager 4.2 without a gatekeeper.
1. When I make a call between the TANDBERG and another phone registered. I have the ip address of the call manager displayed on the screen instead of the extension number. Is it possible to display the extension number without a gatekeeper ?
2. The other question is about the streaming. Can I get and read the multicast stream on a PC with a software client ? I tried with VLC (VideoLan Client), with the multicast address I don't get the stream. When I activate the Session Announcement Protocol (SAP) I can receive the stream, but the video is bad (probably a codec problem). Did somebody find a software client which reads the multicast stream?
Thanks,

Call manager provides the called party with the extension or directory number of the calling party on a display. You can use the Calling Line ID Presentation field in the Gateway Configuration window to control whether the CLID displays for all outgoing calls on the gateway.Refer URL
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008070e48b.html#wp1051056

No CLI Access to Call Manager MCS 7800

Hello, my client has a Call Manager server MCS 7800. We have access to the GUI, but the former IT Manager's network docu
mentation has an incorrect password for telnet access. How can we go about recovering or resetting this password?
thanks for any help in advance.

Bad news:
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucmbe/rel_notes/6_1_2/cucmbe-rel_note-612_2.html#wp339319
Cisco Unified Communications Manager Does Not Support Recovery of Administration or Security Passwords
Cisco Unified Communications Manager does not support recovery of administration or security passwords. If you lose these passwords, you must reset the passwords, as described in the Cisco Unified Communications Operating System Administration Guide.
The Cisco Unified Communications Operating System Administration Guide calls the section, "Recovering the Administrator or Security Passwords," instead of "Resetting the Administrator or Security Passwords." Access the "Recovering the Administrator or Security Passwords" section to reset the passwords.
Find your version here and follow the docs.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_documentation_roadmaps_list.html
For 7.1.2, that would be here:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/7_1_2/cucos/iptpch2.html#wp1044244
Hope this helps....

Cisco Call Manager Question

Hi,
I have a quick question on the Cisco Call Manger. Is it possible to have a Cisco call manager store a list of contacts, and then transfer all the list to any phone that registers to it? Furthermore, if this is possible, will the contacts be stored in the phones volatile or nonvolatile memory?
Thanks!
Billy

in call manager the directory list can be search by ip phones
the rource is from the end users page/settings
these details are either entered manually by system admin
or can be automatic when you integrate with LDAP like Microsoft AD
HTH

Multi-Site WAN With Centralized Call Manager

The customer has HQ with 15 Branches. Head quarter has about 4300 Phones, and Branches has:
Branch 1 = 420
Branch 2 = 256
Branch 3 = 385
Brnach 4 = 298
Branch 5 = 262
Branch 6 = 171
Branch 7 = 200
Branch 8 = 97
Branch 9 = 198
Branch 10 = 254
Branch 11 = 269
Branch 12 = 224
Branch 13 = 90
I would still like to propose Centralized Call Manager Cluster with SRST, but little confused since the number of phones per branch is very high.
What would be best deployment model for this type of scenerio along with VoiceMail and CER.

Of course you can use CME/CUE, but the problem is that you need 3845 for SRST with CME/CUE, which cost a lot of money CISCO3845-CCME/K9 is $16495 plus CUE ($3000, not include voice mail subscriber box).
So I will agree what people suggest here.
I have centralized design (Publisher, subscriber) at Main Location, and another subscriber at remote site coz 500 users. I chose put subscriber there rather than use 3845 with SRST.
They share voice box at Main site (Unified).
The rest of remote site use SRST for backup.
Large remote site with T1 PRI with SRST if WAN down.
Small remote site with vic2-2fxo/4fxo with SRST if WAN down.
You can read the SRND here:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_book09186a00806e8a79.html
Also, you can design multiple cluster depends the location of main and branches. For example, half office located at West Coast, another half at East Coast. If I were you, I will create two clusters.
Again, it depends a lot of things, for example bandwidth, round-trip delay etc...
Hopefully, thats can help you.
Ken

Cisco Call Manager 9 ELM question

Hello everyone!
We start using ELM for Call Manager Licensing and I just can not understand they way it works. I am used to DLUs and can not undertand the new scheme
Just to make is clear:
We have
Enhanced (9.x) - Unified CM Available - 180 Installed
Does it mean we can register 180 IP phones?
And we also have
Used - 8, Required
What is meant by Required?
Thank you!

It means you are entitled to 180Enhanced UCSL users (users with up to 2 devices), you have 18 such users currently provisioned and 132 still available. This does not mean you can have 180 such devices, it means you can have 180 users with 2 devices such as phone or softphone assigned to them or any 79XX, 89XX, 99XX phones.
The 21 and 9 are basic (69XX phone) and essential (3905, 6901 phone), since you don't have any of these licenses, they are being borrowed from the upper tier.
HTH,
Chris

Cisco Call Manager and MGCP Question

Hello,
I appreciate if somebody can help.
Scenario:
Site 1 PSTN E1----VG----Call Manager----VG--- PABX---Site 2 PSTN1 E1
I have configured a dialing pattern on Cisco call manager 6.xxxxxx to Send to VG on Site1
Both VG routers are using MGCP with call Manager.
The problem if from Site 2 tries to call 6xxxxxx the call manager is not routing the call to the VG in site 1.
I did debug ccapi inout and on Site 2 VG the call response was the number unassigned. This means that the call Manager is searching the directory for the destination but it is not searching the route patterns.
Any ideas to override this and ask the call manager to check it's destination pattern?
Thanks,

Problem solved. The VG in Site 2 was in a CSS that is not allowed to dial PSTN.
Regards,

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

Can Oracle Access Manager (OAM) be used for configuring data-level access?

Hi,
Can OAM 10gR3 or 11g be used for configuring data-level access in PS 9.10 and OBI 11g.
This would mean that data-level configuration are stored centrally within OAM database and PS 9.10 pulls these settings for its appropriate Row and ChartField security and OBI 11g queries OAM database tables to populate the initialization blocks during login.
If you have configured this successfuly, please feel free to provide highlevel details/steps about the configuration and any complexities that can pre present.
Thanks.

I recently worked on an interesting implementation that utilized OAM configured against OID in which OAM was configured to control authentication/authorization and then all applications that were serviced by OAM utilized the users login to pass to the database at which point the Oracle database utilized VPD, which was configured against the same OID instance, to leverage user attributes/groups in order to configure fine-grained access control to data within the database. Not sure if this would work for you unless your requirements are similar but it worked very well for this particular implementation as it allowed them to have SSO for their own apps and for BI Publisher which then could pass allong the userid to be leveraged by the database to do fine grained access control.
Edited by: oblix_fan on Jun 23, 2011 8:10 AM

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