How To

How to Create Menu in Blog :

It is logical to put all you article of a common topic under a common heading ,This can be accompalished in blogger using Label.

To enable a label Create a new Page .

To create a new page browse to Blog dashboard and click on Page and create a new page

If label contain space between the word the use %20 for example in this case "How To do" contain space  hence in url mention "How&To&do"

Next Step is to go to
 Layout > Add  Gadget, Select   "Pages" from the available list of gadget and add the Page.

Now Create a new post and label it with "How To Do"  as shown in the screenshot

Done,....

Share Mailbox in Unity Connection

Recently came acoss a requirement from one of the customer they want a common mailbox from a group of user and also want to monitor the Mwi if someone left a message  on the mailbox.

There can be many way to acheive this but i opted to create a common mailbox and create alternate extension for the mailbox same as extension number of the other users in the group

You can find the option from user>edit>Alternate Extenstion.

The second option can be as follows:

Create a call handler for the common mailbox with the same extension number as the Call handler DN
Next go to Edit>Greeting setting in the call handler and set the after greeting option to go to Mailbox.

Choose the common mailbox user from the list ..Done....

In the CUCM assign the shared Extension for the mailbox t all the user in the group so that they can get a indication of new messages...

Dial Peer and Rule --Playing with Digit

To see about the digit translation:

Dial peer is a topic where i always try to be more focussed because it has many variation in terms of pattern and match Translation profile,Translation rule, Destination pattern and VOIP or Pots Dial peer

Lets discuss it one by one and start with translation rule:

Voice translatio Rule: Voice translation rule  is used to define a pattern to match and relace it with another pattern:
below example will provide more clarity

voice translation−rule 1

rule 1 /^123/ /456/

so it will match all pattern that begin with 123  and replace it with 456,if the number is 123654 it will be replaced with 456654

You can verify each rule using the test commanfd as shown below:

router#

test voice translation−rule 1 123

Matched with rule 1

Original number: 123 Translated number: 456



Once you have created a translation rule next step is to create a translation profile and bid the transllation rule to that profile

The rule can be defined under the profile for calling and called number

For example:

voice translation−profile <name>

translate called <translation−rule num>

translate calling <translation−rule num>

translate redirect−called <translation−rule num>

no

The key pattern of the game are
debug voice translation will be a useful command to see realtime digit manipulation

debug voice translation-rule
debug voice translation-rule


To be Continued............!

https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

https://supportforums.cisco.com/t5/ip-telephony/inbound-dial-peer-matching-question/td-p/2321861

Unified messaging vs Integrated messaging

I  have been searching a lot about what is the differentiator between Unified and Integrated messaging in Cisco Unified communication.But after doing a lot of search and reading articles able to findout the boundary line between these two.

The deployment of messaging can be broadly classified as follows:

1. Normal voice mail
2. Integrated messaging
3. Unified messaging

Normal Voice mail is the most simple one where user can retreive the voice mail from TUI interface.
Howwever the more interesting is the difference between Intergrated messaging and unified messaging

Integrated messaging uses separate database to store the voicemail of the user which is kept separted from email storage,The voice mail can be seen on outlook in new folder inside oulook apart from regular mail folder,This folder is mapped to unity connection using IMAP protocol.So when a caller leave a voicemail the mail is stored in the local voicemail server which can be a unity connection.MWI syncronization works fine in integrated messaging

In case of unified messaging the voicemail is stored in same storage as email which can be a exchange server.So when a caller left a message the message is stored in a single storage.In unified messaging the unity connection also keep a local copy of the voice mail.MWI is syncronized between the two

Also in unity connection you can opt for relay the messages if anyhow you have browsed the message setting you might have encountered the below two field "Relay" or "Accept+ Relay".

Now in case of Relay : When a caller leave a voice mail to the  voice mail is relayed to the external server mentioned ,unity connection will not store a local copy of the message

Relay + Accept :In this case the unity connection apart from relay also store a local copy of the messages..


To be Continued...........

Cofiguring Cisco 7915 extention module in CME

Confiuring a extention Module in CME is pretty straignt fowards:

First of all you need to check the compatibility of the Phone and the Extension Module and then Also check the firmware compatibility

Second step is to copy the Firmware file to flash  the below command can be used

Copy tftp flash:

Next define the tftp-server command in global config mode:

tftp-server flash:B015-1-0-2.SBN

Next define the Load command under telephony-services

load 7915-12 B015-1-0-2

* Remember to exlude the .sbn extention.

Next go to the ephone and type the below command

ephone 20
type 7962 addon 1 7915-12
Configure the additional Line

* i spent around 5 hour to make the Extention module register to the CME only because i haven't created the cnf file again so make a note that sometime you may need to create a cnf file again in telephony-service

Cisco Visual Voicemail

Cisco Visual Voice Mail :

Recently got an oppurunity with one of the customer to work on Cisco Visual Voice Mail.Its a very good feature as it will display the voice mail that is present in the user mail box.

Also there are many good document available in the internet will discuss what is Visual Voice mail and how to configure it.

Here i just want to share my experience with the community about this.

These are the below steps to configure visual voice mail

First step in this is to configure a Phone Service , In the phone service you need to define the below url in the service url field



http://<voicemail server info>/midlets/VisualVoicemail/VisualVoicemail.jad

Save the phone Service

Second Step is to Create a Visual Voice Mail Pilot  and assign relevant CSS.



 Now add a Route pattern (Same as Voice Mail Pilot) to send the request to Unity Connection using SIP trunk (Hunt list in case if SCCP Integration)





Add a  new Voice mail pilot ,remember this number will be different from that of Visual Voice Mail Pilot.



This all from CUCM side

Now the next you need to login to Unity Connection,

In unity Connection go to Advance > Connection administration and enyter the belwo value

Visual Voice mail web service : Visual Voice mail pilot
Voice mail web service: Voice mail pilot

Now You need to create a Direct Routing Rule form call routing>  Routing Rule

Configuring Unity Connection in a HA cluster

Yesterday got an oppurtunity to work on implemeting Unity connection in HA using SIP.
The steps are staright forward :

In CUCM

1. Create a two SIP trunk to Unity Connection
2. Create a Route group
3. Select the two SIP trunk in the route group
4. Create a Route List and add the route group to Route list
5. Create a specific Route pattern  for voice mail will be same as voice mil  pilot and select the voice mail route list in the route pattern
6. Create a voice mail pilot same as the route pattern
7. Create a voice mail profile

Unity Connection config

Note : We can only  create cluster of two server for unity connection and each server license will be based on its own mac address( Not same as CUCM licenseing )

1. Create Phone System
2. Create a Port group and select SIP as a protocol
3. Add a port group, remember to add different port group to different server
4.  Add the CUCM server IP address in the SIP IP Address field which you can get from the Edit tab


Now check the integration by dialing the voice mail number..It should work.. :-) 

Cisco Unity Connection

Cisco Unity Connection deployment in HA mode.

There are few basic task that need to be done in the unity connection to create a cluster  in HA.These are the high level steps that needs to  be done.

Add a Unity Connection Subscriber to the Unity Connection Subscriber from the Cluster Config

Install the unity connection  and follow the wizard for secondary node and enter the primary server details

Create separate SIP trunk in CUCM for the Second Unity Connection and add it to existing route group for Voicemail

Choose the required Trunk selection algorithm in Route Group

Enable Access Alert Logg

Next step is to understand the Licensing ...

On the Licensing part we need to assign license port separately to each server Do the License port allocation as per the design,The option can be found from Telephony Integrations >  Port of Cisco Unity Connection Administration.

Below is the list of setting available :

Field	# Considerations
Enabled	Check this check box to enable the port. The port is enabled during  normal operation. Uncheck this check box to disable the port. When the port is disabled,  calls to the port get a ringing tone but are not answered. Typically,  the port is disabled only by the installer during testing.
Server Name	(For Cisco Unity Connection clusters only) Click the name of the Cisco Unity Connection server that you want to  handle this port. For details, see Chapter 7,  "Configuring Voice Messaging Ports for a Cisco Unity Connection  Cluster."
Extension	Enter the extension for the port as assigned on the phone system.
Answer Calls	Check this check box to designate the port for answering calls. These  calls can be incoming calls from unidentified callers or from users.
Perform Message Notification	Check this check box to designate the port for notifying users of  messages. Assign Perform Message Notification to the least busy ports.
Send MWI Requests	Check this check box to designate the port for turning MWIs on and off.  Assign Send MWI Requests to the least busy ports.
Allow TRAP Connections	Check this check box so that users can use the phone as a recording and  playback device in Cisco Unity Connection web applications. Assign Allow  TRAP Connections to the least busy ports.
Outgoing Hunt Order	Enter the priority order in which Cisco Unity Connection will use the  ports when dialing out (for example, if the Perform Message  Notification, Send MWI Requests, or Allow TRAP Connections check box is  checked). The highest numbers are used first. However, when multiple  ports have the same Outgoing Hunt Order number, Cisco Unity Connection  will use the port that has been idle the longest.
Security Mode	Click the applicable security mode: • Non-secure—The  integrity and privacy of call-signaling messages will not be ensured  because call-signaling messages will be sent as clear (unencrypted) text  and will be connected to Cisco Unified CM through a non-authenticated  port rather than an authenticated TLS port. In addition, the media  stream will not be encrypted. • Authenticated—The  integrity of call-signaling messages will be ensured because they will  be connected to Cisco Unified CM through an authenticated TLS port.  However, the privacy of call-signaling messages will not be ensured  because they will be sent as clear (unencrypted) text. In addition, the  media stream will not be encrypted. • Encrypted—The  integrity and privacy of call-signaling messages will be ensured on this  port because they will be connected to Cisco Unified CM through an  authenticated TLS port, and the call-signaling messages will be  encrypted. In addition, the media stream will be encrypted.

Regarding the licensing part you need to have license port divided between the two unity connection servers


 What will be the behaviour when both Cisco Unity Connection Servers Are Functioning Normally
•The phone system is provisioned with  twice the number SCCP voice mail port devices needed to handle the voice  messaging traffic.

•A hunt group is configured to send  incoming calls first to the subscriber server, then to the publisher  server if no answering ports are available on the subscriber server.

•Both Cisco Unity Connection servers are  active and handle voice messaging traffic for the system.

•In Cisco Unity Connection  Administration, the voice messaging ports are assigned in the following  manner:

–The subscriber server answers most  incoming calls for the system.

–The publisher server handles most  dial-out calls (MWI requests and notifications).

This guide directs you to assign the voice messaging ports to their  specific Cisco Unity Connection server at the applicable time.

•The voice messaging ports on both  Cisco Unity Connection servers are registered with the phone system.

•The number of voice messaging ports  that are assigned to one Cisco Unity Connection server must be  sufficient to handle all of the voice messaging traffic for the system  (answering calls and dialing out) when the other Cisco Unity Connection  server stops functioning.

If both Cisco Unity Connection servers must be functioning to handle the  voice messaging traffic, the system will not have sufficient capacity  when one of the servers stops functioning.

•Each Cisco Unity Connection server is  assigned half the total number of voice messaging ports.

If all the voice messaging ports are assigned to one Cisco Unity  Connection server, the other Cisco Unity Connection server will not be  able to answer calls or to dial out.

•Each Cisco Unity Connection server must  be assigned voice messaging ports that will answer calls and that can  dial out (for example, to set MWIs).

Configuring fax in IOS

Configuring Fax for MGCP.

Below is the procedure to configure a FXS card in CME

config terminal
  ccm-manager fax protocol cisco (Select the fax protocol for MGCP in out case it is Cisco)

mgcp fax t38 ecm ( fax Configure MGCP Fax Parameters, t38 Configure MGCP Fax T.38 
Parameters, ecm Enable Error Correction Mode (ECM) )
Configure the Voice Port config (In our case the for fxs port is used which is  voice port 0/0/0)

voice-port 0/0/0 (Port where FAX Machine is connected) description FAXMachine-01
caller-id enable
 
Create a dial peer to send the call to Fax Machine

 dial-peer voice 10 pots destination-pattern 6055 (PSTN Number’s last 4 digit) port 1/0/0

Let me put some more fact on the type of Fax protocol as initially i found difficulty in remembering these ,Hence decided to record this in form of blog once i understand this ..



1. pass-through : In this the communication occurs in low  compression code like g711

2. NSE based pass-through: in this also the communiction occure in low compression codec like g711 but use a cisco propritory NSE protocol ehich help in detection of fax call aand on detection of CED tone switch from Voice to fax
  
3. Protocol based pass through : It is same as Normal passthrough only difference is switchover from Voice to fax mode is trigered by protocol message for h323 and SIP stack based on protocol , for h323 h245 messages are used to trigger the switchover

Relay Method : This method is basically demodulate the fax signal and remodulate/repack the signal as per defined protocol  there are two type T.38 (open standard) other is cisco proprietory:

Good Link from cisco on Fax : "supportforums.cisco.com/docs/DOC-1360"

Few troubleshooting commands:

For NSE Pass through
debug voip rtp session named-event  : for NSE passthrough troubleshooting check for nse messages like (192 etc)

Protocol passthrough

debug h245 asn1  : check for trigger messages for pass through
debug ccsip messages   
debug sccp messages 

Cisco Call Manager Express

Integration of CME with CUCM on SIP.

Recently done a CME integration with CUCM based on SIP,Its pretty straight forward same as H323 but few points need to be kept in Mind

1. Create a SIP trunk Pointing to the CME IP address and CSS which have access to all the Cluster DN

2. Go to the CME router Do the below config .

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  no h225 timeout keepalive
  call preserve
 sip
  bind control source-interface GigabitEthernet0/1.2 (This will be the VOIP Interface used for creating the trunk)
  bind media source-interface GigabitEthernet0/1.2  (This will be the VOIP Interface used for creating the trunk)

3. Now create a Dial peer to send the dialer number to the CUCM 


dial-peer voice 111 voip
 corlist outgoing int
 preference 1
 destination-pattern 1....
 session protocol sipv2
 session target ipv4:XXX.XXX.XXX.XXX (CUCM IP Address)
 dtmf-relay sip-notify
 no vad
!

Done......:-)

Also few troubleshooting command you can use and are really helpful when you stuck..

debug voip ccapi inout
debug  ccsip messages.

Integration of CME with CUE

The integration between CME an CUE is based on SIP , so to create a integration you need to do few configuration in CUCM and CUE.
Below are list list of step i peformed and it started working ..

1.  Define voice parameter in IOS

config t
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip

2. Configure the CUE interface to communicate

interface ISM0/0
 description *** Connected to CUE ***
 ip unnumbered GigabitEthernet0/1
 service-module ip address 192.168.30.3 255.255.255.0 <CUE IP Adress> !Application: CUE Running on ISM
 service-module ip default-gateway 192.168.30.1
 hold-queue 512 out
!
3. Next steps are to define the dial peer

dial-peer voice 101 voip
 description ***To Internal Cisco Unity Express VoiceMail Pilot ***
destination-pattern 19999 < Voice mail Pilot number > 
 session protocol sipv2
 session target ipv4:192.168.30.3
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!

sip-ua
 mwi-server ipv4:192.168.1.3 expires 3600 port 5060 transport udp
!

4. Define the voice mail pilot number under telephony-service

telephony-service
voicemail 19999

5. Define call forward under ephone-dn

ephone-dn  17  dual-line
 number 54321
 Label TestPhone
call-forward busy 19999 < Forward to voice mail Pilot>
call-forward noan 19999 timeout 10

6.Define ephone DN for Mwi

ephone-dn  40
 number 8000..........
 description MWI ON
 mwi on
!
!
ephone-dn  41
 number 8001..........
 description MWI OFF
 mwi off
!


Addition of Extension module to CME ..

To add a extesion module in CME

Go to the ephone config and type below command

Type <Phone Model> addon <Type Extenstion Module Model>

Thats all from CME Point of view..Now need to proceed to CUE config..

In CUE below steps needs to be followed

1. Create a Voice Mail application
ccn application voicemail
 description "voicemail"
 enabled
 maxsessions 10

2 Create a SIP Trunk to CME

ccn subsystem sip
 gateway address "192.168.30.1
 end subsystem


3. Create a Voice Mail trigger

ccn trigger sip phonenumber 19999
 application "voicemail"
 enabled
 locale "en_US"
 maxsessions 3
 end trigger

4. Create a User in CUE

username cisco create

voicemail mailbox owner "cisco" size 4320
 description "Test mailbox"
 end mailbox

username cisco phonenumber "54321"

Once this basic configis complete it should forward all busy call to voice mail and mwi should also work...


Creating a hardware resources for transcoding and conferencing  require to create profile for those and bind it to cucm usng the ccm group command .



Changing time Zone in CME

Go to Telephony-Service
Time-Zone <Time Zone Number>


Also in the config mode you can define timezone and DST for Router IOS

to define DSt the command is

clock summer-time EST recurring 2 Sun Mar 2:00 1 Sun Nov 2:00  .....................:-)

Debug Command:

For sip 
debug voice register pool 
sh voice register pool registered
 
 

UCCX version 8 Installation Procedure with Screenshots

Here i want to share my experience with UCCX,I recently got an opprurtunity to install UCCX version 8.X

Here are the steps i  followed to install in VMware .

1. Create a VM Machine with required specification mentioned in cisco.com
2. Boot the machine from UCCX bootable CD.
3. Once it start booting it will it will do the media check..
4. Do a media check to check whether the media is good enough for th installation

5.    In the Product Deployment window, click OK to install the Unified Contact Center Express    product suite. The Proceed with Install window appears.

6. The Platform Installation Wizard window displays.Select the Proceed

7. The following window appears for applying an upgrade patch.Choose ‘Yes’ to apply an upgrade patch or ‘No’ to continue

8. Click Continue on the Basic installation page
9. Next Step will ask you to enter the timezone, Enter the required details
10.Next step will ask you to enter the NIC negotition setting ,recommended to put it in Auto
11. Next step will ask you to enter the MTU Size,It is recommended to have the default setting which is "No"

12. Next step will ask you to enter the network setting, Enter the required value.

13. DNS Client Configuration,Select ‘Yes’ to install the DNS client on the server and ‘NO’ to skip.If you select ‘YES’ then enter the primary DNS,secondary DNS and domain in the next window.Here we are choosing ‘NO’ and continue the installation:

14. Enter Administrator Login Information for the Platform (Cisco OS)

15. Enter the certificate related details in thenext window
16. Since we are installing Contact Center Express in HA mode,this server will be the first node(Primary or Publisher) and consequent server will be the Secondary or the Subscriber 

17. In the next screen it will ask if you want to enable NTP, Choose the required option in this case we used Yes

18. Next step will give the field to enter the IP Address of the NTP server.

19. Enter the database security password in this page

20.Next step will ask you to enter the SMTP details if required

21. Next step will ask you to enter the Application userid and password

22. Your Platform configuration is complete now ,click Ok

23. Next step will ask you to enter the Deployment type , As we are doing the integration with CUCM select the appropriate Option.

24. Click Ok, The system will start installting the application.

Go back to your seat and have some coffee, it will take around 30-45 minutes to complete the installation of CCX ..

Next post .. For UCCX integartion with CUCM