CRS call flow sample requires

Dear all,
I want to take some CRS 3.53 sample call flow. depend on the CSQ, I want to take report which is about the Skill level. But I cannot find this report in Historical report. So I want to find some call flow which can take this report.
Regards
Lawrence Ng

Try this link
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801bf503.shtml

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UCCE Call flow to load balance & redundancy

Dear NetPro gurus,
I have used Cisco UCCX for a number of years but I'm a newbie to UCCE.
1. For UCCE, can each site have multiple PGs, Routers, Loggers & ICMs (for instance, can i have 2 PGs on site A & 2 PGs on site B (a total of 4 x PGs)? Or each site can only have 1 PG, 1 Router / 1 Logger and 1 ICM?
2. Is there a way where i can say force all my Customer Services CSQ to go via Site A? And all my IT Helpdesk CSQ to go via Site B? But if either site failover, the traffic will automatically fialover to the other sites PG, Loggers / Routers, and ICMs?
Would greatly appreciated if anyone can shed some lights on this.
Cheers,
Hunt

Hi Lee,
answer to your first Q:
UCCE will be having one router, one logger in each sites for one UCCE instance.
with respect to PG, you can have 2 PG on each side. It will work as active stand by mode.
for second question, you can do this based on selecting routing client from different site. Please provide call flow and what will be the VRU (CVP or IPIVR)
hope above will give you some light on your query.
Regards,
Shalid K.C

Cvp call flow trace

i am trying to disable call flow trace on CVP call server 8.0.1. when i use the commande setcalltrace off on voice browser administration i got a message saying :
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Packet Trace or Call Flow for IP Softphone

I have been searching around for a while not finding any packet trace or call flow document on the IP softphone. Is there anything like this available ?
Regards,
Mat

For Tracing the calls on IP softphone you can refer the following document. This document has the following section
Tracing the Cisco IP SoftPhone GUI Application - which explains how to setup
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/softphon/admin/ad_trbl.htm#1037447

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can we configure Call flow and DNIS is siebel ? also can we upload Prompt in Siebel CTI?

You can but not IVR stuff.

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What are the things that will happen when IP phone-A calls IPphone-B?

Hi Jaime,
In another discussion, you have replied that the call flow information can be obtained from this link:
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaaph.html
But I am not able to acces it. Please explain it or mail the document to me: [email protected]
Thanks in advance
Fayiz

BCM call flow architecture with detailed description of all the components

Hi Experts,
Please provide me BCM call flow architecture and detailed description about all the component Eg. When a customer calls on xxxx number call hits to IVR and than as per customer DTMF routed to available agent. In that call what are the components that participate and what is there roles. Is there any architecture diagram that can explain all the functionality?

Hello Raman,
Start with SAP BCM 7.0 SP6 Basic Installation Example document, on Service Marketplace. That document shows voice routing paths and explains the main functions for every component. The component functions are also touched in Master Guide and Security Guide, also on Service Marketplace.
Kind Regards,
Tomi

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Hello,
I'm considering purchasing MediaSense. The goal is 100% call recording. Our environment is CUCM 10. Publisher, sub, and gateway are all located at HQ. Remote sites (which include 7900 series phones) are located across a WAN.
My question is about the call flow if we add MediaSense. According to the MediaSense SRND v10, it appears to launch a new RTP stream from the phone to the MediaSense server. That would appear to double our bandwidth to remote sites (one stream from gateway to phone, another stream from phone to mediasense).
Is that correct? Any other deployment options that wouldn't have that impact?
Thanks!

The last time I checked for internal recording that was the only way, and they only supported phones that can utilize built in bridge.
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Call failed - reason required

In the attached log file, a call comes in, which shows up reports as "Aborted". I would like to know what was the reason this call failed.
Call flow:
PSTN --- PRI -- Voice Gateway (2900 series) -- SIP -- CUCM9 ---- UCCX9
On UCCX, it should go to a script which determines that it's within business hours, and redirects the call to the hunt pilot with phone number +33173150110
Details:
Called nr: +33173150194
Calling nr: +33628329975
Timestamp: Date: Mon, 16 Jun 2014 07:03:33 GMT
It starts on line 4136.
I can see that the call gets reINVITED a couple of times, and it seems to me that after the last INVITE, there's no media path, so I understand why the voice gateway would send a BYE. However, I don't understand why there's no media path.
Thanks

Hi Tom,
From the logs I can see that the call goes to UCCX using the CTI port and gets transferred back to a Hunt List, with DN 33173150110 for the pilot.
The Hunt List gets exhausted.
08729403.006 |09:03:37.554 |AppInfo |UNKNOWN_ALARM:HuntListExhausted - HuntListName:Card Team France App ID:Cisco CallManager Cluster ID:StandAloneCluster Node ID:UC-CUCM-CERGY
08729403.009 |09:03:37.555 |AppInfo |HuntListCdrc::terminateCall - Sending CcRejInd, with cause code (17), to Cc because it has not sent CcRegisterPartyB to Cc.
The call then connects to another entity anyway, after exhaustion of the hunt list. IP address of that entity is below. This should give you an idea of where the caller is connected.
08729567.001 |09:03:38.467 |AppInfo |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.25.56.130:[5060]:
[358209,NET]
ACK sip:[email protected]:5060 SIP/2.0
c=IN IP4 172.25.56.130
m=audio 24558 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
Caller then hangs up the call. This bye comes in from gateway with normal disconnect cause 16.
08729858.003 |09:03:52.094 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 600 from 172.25.56.130:[56693]:
[358217,NET]
BYE sip:[email protected]:5060 SIP/2.0

Can anyone pls give step by step call flow of ICM pre-routing

dear all,can any of you give the ICM pre-route call flow step by step?

dear all,can any of you give the ICM pre-route call flow step by step?

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Hello everyone,
I upgrade CCM 3.3(5) to CCM4.1(3) without any errors. Yet the follow call flow seems not working.
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Outside callers call to hotline#
- Carrier translates it to local#
- CallManager translates it to 1500 and sends it to Unity
- Unity plays greeting of hotline call handler and transfers call to extension #1001
- If 1001 is busy or not answer, call is forwarded to 1002.
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Howard

Hi Howard,
Hope things are well for you:) It seems like this should work regardless of the Unity version. What number/DN do you have setup up for the Voicemail box for the Hotline? It can't be 1500 because that is the Call Handler correct. You might want to do a test setup and make the last forward step a phone like a 7940 so that you can see what number is being presented to Unity in the final forward step (you may be able to use Unity Call Viewer, not sure about your older version).
It seems likely the problem is in the number being presented vs the number you have configured for the Hotline Mailbox.
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The start samples are really good and useful.
However I get an error when connecting to the API App on Azure. The code generated when "adding the reference" has a file called Values.cs. I get an error on line 219:
resultModel = StringCollection.DeserializeJson(responseDoc);
The correct line should be
resultModel = MyContactsListConsole.Models.
StringCollection.DeserializeJson(responseDoc);
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//Mikael Sand (MCTS, ICC 2011) -
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Sorry you had to discover this bug, Michael. It is a known issue we outlined in the release notes, and have since repaired it in the upcoming release. This is only an issue when your API returns an array of strings, as is the case for the default ValuesController
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hi,
What is the configuration on CVP that forwards call to ICM.
as i am not able to find any configuration in OAMP or CVP that target to ICM.
the flow is as follows-
VG-CUSP-CVP-VRUPG-ICM-CCMPG-CUCM
Thanks and Regards-
PK

Hi,
There is no much configuration in CVP.
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and in PIM configuration you should mention the CVP host name.
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AQxmlCallback is not called. demo sample did not work for me

I try to follow sample in
$ORACLE_HOME/rmdbs/demo
Here is steps:
1>create Database user by running SQL>@aqxmlusr.sql
2>create queue scripte by SQL>@aqxmldmo.sql
3>Startting Jserv and successfully getting /servlet/AQDemoServlet
It display on my Browser:
Sample AQ Servlet
AQxmlSevlet is working! 09/13/2002 03:26:21
4>try to run
C:\bin>java AQHttpRq <my-machine> 8888 POST http /aqdemo/AQDemoServlet john welcome aqxml01.xml
I got the following
server: <my-machine>
port : 8888
method POST
protocol http
URL /aqdemo/AQDemoServlet
username john
password welcome
xmlfiles aqxml01.xml
URL: http://<my-machine>:8888/aqdemo/AQDemoServlet
Setting User:: john:welcome
Setting Encoded:: null
POSTing file: aqxml01.xml
Cookie am9objp3ZWxjb21l
HTTP response code is 200
HTTP Content Length: -1
HTTP Content Type: text/plain; charset=iso-8859-1
HTTP Content:
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Se
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Se
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Ht
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Ht
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].util.ThreadPoo
 </PRE></BODY></HTML>read.java:62)
5> Then I try to create a callback like:
public class TestCallback implements oracle.AQ.xml.AQxmlCallback
/** Callback invoked before any AQ operations are performed by the servlet */
public void beforeAQOperation(HttpServletRequest request,HttpServletResponse response,AQxmlCallbackContext ctx)
System.out.println("Entering BeforeAQ Callback ...");
/** Callback invoked after any AQ operations are performed by the servlet */
public void afterAQOperation(HttpServletRequest request, HttpServletResponse
response,
AQxmlCallbackContext ctx)
System.out.println("Entering afterAQ Callback ...");
And modify
AQDemoServlet.java as putting the following code in init()
AQxmlCallback serv_cbk = new TestCallback();
setUserCallback(serv_cbk);
After restarting my Jserv,
I running
C:\bin>java AQHttpRq <my-machine> 8888 POST http /aqdemo/AQDemoServlet john welcome aqxml02.xml
a> I don't see callback is called
b> I still have the same error message:
server: <my-machine>
port : 8888
method POST
protocol http
URL /aqdemo/AQDemoServlet
username john
password welcome
xmlfiles aqxml01.xml
URL: http://<my-machine>:8888/aqdemo/AQDemoServlet
Setting User:: john:welcome
Setting Encoded:: null
POSTing file: aqxml01.xml
Cookie am9objp3ZWxjb21l
HTTP response code is 200
HTTP Content Length: -1
HTTP Content Type: text/plain; charset=iso-8859-1
HTTP Content:
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Se
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Se
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Ht
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.Ht
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].util.ThreadPoo
 </PRE></BODY></HTML>read.java:62)
Can someone help me out?
Thanks

on log file, it says
9/17/02 2:17 PM AQ-demo: AQDemoServlet: init
9/17/02 2:17 PM AQ-demo: Servlet error
java.lang.NullPointerException
 at oracle.AQ.xml.AQxmlServlet20.doPost(AQxmlServlet20.java)
 at javax.servlet.http.HttpServlet.service(HttpServlet.java:211)
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 at javax.servlet.http.HttpServlet.service(HttpServlet.java:336)
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.ServletRequestDispatcher.invoke(ServletRequestDispatcher.java:633)
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.ServletRequestDispatcher.forwardInternal(ServletRequestDispatcher.java:235)
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.HttpRequestHandler.processRequest(HttpRequestHandler.java:695)
 at com.evermind[Oracle9iAS (2.0.0.0) Containers for J2EE].server.http.HttpRequestHandler.run(HttpRequestHandler.java:248)
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CME Call Pickup Sample

I need to create a new call pickup group and need assistance.
Does anyone have a sample config for setting up a call pickup group?
Thanks

Yes we have 15 devices and we are ruinning 6.1 software and we can only get 9 devices to register at any one time!
I've also tried to get the Multicasting part working using an ephone-dn, with the feed ip cmd as follows;
ephone-dn 310
name paging zone 1
feed ip 239.1.1.100 port 8560
This does not work, I have multicast working on the network so far with paging-dn's and MOH working fine!
Lee
PS. Please let us know how your TAC case works out!!

CRS call flow sample requires

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