Changing Bearer Capability in ISDN gateway

Many time it had happend with me that i get a bearer capability mismatch between my gateway and service provider switch due to which the call didn/t mature
and during troubleshooting you run the command " debug isdn q931 " in the router and get a error similar to "Invalid information element contents and Incompatible destination"

This signifies that the bearer capability of the service provider switch and the voice gateway didn't match.

In order to resolve this kind of issue there is a option  in the IOS gateway that you can  overwrite the bearer capability  using the IOS command in the voice-port config mode...

The command for this is

config−voiceport)#bearer−cap ?

This will give you option to overwrite the capability set....

Abbreviated dialing

Abbreviated dialing is similiar to speed dial and can be invoked by pressing the " abbvr " softkey.

The abbreviated dialing can be configued from the ccm end user page and selecting the device and configure the speed dial.

for example lets the 6th abbdial number you configured in end user page is 12345678.

Now in phone press a new call , enter 6 and press the "abbvr" softkey .

once pressed it will change the number to the number (12345678) you configured in end user page ..

Sending Mwi in different extension other then voice mail

During my various implementation project i came across a common requirement which is having a common voice mail box to a group of users
and the requirement is that they get a visual mwi indication to there extension when a call came to the common voice mailbox.

How to acheive this ????


Now to start with lets say your common voice mail box number is 1000 and you want the indication on the extension 2001,2002  etc

for enabling this go the the common mail box user setting in unity and navigate to Edit > Message waiting indicator

Add a new message waiting indicator  and mention the extension 2001 and 2002 etc

This will acheive the message indication when a  message got left in the common voice mail box ..

Changing Phone background in Cisco IP Phone

This is a very common requirement to customize the background image of phone for various deployment and is very commond practice across organization for branding purpose.

Here i am going to discuss the steps that you need to change the background image  for cisco phone.

Below are the few point that you need to keep in mind to acheive the result:

1. Keep with you the List.xml file remember the name of the file is case senstive

The xml file should contain the below text
 <CiscoIPPhoneImageList>
<ImageItem Image="TFTP:Desktops/320x212x16/TN-Image.png"
URL="TFTP:Desktops/320x212x16/Image.png"/>
</CiscoIPPhoneImageList>

2. Keep with you the image file that you want to upload  ,Actual Image size should be 320 x 212 pixels and the thumbnail image size should be 80x53 pixels

e.g Name the original image as  Image.png and the thumbnail as TN-Image.png

Now upload all the file  files to the TFTP file management by navigating to Operating Systme Administration > TFTP file management and upload the file , mention the directory for the file which will be specific to phone model for e.g Cisco 7941-61 uses /Desktops/320x196x4 as a directory.

Now once all the file uploaded restart the tftp server.

Now you can change the phone background from the Phone setting ..

How to configure Cisco Webdialer

Cisco web dialer is a web based utility which can be used to dial any number using web interface that will generate a call from your deskphone
The configuration of web dialer is pretty straight forward and i am going to mention the points  as  below:

1. Verify that the Cisco Web dialer service is activated from the servicability page
2. Populate the CTI server, Timer value and the application dial rule setting from the service parameter as per below

3. Create  a end user and associate the phone to the end user, Assign the standard end user role to the user

 Done with the config...

Now to use we dialer browse to web dialer URL
https://<ip add of cucm>:8443/webdialer/Webdialer

Its will ask for authentication, Enter the useid and password for the user .

 
The next page will give you option to dial a number 

 

Enter the number and click dial , It will make your deskphone to dial the number you have entered...

you can use the webdialer to integrate  with any third party application ...

 

 

Cisco IPMA -Manager Assistant feature

Cisco Manager assistant feature with proxy line support is a service in CUCM that intercept the manager call and send it to the assistant, So that the assistant answer the call. The manager can control this behaviour from his phone and softkey and enable or disable the divert .

There are a few steps to acheive this configuration which i am going to discuss here one by one.

1. First  Create  a Partition for the Manager Phone name it as PT-Manager
2. Create a partition for all Phone name in PT_All
3. Create two CSS one for all Phone name it CSS_All  -- PT_All, CSS_IPMA -- PT_All + PT_Manager

4. Create a CTI Route Point and assign it a DN 10000X (Remember the DN should be a Full number or mask that matches with the manager DN in our case manager DN is 10001

The CSS of the CTI RP should be CSS_IPMA that has access  to the Manager Phone and Agent Phone

5. Create a IPMA service from the Phone Service options

6. Go to service parameter and choose IPMA and define the below mentioned paramater

 

7. Assign the manager Phone the Softkey template for manager and subscribe to IPMA services and in the Assistant Phone associate the Softkey template for assistant

8. Go to Enduser and choose the manager user and associate the Manager Phone and assign the CTI role to the end user

9. Go to the enduser page and associate the assistant phone to the assistant User  and assign the CTI role 

10.In the Manager user from the related link choose the manager configuration and define the mapping for asistant phone  with the Manager phone

many time after the configuration you may need to restart the below services for troubleshooting

1. Cisco TFTP server

2. Cisco Manager Assistant server

3. Cisco CTI server 

Now login to the assistnat console that you can download from the plugin page using the assistant credentail .

Counting Length of a String Variable in CCX editior

Many time while designing a ccx script i came across a common requirement that is checking the length of the number of character in a variable .
Here i am defining a method that can be used to count the number of character in a string.

For this define two variable , One is of type string and other is of type interger

Type integer variable (intExtnNo)
Type  string variable (strExtno)

The use the below statement to count the number of character of the string variable

intExtnNo= strExtno .length()

The function strExtno .length() will calculate the number of character and return a integer value.

Barge-in feature in CUCM

Barge in feature are used with shared line to be a part of a active call between parties.
Lets take an example.

Phone A and Phone B have one shared line , Phone C call Phone B and two are in a call the Phone A can barge into the active call and can listren to the converstaion.
Now based on the Party Entrance Tone parameter the parties alreday in call may hear a beep done when someone barge into  the call .

There are few basic parameter that need to be taken care of for configuring barge .

Turn Privacy to OFF (default is ON)

Set Built In Bridge to ON

Select Single Button Barge to either Barge or cBarge (default is OFF)

Select a Softkey Template. For Barge, a Standard User template is sufficient

 

The softkey template layout can be changed from Device Setting > Softphone layout

Add a shared-line DN, i.e   10005 for both test phones .

Now when some one is in conversation on the share line you can select the  share line to barge in

To disconnect from call just select the EndCall.

Done..........

Configuring callback feature in CUCM

Callback feature in CUCM as the name suggest is used to indicate a dialed number availability status on your IP Phone ..

Take a example Phone A call Phone B ,Suppose the call is not successful due to any reason (Busy ., no answer or use doesn't pick the call) and you want that when the user is available to take call next time will be notified to you you need to press the "Callback Softkey " on the phone and the phone will monitor the status of the Called phone Phone B .

The below are the steps to configure the Callback Feature

1. Create a Softkey template
2. Assign the Callback softkey to the template
3. Assign the Softkey template to the Phone

done..

Now to check

 From Phone A call Phone B
Press the Callback Softkey in Phone A

The below screen will appear, Press Ok to activate callback

Press Exit

Now if the Phone B will become onhook or in state of taking call below message will appear in the screen of Phone A with a Audio  beep ..

Cisco Unified attendant console searching using multiple field

In the attendant console  client version by default you can search a contact based only on one field.
However you can easily tweak the behaviour by changing one parameter

The Parameter can be found in Option >Preference. filter search.

Click/Check the checkbox before " I want to use AND searching "

This will enable search on multiple field

MGCP gateway Call preservation

Many times when i think about the behaviour of  H323 and MGCP one think that commonly came to my mind is the call preservation.
after doing some research in internet i tried to consolidate the behaviour of call preservation in case of MGCP.

MGCP actually do a PRI backhaul to CUCM to pass the   layer 3(q931) information to the CUCM from the gateway so What will happen if the MGCP gateway is registered to a Subscriber in a cluster and subscriber goes down.

The information that is received by the subsciber will be shared by the publisher , So if the subscriber goes down those information will be available with the Publisher.As the D channel terminate on the MGCP gateway so it will be UP

• CUCM send AUEP (Audit Endpoint) to the gateway to find the state of each B-channel
• CUCM send AUCX for any Endpoint for which the gateway reports a preserved call
• Send Q.931 Status Enquiry messages to the PRI device attached to the gateway to confirm the status of any calls CUCM believes are preserved.

hence the call in this case will be preserved

Now what will happen in case of SRST fall back ????

In case of SRST fallback  as it will reset the PRI backhaul hence the Layer 2 (Q921 ) will also reset and the call will drop, so no call preservation will work in case of SRST fallback ..

Configuring Queue based hunting in Cisco unified communication manager version ( CUCM ) 9

There is a new feature added on CUCM version 9 which provide call queueing while doing hunting  .
Few days back i tried simulating it in my Lab which i want to share in this blog.

To start with you first need to create a Line group i created a test  one as below :

Then add the DN to the Line group ..

Next create a Hunt list and include the Line group in that Hunt list .

Next is to create a Hunt Pilot ..

To create  a hunt pilot by navigating to Call Routing > Route/Hunt > Hunt Pilot

Then create a Hunt Pilot and navigate to bottom section of Hunt pilot   in the queue section mention the Audio file and queue settings.

Now whenever all the line group member are busy the caller will be put into the queue and will hear the queue music ..

UCCX Script for passing timezone as a parameter

Recently faced a requirement from one of our client to implement script for there different location which are in different timezone in in the same time to check the Working hour and weekdays.

After doing some research in the internet came across few valuable post but the one that i found will fulfill my requirement is from Johnathan in cisco support forum

In my script i have used the same one as a subflow for many of the subflow that  i have used in the main script.

That script will used predefined java classes   and fetch the data from the calender value in java

We defined a  a string value called currentTime as a string  and from the below code pass the value of the java script to currentTime and then change the current time to integer value as below .

Below script can be user to get the time and change the time zone based on the parameter of a string variable called timeZone :

You can use the set step to define the return value to currentTime (an integret variable defined in the script)

{
    //Initialize variables
    String strHourOfDay = "";
    String strMinute = "";
    String strSecond = "";
    //Set the time from local values
    java.util.Calendar callCalendar = new java.util.GregorianCalendar();
    int intHourOfDay = callCalendar.get(java.util.Calendar.HOUR_OF_DAY);
    int intMinute = callCalendar.get(java.util.Calendar.MINUTE);
    int intSecond = callCalendar.get(java.util.Calendar.SECOND);
    //Adjust to the timezone if specified
    if((timeZone != null) && (timeZone.length() >= 1))
    {
        callCalendar = new java.util.GregorianCalendar(java.util.TimeZone.getTimeZone( timeZone ));
        //Update the time for the timezone
        intHourOfDay = callCalendar.get(java.util.Calendar.HOUR_OF_DAY);
        intMinute = callCalendar.get(java.util.Calendar.MINUTE);
        intSecond = callCalendar.get(java.util.Calendar.SECOND);
    }
 
       if (intHourOfDay < 10)
                strHourOfDay  =  "0" + intHourOfDay;
    else
                strHourOfDay = "" + intHourOfDay;

    if (intMinute < 10)
                strMinute = "0" + intMinute;
    else
                strMinute = "" + intMinute;
      if (intSecond < 10)
                strSecond  =  "0" + intSecond;
    else
                strSecond = "" + intSecond;
    //STOP changing to string values

    //Return the time in a custom formatted form
    return strHourOfDay + strMinute + strSecond;
}

The current timeZone(string varaible) value  can be assigned from its preceding script using the subflow step and pass into this script.

and you can pass the time returned in the variable currentTimeint  to the preceeding call flow  and preeceding call flow can be can use this varaible to check the value with some predefined  integer variable passed as parameter in format hhmmss  and use an if statement to chenage boolean variable to change in true or flase based on working hour or not


I used it in my lab and  seesms to be good....

Error "java.lang.NullPointerException" while adding variable in CCX editor Installed in Windows 7

few days back i was getting a weird behaviour from the instance of the installation of CCX editor(version 9.x) installed in my window 7 machine.
Every time i try to add a variable it is giving me a java error "java.lang.NullPointerException " . After going some google found a cisco post which explain the issue with new version of windows with CCX editor 7.x .

Here is the error i was getting

Resolution to this is as folows"

1. Right click on the icon for CRS editior

2. Go to Compatibility and choose run as windows 2000

Now try to re-launch the CCX/CRS editor and the above error should not re appear again...

:-)

How to Add your Site in Bidvertiser

There are many option apart from Adsense to monetize your blog . These days the google Adsense has become very strict in there policy of approving Ad sense Request and your request may get rejected, But don't be disappointed with that  there are many other option .

Here i will discuss how to add Bidvertiser on your blog.

To start with........

Loging to  http://www.bidvertiser.com and start your registration process by clicking on Join Free

Create a User id and possword and follow the next step .

Now enter your site details and details for what is your site about

Once done go to next page of the registration wizard and click on " Get Ad Code"

This step will generate a html code for your . Copy that html code on a notepad

Go to your blogspot profilr and navigate to the Layout  , add a  html gadget and paste the generated html code and save..

With this your Blog is now added for Bidvertiser

Changing MAC address of UCCX 9

Disclaimer : This blog is just for educational purpose and doesn't involved in any unlicensed and crack software .

The recent license approach of cisco is based on License MAC which is generated based on few parameter you provide during the installation.
However  i am writing this blog for the LAB users only for educational purpose .Plz don't apply this on our production servers.

Before changing the license MAC  you need to boot the UCCX box in rescue mode.You can use any bootbable linux CD however i used linux 5.

when the system boot it gives you a linux rescue mode.

Type Linux rescue in the boot mode

once the server enter into the rescue mode .

type chroot /mnt/sysimage

After that open the file at location /usr/local/bin/base_scripts/LicenseMac.sh and edit this with any editor ( i used the VI editior)

Search for the field ..
FinalString=`expr substr "$SHA1sum" 1 12`

Replace it with

FinalString="Your License MAC Addres"

to save and quit once you have modified the field  press "Escape" the :wq!

:wq!  will save and quit  the vi editor.

Now next step is to modify the  "/etc/selinux/config"

Open the file with VI editor and modify the file as below;

Change the SELINUX=permissive ( by default it will be enforcing mode)

Save and exit the file 

Reboot your uccx box and check using the command show status your new license MAC , upload the permanent license you have with license MAC

Adding Your blog to Alexa

Adding site to alexa.com to monitor your site ranking .For doing this you need to have a login account in alexa.com which can be created easily .
Once the login is created login to the Alexa.com with yiur alexa userid and password.

 

Go to Dashboard and add your website link

Enter the website detail and click on claim your site .

Next use the free option to sign up

Use the second option and copy the  verification code.

Add this verification code in your Blog HTML editor and save.

Click on verify my ID

Done.. 

Transfer button in cisco Phone greyed out

Few days back encountered a issue on one cutomer location related to soft key, The transfer soft key in the cisco ip phone 7941 is inactive .
after doing a hour of reaearch came across a interesting fact that the maximum call in the Phone conmfiguration is set to 1 :-(

This is the root cause for transfer soft key to be inactive.

increasing the maximum call to a value 3  resolve the issue

Discover new phone in Informacast

Here are the steps that need to be done for discovering any new phone added .

1. Go to  Receipent > Edit receipent Group > Discover recepient

Click on Update

In the next screen click on update

 It will take sometime to discover based on the CUCM database .
Note: You need to make sure that all the server in the cluster have the SNMP string available and the web access on the phone is enabled
Also make sure authentication url is mentione in the phone setting

Creating Recepient Group in InformaCast

For creating a recepient group go to the informacast url https://<IP Address of InformaCast>:8444/InformaCast/admin

1. Go to Recipients >Edit Receipients Group

Option for adding device in recepient group are

Individual
Filter with recepient group
Filter wit rules
Exclusions

tag can be used for group a search so that control over the search results is good.  

You can select the selection criteria based on the above options

To send a message to group...

Go to  Message > Send and Edit Message
under the action tab select send

Click the option where you want to send the messages...For sending a message you can choose a recepeint group or a DN

Cisco Jabber configuration for iOS (i phone)

Here in this post i am giving the overview of the steps that we need to do for configuring Cisco Jabber on apple iphone.

Before doing anything we need to install the cop file for jabber

1. Download the cop file from the CCO

2. Install the cop file from the menu 

3. CUCM -> OS Administration -> Software Upgrades -> Install/Upgrade

once the upload is successful reboot the server 

4. Repeat the procedure for all the subscriber 

5. Repeat the Tomcat service so that the device icon can be displayed

6. Increase the SIP Dual Mode Alert Timer value from

CM Administration -> System > Service Parameters -> [Server] Cisco CallManager (Active) -> Increase the SIP Dual Mode Alert Time to

3000 milliseconds -> Save



7. create a dedicated SIP Profile and modify the below parameter in the SIP profile to 660

    Timer Register Expires
 * Timer Keep Alive Expires
 * Timer Subscribe Expires
8. Add the Phone:

CM Administration -> Device -> Cisco Dual Mode for iPhone -> Add New ->TCT<name>

9. Add the Line to the phone and Associate the Device with the End User



10.Launch the App on the iPhone .

 

 

H323 slow start and fast start .. With Cisco IOS gateway configuration

Before discussing on this topic we must be aware of how the basic call flow occurs on h323 protocol.
here is the basic call signalling that happens with h323

here the the h225 and h245 signalling uses two different TCP connection.

In case of h323 fast start the h245 steps which is the capability and media negotiation steps occurs in the setup message of h225 step.

In normal scenario as per the above diagram can be considered as slow start as h245 negotiation is occurring with separate TCP connection .

so with fast start few message transaction occurs compare to slow start.

Now lets focus on the configuration for the same in cisco IOS gateway.

we can configure this either globally or dial peer level.

In case of global level.

1. Go to configutaion mode

2. Go to voice service mode. with command " voice service voip"

3 Configure the command "h323 call start slow" to revert to fast start configure "h323 call start fast"


In case of individual dial peer level


1. Go to configutaion mode

2. Create a Voice class using command " voice class h323 tag "

3. under the voice class command configure 'call start slow"

4. use the voice class command under the the voip dial peer  command 



Done...:-)

SIP Early offer/early Media vs Delay Offer/Delay Media

Sip is a Request/Response protocol and it uses different mechanism to establish session.This article i am going to discuss about early/delay offer used by SIP for
capability negotiation

Early offer: SIP uses SDP (Session description protocol) in order to negotiate capability between two parties.In case of the early offer the SDP message is sent from the Calling party to calllee party in the initial invite message and the callee party respond with the negotiated capability  in the 200 OK it send to the calling
 party

Delay Offer : In this case the called party send the SDP message in the 200 OK message to the calling party and the calling party respond back with the negotiated parameter in the ACK message it send to callee party

Early Media : In case of early media the media between the callee and calling party establish before the session established between the two.In normal condition the SIP endpoint play ringing sound from its local source when it recieve a ringing message from the callee party.This is sometime confusing as no one knows what is happening at the callee party,Early media play a important role here what t does is the calle party send a message 183 instead of 180 and let the calling party establish a bearer channel between the two party so that they can share the message directly.

Delay Media : In case of delay media the media is established after the SIP session negotiation is complete

CAD Login Step in UCCX

Here is what all event occurs when a CAD uses login to UCCX .

1. CAD will connect to the UCCX (LRM service) to check if there are licenses available. If it fails at this stage, we get the error "License Server is Down". The login screen is not even provided.
2. Once the user enters the userid/password, the same is sent to the UCCX. Now, the user is searched within the LDAP database of the UCCX. If the user is not present, we get the "Invalid User" error.
3. The AXL request is sent to the CUCM for authentication. Post that, the CTI Control is opened on the agent device. Here we either get the "Authentication fail" or the "Error with CTI or RMJTAPI" respentively if either fails.
4. User configuration is loaded from the LDAP of UCCX i.e. agent specific config such as integrated browser setting etc. and also the connection is made to all the services such as recording, chat, playback and so on.

SIP Message transaction

In my previous post i mentioned the basic architecture of a SIP network and the component involve in SIP .
In this post i am going to mention several messages that the SIP uses to initiate and drop a call.

SIP works on request/response model

The most common messages used by SIP are as follows

Register: UA (User Agent) send a message about its location to SIP server in register message
Invite: This message is sent by the caller in order to other agent to join a SIP call , This message can also be used in order to change any call parameter for a established call.
ACK : The SIP UA receive several response for a invite , This message acknowledge the final response
CANCEL: This message end a call that has not yet been established
BYE:  This message end a session or decline to take a call
OPTION: This message queries about the capability of the server.

Response for a request in SIP fall under the below categories:

1XX : This reponse designate a provisional and informational message
2XX: This response designate a  repsonse to a request  that is successful .
3XX : This response is used to designate a redirect or location change for the called UA
4XX: This response is used to designate a error condition from UA side (Request failure)
5XX: This response is used to designate a failure condition from the Server Side
6XX: This response is used to designate a global failure

Informational response:

100 Trying
180 Ringing
181 Call Is Being Forwarded
182 Queued
183 Session Progress

Success

200 OK

Redirection
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service

Client-error

400 Bad Request
401 Unauthorized
402 Payment Required
403 Forbidden
404 Not Found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout
410 Gone
413 Request Entity Too Large
414 Requested URL Too Large
415 Unsupported Media Type
416 Unsupported URI1 Scheme
420 Bad Extension
421 Extension Required
423 Interval Too Brief
480 Temporarily Not Available
481 Call Leg or Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable

Server-error

500 Internal Server Error
501 Not Implemented
502 Bad Gateway
503 Service Unavailable
504 Server Timeout
505 SIP Version Not Supported
513 Message Too Large

Global failure

600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable

SIP Basics

SIP is a VOIP protocol and it stands for session initiated protocol , It uses text message same as http for commnication, The SIP network consist of below component.

SIP Proxy server:The proxy server works as an intermediate device that receives SIP requests from  a client and then forwards the requests on the behalf of the client. Proxy servers can provide functions such as authentication, authorization,network access control, routing, reliable request retransmission, and security. 

Redirect Server: The redirect server provides the client with information about the next hop or hops that a message should take, and the client then contacts the next hop server or user agent server directly.

Registrar Server: The registrar server processes requests from user agent clients for registration of their current location. Redirect or proxy servers often contain registrar servers

User Agent (UA) : UA comprises a combination of user agent client (UAC) and user agent server (UAS) that initiates and receives calls. A UAC initiates a SIP request. A UAS, a server application,contacts the user when it receives a SIP request.The UAS then responds on behalf of the user. CiscoUnified Communications Manager can act as both a server and a client

SIP user uses URI in a email id format for its identification,Unencrypted SIP works on port 5060 and encrypted(TLS) sip protocol works on Port 5061

SIP Packet exchange ... Coming Next...

How Gateway discover gatekeeper

We can use gatekeeper to register different gateway to a particular zone.Each zone will have only one active gatekeeper

Now how the gateway find s its gatekeeper ???

Gateway/Gatekeeper works on h323 protocol suite and h323 protocol suite is split into 3signalling

1. h225 (RAS-Registration,admission and Status)
2. h225 control signalling 
3. h245 (Media negotaiation and logical channel or transport signalling )

H225 RAS: RAS signalling work between gateway and gatekeeper on UDP port 1719 and for multicast work on UDP port (1718). In case of multicast it used the IP 224.0.1.41 to discover the gatekeeper

H225 control signalling:This is used  to setup connection between two H323 endpoints,This signalling works on TCP port 1720

H245 Media control and transport: this signalling is used to negotiate media capabilities and parameter fpor establishing logical channel,Adminssion control is also done by h245 signalling.

Gatekeeper discovery process:

1. The gateway or h323 endpoint send a GRQ(Gatekeeper request packet ) on port 224.0.1.41 of udp port 1718
2. the gatekeeper  either respond using a GCF (Gatekeepr_ confirm) or GRJ (Gatekeeper Reject ), In case of rejection it also include the reason of rejection .

 

Outbound dialer in uccx

In case of UCCX outbound dialing are of two type : Direct preview and IVR mode (IVR mode is further divided in progressive and predictive)

Direct preview:

In case of direct preview dialing

1. when a agent is in ready state and the dialer has selected a contact from the campaign manager the dialer ask the resource manager to reserve the resources

2. The resource manager change the agent to reserve state

3.  The dialer send a call to the Agent and same time initiate a popup in the agent desktop with the details of the contact, The agent either chooses to accept the call or reject the call if the Agent accept the call the call is connected , The dialer instruct the resource manager to place a outbound call from the Agent phone

4. Once the call is answered the the contact is closed by the dialer and marked as a voice call and send the details to the campain manager

IVR Mode:

In case of outbound dialer in IVR mode below  steps occurs;

1. The dialer chooses a contact  and as per the dialing mode (predictive or progressive based on algorithm ) dial the contact using the SIP via voice gateway

2. If the contact is not a live contcat which the SIP gateway can determine by using the CPA capability ,the dialer update the status accordingly in the configuration database

3. if the contact is a live contact then the dialer update the configuration database accordingly and send a SIP refer message to the SIP gateway which in turn transfer the call to the CTI route point associated with the IVR application which in turn choosea a IVR port and trigger the IVR application associated with the campaign .


The difference between these two is that in case of preview dialer the call is dialer from the Agent Phone while in IVR mode the acll is handled by a IVR application associated with the campaign.

Login issue in Cisco Agent Desktop

Recently during a deployment  project faced issue with Agent not able to login to CAD and giving error in the logs located at C:\Program Files (x86)\Cisco\Desktop\log ..

The error contain the message "Retrive value for LDAP Request Timeout from preferences"

So from the logs it seems that the CAD is timeout while initiating a login to UCCX .

Hence i increased the Timer for LDAP Authentication in the phonedev.cfg in
C:\Program Files\Cisco\Desktop\Config

and add the below line

[ReqTimeout]

Milliseconds=20000


it will increase the timeout value.

Next chnage in registry setting  in the below location
HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Calabrio\CAD\Site Setup

Change the LDAP Connection Timeout parameter value to 60

it resolved my login issue . Hope it help someone else.

ISDN CAS ( Channel associated Signalling )

CAS ( Channel associated sgnalling) is a kind of PSTN signalling for E1/T1 Link.

In case of CAS the signalling information is sent over the B Channel and hence the signalling is In Band.
Hence CAS is alos called a "Robbed bit signalling".

Below are the list of signalling used by CAS to detect phone and network events.
1. Loopstart
2. groundstart
3. E&M

The disadvantage of loopstart is that it is prone to glare (A situation where incoming and outgoing call both try to sieze a channel )

The best to use is E&M as it  provide both disconnect and answer supervision and is not prone to glare problem

Typical configuration of CAS include below command:

controller T1 0

clock source line primary

ds0-group 1 timeslots 1-24 type e&m-fgd dtmf dnis
In this case the below type command mean that it card will collect dual tone multifrequecy for DNIS for E&M
"type e&m-fgd dtmf dnis " .

You can find many such frequency or type you can select one as per your requirement....

Good Link from Cisco for T1 CAS troubleshoot..

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00801040bc.shtml

Showing Enterprise Data in Cisco Agent Desktop

Few days back i got a requirement from one of our customer.They want to deploy CCX in there setup and the requirement is when someone call a trigger point from PSTN and when the call reaches the the CAD, The CAD should display the name of the Customer for whom the trigger point has been configured .

So to acheive this we need to do few configuration in the Desktop administrator and the CCX script editior.
so first in the cisco desktop administrator.
Go to Service Configuration> Enterprise data > Field
Modify the Call Variable change the call variable name to anything meaningful (I changed it to Customer Name)

Next step is to define a layout i created and named it " Custom" and added i want to incorporate in the layout this layout will be displayed in the CAD window.

 Before that we need to get the called number that has been dialed,This you can get from "Get Call Contact Info" and assign it to a  string variable "strCallingNumber"

 

After that you can either use the caller number info to configure the customer name by using either the Switch or If statement
 Next step will be in the CCX script editor and add a " set enterprise variable ' step.

Remember before defining the "set enterprise variable" step you need to define the ECC varible from setting > Expanded call varible

Name this variable as "user.layout"

Now add a "Set Enterprise variable"