Jun 20

Finding and Changing the Default Directory URI Partition

By default Cisco CUCM 10 includes a system partition called “Directory URI”. All SIP URIs are placed into this partition automatically.

As this is a system partition, it cannot be removed from the system, however you can specify another partition where all SIP URIs can be automatically placed into.

System -> Enterprise Parameters -> Directory URI Alias Partition. Drop the menu down and select a partition you would like the SIP URIs to be placed into.

As you can see I have selected the partition called INTERNAL_PT for my SIP URIs.

Cisco CUCM - Directory URI Parameter

Jun 01

Blocking Inbound Calls based on Caller-ID using Cisco Voice Gateway

Blocking Inbound Calls using Cisco ISR Voice Gateway is a simple 3 step process. First step is to create or append to a Voice Translation Rule, next is to map the Voice Translation Rule to a Voice Translation Profile. Then finally attached the Voice Translation Profile to either a Voice-Port or Inbound Dial-Peer.

Lets looks at this more closely.

Step 1. Create a Voice Translation Rule.

From the Global Configuration Mode enter in to the Voice Translation Rule Sub Menu, be sure to select a unique Rule Set ID. Now enter the new Translation Rule identifying the Calling Number ( Caller-ID) of the caller you would like to block.

(config)# voice translation-rule 200
(cfg-translation-rule)# rule 1 reject /06408343425/

Step 2. Create a Voice Translation Profile and apply the Voice Translation Rule

Next step is to create a Voice Translation Profile. Within the Voice Translation Profile we need to translate the Calling party referencing the Rules Set we created above.

(config)# voice transalation-profile ISDN-INBOUND
(cfg-translation-profile)# translate calling 200

Step 3. Attach the Voice Translation Profile to either the Voice Port or Inbound Dial-Peer

I prefer to attached Voice Translation Profiles on the Voice-Port, however attaching to the Dial-Peer is also very acceptable and correct. In either instance the commands for attaching Voice Translation Profiles is the same.

Voice Port Configuration

(config)# voice-port 0/0/0:15
(cfg-voice-port)# translation-profile inbound ISDN-INBOUND

Dial-Peer Configuration

(config)# dial-peer voice 1 pots
(cfg-dial-peer)# translation-profile inbound ISDN-INBOUND

Troubleshooting

Debug isdn q931 – Displays the Calling Party Number format

Debug voice translation – Displays realtime verbose output of matching of Voice Translation Rules for incoming calls.

Show run | s voice – Provides a Running Configuration output focusing on the Voice configuration section.

Show voice translation-rule – Displays all configured Voice Translation Rules

Show voice translation-profile – Displays all configured Voice Translation Profiles.

Test voice translation-rule RULE-ID Phone-Number – Tests Voice Translation Rules, will display translated number.

May 25

UCCX 10 – Intermittent MoH

Recently I had an issue where MOH intermittently was not playing to callers in a UCCX Queue. The symptoms were not logical. MOH played ok to a PSTN caller who connected to a CTI Port that was later also used for another PSTN caller where MOH did not play.

After checking the Data Sync in the AppAdmin portal, I could see that the renaming of the MOH file caused the UCCX and CUCM to become out of sync. A Simple ‘Data Resync’ on the Data Synchronisation page resolved the MOH issues.

Data Synchronisation - Cisco UCCX 10

May 09

Cisco Licensing Snippet – Single Number Reach (SNR)

Cisco Licensing for CUCM 9.0+ caters for a few different scenarios through the use of licensing levels/types. CUWL Pro, CUWL Standard, UCL Enhanced / Plus, Basic and Essential. I will go through the main type of licensing in a later blog, but for now I’ll explain the licensing requirements / quirks around Single Number Reach (SNR).

Mobile Connect and SNR are to describe the same feature in CUCM, which is to allow users to have inbound calls redirected to their mobile devices (or other PSTN Numbers) automatically. This feature can be managed by the user themselves, configuring schedules, disabling the service, changing phone numbers, or configuring the number of times my deskphone will ring before I want the call to be directed to my mobile phone etc. Just about all of my Telephony deployments have included this requirement.

So now, for the licensing aspect. An End User with the ‘Mobile Connect’ box checked will consume a single Basic License. For those who don’t manage their CUCM license with care, enabling SNR for all users in the Business will blow out your licensing budget.

As shown in the below image, I have 14 Basic Licenses being consumed by the SNR feature alone. This can tip me over the licensing edge if I don’t have my devices configured correctly.

CUCM License Summary

The next image shows that I have consumed a Basic License and I do not own any other Device. This is important to note.

CUCM Basic License

What happens if I didn’t purchase any Basic Licenses?

As we all know Cisco CUCM licensing levels allow the lower levels to borrow licenses from the upper levels. Example. SNR will consume a Basic License, but if there are no Basic Licenses to allocate, it will look to the next level and check if the Enhanced license level has any available licenses to allocate, if not, the system will check the next level repeating the process all the way to CUWL Professional. So in effect your single SNR enabled feature can consume a CUWL Pro License, and I’m sure you didn’t spend all that money of CUWL Pro licenses for it just to be consumed by the SNR feature.

How to Avoid Consuming a Basic License

If you know your CUCM licensing basics, you’ll know that UCL Enhanced allows an end user to own a single device, a UCL Enhanced Plus license allows you to own two devices, CUWL Stand and Pro allows you to own 10 devices, so you could say CUCM 9.0+ licensing is now very user centric.

So, here is the trick, not a trick really just configuring CUCM Licenses correctly. You need to assign ownership for devices to your end users. That’s it. So if you have UCL Enhanced Plus, configure the users deskphone to have an owner. By configuring this, the SNR feature will sit within the boundaries of the UCL License and NOT consume any additional licenses. I did say CUCM licensing is user centric, this is just a way for Cisco to force you to configure Owners for Devices.

The below image is a snippet from a Device Configuration page, showing the Owner Field.

CUCM Device Page

Couple more screen shots displaying the UCL Enhance Plus license summary page in CUCM, detailing a user who owns two devices and also has the SNR Feature enabled.

CUCM UCL Enhance Plus Summary

Continuing from the above screen shot, displays the actual details for this UCL Enhanced Plus User. As you can I also added an Analogue device to this End User to show that both IP Phones and Analogue Devices can be assigned ownership.

CUCM UCL Enhanced PLus License Detail

May 05

On Demand Call Recording with UCCX v10

On Demand Call Recording SPANs RTP Packets from the IP Phone to the PC the CAD Software is installed, hence the PC must be directly connected to the IP Phone for this to work.

The CAD Software will then send the voice stream to the UCCX Server to store the recorded conversation as a WAV file. The WAV files can then be accessed via the Supervisor Console to playback and also option to save the WAV file for long term retention.

Lets get started on how to configure on Demand Call Recording.

Requirements:

1. Enhanced or Premium licenses
2. PC NIC Card supporting tagged vlans
3. IP Phones supporting SPAN to PC feature
4. PC directly connected to the PC port on the IP Phone

Configuration Steps

CUCM

1. Enable the SPAN to PC feature on the device configuration page.
2. Disable G722 codec negotiation either on the device configuration page or system wide via the Enterprise Parameters

UCCX Desktop Admin (Workflow Administrator)

1. Navigate to the Workflow created for the Agents. Under the appropriate Workflow Group, go to CAD Agent then User Interface. The CAD User Interface configuration page will be displayed in the right pane.

2. From the Toolbar Tab select the first available Task. Ensure the Visible checkbox is ticked, and then Add an Action. The Select Window is displayed. Here is where we will add the required action. Click ‘Utility Action’, now Click ‘New’. Give the Action a Name, Select ‘Record’ for the Action, then ‘Start’ for the Attribute. Click OK to save.

3. Complete the process again, but this time for the Stop Recording Action.

4. Now we have two Actions configured. We will select the Start Recording Action, and click ‘Add Action’. This adds the Start Recording action into the Task pane. Give the Task a Hint. Also customise the icon, this is optional. Then click ‘Apply’

5. Select the next available Task and Ensure the Visible checkbox is ticked, then Add an Action. The Select Window is displayed. Goto the Utility Action Tab. You will see the previous two Actions we configured. This time we will select ‘Stop Recording’ and click Add Action.

6. Give the Task a Hint and customise the icon. Click ‘Apply’ when complete.

Cisco Desktop Administrator

1. Access the Cisco Desktop Administrator through web administration pages for UCCX.

2. Navigate to Personnel, then Agents. On the Agents Window, for the required Agents select the Workflow Group where the Call Recording Actions are configured.

3. Navigate to Services Configuration, then Multiline, Monitoring & Recording, then VoIP Monitoring Device. Select the Monitor Service and enable Desktop Monitoring. The Monitor Service will more than likely be the UCCX Server.

4. Click SAVE.

Agents PC

1. Navigate to c:\Programs Files\Cisco\Desktop\bin – Then double-click the PostInstall.exe. Ensure the UCCX Server IP Address is correct. Click Next, now select the NIC that is connected to the IP Phone. Click OK.

2. Restart CAD software if you had this open. You will now see the Two new Tasks that were

3. As a side note, just ensure the NIC supports Priority and VLAN Tagging.

Retention Notes:

By default each recording message is retained for 7 Days only. The Supervisor has the option to extend the retention to 30 Days for each individual recording. If you wish to store the recording for a longer period, you will need to Save the recording as a WAV File on your PC or Storage Server.

Apr 25

Single Inbox CUCM 9.1.2 with Microsoft Exchange 2010

Quick note on how to setup Single Inbox between CUCM 9.1.2 and Exchange 2010, I won’t go into to much detail as this will provide an overview of the configuration steps.

Configuring a Connection to the Exchange Server

Under Unified Messaging -> Unified Messaging Services Add New Messaging Service. Give the Service a Name, then select the Web Based Authentication Moe and Protocol. Important Note. Ensure these two parameters match the IIS configured parameters for the EWS Virtual Directory. If they do not match, you will receive either a 403 Error (Protocol – HTTP/HTTPS) or a 401 Error (Authentication).

Optional: If you select HTTPS for the protocol, you can selection to validate the Server certificate or not. If you wish to validate the server certificate, you will need to export the certificate chain and import into CUCM.

I specify the Exchange Server, if the CAS Servers are Load Balanced, then this name/IP Address can be for the LB Address. The select the appropriate Exchange Server version.

For the Active Directory Account information section, have a standard user created in Active Directory, this user must not have a mailbox associated to the account and no additional right in Active Directory. We will need to assign this user Impersonation rights on the Exchange Server. I will cover this below.

Be sure to include the Domain prefix in the username field. Eg. DOMAIN\username

Ensure the “Synchronise Connection and Exchange Mailboxes (Single Inbox)” is checked.

Save the configuration and test. The Test basically checks for connectivity to the Exchange Server, this test will not check authentication mode or protocol.

Configured Smart Host Settings and Allow Unity to Send SMTP to Smart Host.

I like to also add the smtp smart host information in. This being configured also allows for email notification via smtp for voicemails missed calls etc.

Under SMTP Configuration -> Smart Host, simply enter the Exchange Hub Transport Server’s IP Address.

Under SMTP Configuration -> Server. Goto Edit -> Search IP Address Access List. Add New Access IP Address, enter the Hub Transport Server’s IP Address in again and check “Allow Connection”

Save the Configuration

Class of Service for Voicemail Users

Class of Service defines the rights/permission or access levels given to the members of the Class of Service object. Basically restricting what the Voicemail user can and can’t do.

By default Voicemail users are placed into the Class of Service called “Voice Mail User COS”. By default this Class of Service does not allow Single Inbox connection, or retrieval of voicemail using IMAP etc.

Under Class of Service -> Class of Service. Click into the required Class of Service object. On the configuration page, under the Licensed Features check the “Allow Users to Access Voice Mail Using an IMAP Client and/or Single Inbox” checkbox. The select the radio button labelled “Allow IMAP Users to Access Messages Bodies”

Save Configuration.

Enabling a Voicemail User for Single Inbox.

Under Users -> Users. Search and select the appropriate user. On the basic configuration page, click Edit -> Unified Messaging Accounts. Add New Unified Messaging Account. Under the Account Information Section, either select the “Use this Email Address” and enter in the email address manually or select the “Use Corporate Email Address: email_address_of_user. In my instance I have imported users from Active Directory and all users have the email address attribute completed. So I simply select use Corporate Email Address. Save Configuration and Run a Test. This test checks the Authentication mode and protocol, like I mentioned above two main error are 401 and 403. Just go back and the IIS Virtual Directory for EWS and ensure both protocol and authentication mode match.

Modifying the User Template for new Voicemail Users

I always modify the User Template to reflect the most up-to-date configuration for new users. This simplifies the configuration of new user.

Under Templates -> User Templates select “voicemailusertemplate”. On the basic configuration page goto Edit -> Unified Messaging Accounts. Add New Unified Messaging Account. Under Account Information check the “Use Corporate Email Address” checkbox. (Note: Users must have the email attribute completed for this to successfully setup a Unified Message Account).

Save configuration. Now when creating new voice mail users, simply import sing the above User Template and the Unified Messaging Account will be automatically created.

Apr 15

Configuring Faxes and EFTPOS Machines on FXS ports SCCP Controlled over SIP Trunk to IOS Router

Call flow as below:

Fax Machine -> FXS -> SCCP -> CUCM -> SIP -> IOS -> ISDN

Quick overview of a working Fax/EFTPOS setup on a SCCP Controlled FXS Module.

Note: SCCP supports only NSE switchover negotiation, not Protocol Based. SIP Support both.

Global Fax Configuration

Voice service voip
fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback none

SCCP Configuration

sccp local GigabitEthernet0/0
sccp ccm 10.10.10.1 identifier 1 priority 1 version 7.0
sccp ccm 10.10.10.2 identifier 2 priority 2 version 7.0
sccp
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2

STCAPP Configuration

stcapp
stcapp ccm group 1

Voice Port Configuration

voice-port 0/1/0
compand-type a-law
cptone AU
timeouts ringing infinity

Dial-Peer Configuration

dial-peer voice 7000 pots
service stcapp
port 0/1/0

Regions – Codecs

Regions between the Analogue Devices and the SIP Trunk at the Main site must talk with no codec compression, otherwise faxing handshake is distorted. This extends to the SIP Dial-Peer on the ISDN Router.

Show Commands

Two show commands to use to identify what codec or fax protocol is being negotiated

Show voice call summary
Show voice dsp