This article describes how to set up Cisco Call Manager 6 with IP phones and SIP trunk line to AlphaCom.
Components used in this example:

Cisco Unified Call Manager version 6.1.2.1000-13
Cisco IP phone 7960
AlphaCom E exchange
AlphaCom stations
SIP Intercom

The network is used as an example throughout this article. The network should also preferably have a DHCP server configured. In our case, the DHCP server is configured with IP addresses 1-99=static, 100-255 DHCP automatic.

Configuration example

Cisco Unified Call Manager with Cisco IP Phones

Install Cisco Unified Call Manager server and operating system

Your IT department or System Administrator installs the Cisco Unified Call Manager.

Configure the Cisco Unified Call Manager

Open a browser window and type in the IP address of your Cisco Unified Call Manager (CUCM).
Log on to Cisco Unified CM Administration with the proper Username and Password provided by your CUCM Administrator.
Execute the following configuration: (Parameters not mentioned should be left untouched to their default setting)

Configure a Date/Time Group

Date/Time Group Configuration

System -> Date/Time Group -> Add New

Group Name: Zenitel
Time Zone: Choose your time zone
Separator: -(dash)
Date Format: Choose your date format
Time Format. Choose your time format
Press Save
Press Reset/Restart

Configure a Region

Region Configuration

System -> Region -> Add New

Name: Zenitel
Press Save
Press Reset/Restart

Configure a SIP Trunk Security Profile

SIP Trunk Security Profile Configuration

System -> Security Profile -> SIP Trunk Security Profile -> Add New

Name: AlphaCom
Description: AlphaCom SIP Secure Profile
Device Security Mode: Non Secure
Incoming Transport Type: TCP+UDP
Outgoing Transport Type: UDP
Enable:

Accept Presence Subscription
Accept Out-of-Dialog REFER
Accept Unsolicited Notification
Accept Replaces Header

Press Save
Press Reset/Restart

Configure a Route Partition

Partition Configuration

Call Routing -> Class of Control -> Partition -> Add New

Name: AllLines
Press Save
Press Reset/Restart

Configure a Calling Search Space

Calling Search Space Configuration

Call Routing -> Class of Control -> Calling Search Space -> Add New

Name: DefaultUser
Description: DefaultUser
Select AllLines in the ”Available Partitions” window and use the ”Move Down” arrow to move it to the ”Selected Partitions” window.
Press Save

Configure a Media Resource Group

Media Resource Group Configuration

Media Resources -> Media Resource Group -> Add New

Name: AlphaCom_MRG
Description: AlphaCom_MRG
Select MTP_2 in the ”Available Media Resources” window and use the ”Move Down” arrow to move it to the ”Selected Media Resources” window.
Press Save
Press Reset/Restart

Configure a Media Resource Group List

Media Resource Group List Configuration

Media Resources -> Media Resource Group List -> Add New

Name: AlphaCom_MRGL
Select AlphaCom_MRG in the ”Available Media Resource Groups” window and use the ”Move Down” arrow to move it to the ”Selected Media Resource Groups” window.
Press Save
Press Reset/Restart

Configure a Device Pool

Device Pool Configuration

Device Pool Configuration, continued

System -> Device Pool -> Add New

Name: Zenitel
Cisco Unified Communications Manager Group: Default
Calling Search Space for Auto-registration: DefaultUser
Reverted Call Focus Priority: Default
Date/Time Group: Zenitel
Region: Zenitel
Media Resource Group: AlphaCom_MRGL
Location: Hub_None
SRST Reference: Use Default Gateway
Press Save
Press Reset/Restart

Configure a Media Termination Point

Media Termination Point Configuration

Media Resources -> Media Termination Point -> Select: MTP_2

Set Device Pool: Zenitel
Press Save
Press Reset/Restart

Configure a SIP Trunk

Trunk Configuration

Trunk Configuration, continued

Device -> Trunk -> Add New

Trunk Type: SIP Trunk
Device Protocol: SIP
Press Next
Device Name: AlphaCom
Description: AlphaCom
Device Pool: Zenitel
Media Resource Group List: AlphaCom_MRGL
Enable: 'Media Termination Point Required'
Calling Search Space: DefaultUser
Destination Adress: Your AlphaCom’s IP adress
SIP Trunk Security Profile: AlphaCom
SIP Profile: Standard SIP Profile
Press Save
Press Reset/Restart

Configure a Route Group

Route Group Configuration

Call Routing -> Route/Hunt -> Route Group -> Add New

Name: AlphaCom_RG
Select AlphaCom from ”Available Devices” and press ”Add to Route Group”
Press Save

Configure a Route List

Route List Configuration

Call Routing -> Route/Hunt -> Route List -> Add New

Name: AlphaCom_RL
Cisco Unified Communications Manager Group: Default
Press Save
Press Add Route Group
Route Group: AlphaCom_RG [NON-QSIG]
Press Save
Press Save
Press Reset/Restart

Configure a Route Pattern

Route Pattern Configuration

Route Pattern Configuration, continued

Call Routing -> Route/Hunt -> Route Pattern -> Add New

Route Pattern: 2xxx (This allows calls to all numbers beginning with 2 in the AlphaCom)
Route Partition: AllLines
Description: AlphaCom_RP
Gateway/Route List: AlphaCom_RL
Press Save

Configure a Phone

Phone Configuration

Phone Configuration, continued

Device -> Phone -> Add New

Phone Type: Choose your IP phone type from the drop down list
Press Next
Select the device protocol: SCCP (This is the standard Cisco IP phone protocol).
Press Next
MAC Address: Enter your IP phones MAC address
Description: Zenitel 1030
Device Pool: Zenitel
Phone button Template: Choose your IP phones specific template
Device security Profile: Choose your IP phone specific profile
SIP Profile: Standard SIP Profile
Press Save
In the upper left corner press ”Line [1] – Add a new DN]
Enter wanted directory number (here 1030)
Route Partition: AllLines
Calling Search Space: defaultUser
ASCII Display (Internal Caller ID): CISCO 7960 (This is the wanted name for the phone. This name will be displayed in the AlphaCom stations, and should thus be configured the same as the Global Number in the AlphaCom)
Press Save
Press Reset/Restart

Activate Cisco Services

Navigation

Navigation -> Cisco Unified Serviceability

Service Activation

Tools -> Service Activation

Activate the following services:

Cisco CallManager
Cisco Tftp
Cisco IP Voice Media Streaming App
Press Save (takes some time)

Configure Cisco 7960 IP phone

Connect a network cable to the phone. In this example we connect to a network with a DHCP server.

Power up the phone (48VDC, center positive).

Click "settings" button (rightmost, below the "i" button).

Select 3 Network Configuration. Notice that many fields (DHCP server, IP etc) is already set up automatically by the DHCP server.

Note: In the Network Configuration menu, settings are locked by default. To be able to change them, you must press this sequence: * * #.

33 Erase configuration: Change to Yes.

Save. The phone will reset. Return to the Network Configuration menu.

32 Alternate TFTP: Yes.

8 TFTP server 1: 10.5.2.43. (IP address for Cisco CM server).

Save. The phone will reset.

The phone will now register with Cisco CallManager, download configuration and be ready for use. (This takes about 1-2 minutes, or longer if an automatic firmware update is performed).

AlphaCom Configuration

AlphaWeb Configuration

Assign IP address to the AlphaCom E Ethernet Port(s)

Log on to AlphaWeb and enter a valid IP address on the Ethernet port. In the example below, Ethernet port 1 is used. Consult your network administrator to obtain the IP address.

Assign IP routing to the AlphaCom E Ethernet Port(s)

Log on to AlphaWeb and create a default routing. This is essential if the CUCM and ACE are on two different networks.

Insert SIP Trunk licenses

Log on to AlphaWeb and install the SIP Trunk License.

Firewall (filter) settings

Enable the SIP protocol (UDP port 5060) on the desired Ethernet port. It is by default enabled for Ethernet port 1.
Enable for VoIP audio (UDP port 61000 - 61150) on the desired Ethernet port. It is by default enabled for Ethernet port 1.

AlphaPro Configuration

Create a SIP Trunk Node

From the AlphaPro main menu, use the ‘+’ button next to the ‘Select Exchange’ dropdown list to create a new exchange. The exchange type must be set to ‘SIP Node’.

Set the parameters as shown in the figure to the right.
The SIP Trunk IP address must be identical to the IP address of the Cisco Call Manager.

Note: If the AlphaCom is configured with a SIP Registar node in addition to the SIP Trunk node, the SIP Registar node must have a lower node number than the SIP Trunk node.

Define the AlphaCom / SIP routing

In Exchange & System > Net Routing use the Insert button to create a route between the AlphaCom and Cisco Call Manager.
Set Preferred codec to G711u and RTP Packet Size to 20 ms.

Create access numbers

The Cisco phones can be accessed in three different ways:

Prefix number: Dial Prefix + Phone number. “Phone number” will be called
Integrated Prefix number: Dial Prefix + Phone number. The prefix will be included as a part of the called telephone number.
Global number: Dial the phone number without using prefix

Prefix number
The directory number must be programmed in the AlphaCom directory table with feature 81 and Node = SIP Trunk node number (100 in this example). In the field “Collect N more digits (SIP)” you must enter the maximum number of digits in a phone number.

When the prefix is dialed, the AlphaCom will wait for more digits. When the number of digits specified in the “Collect N more digits (SIP)” is collected, a call setup message is sent to the Cisco Call Manager. If fewer digits are entered, the AlphaCom will time out after 4 seconds, and the call setup message will be sent. You can also terminate the digit collection by pressing the M-key. The call setup message will then be sent immediately.

In the example to the right the directory number 0 is used as a prefix.
Dialing examples:

0 + 12345678: Telephone number 12345678 will be called

0 + 1234: After a 4 second timeout, telephone number 1234 will be called

0 + 1234 + M: Telephone number 1234 will be called

Integrated Prefix number
The directory number must be programmed in the AlphaCom directory table with feature 83 and Node = SIP Trunk node number (100 in this example). In the field “Collect N more digits (SIP)” you must enter the maximum number of digits in a phone number.

When the prefix is dialed, the AlphaCom will wait for further digits. When the number of digits specified in the “Collect N more digits (SIP)” is collected, a call setup message is sent to the Cisco Call Manager. If fewer digits are entered, the AlphaCom will time out after 4 seconds, and the call setup message will be sent. You can also terminate the digit collection by pressing the M-key. The call setup message will then be sent immediately.

In the example to the right the directory number 57 is used as a prefix.
Dialing examples:

57 + 12345678: Telephone number 5712345678 will be called

57 + 1234: After a 4 second timeout, telephone number 571234 will be called

57 + 1234 + M: Telephone number 571234 will be called

Global number
The directory number must be programmed in the AlphaCom directory table with feature 83 and Node = SIP Trunk node number (100 in this example). The field “Collect N more digits (SIP)” must be left blank.

When the global number is dialed, the AlphaCom will immediately send a call setup message to the Cisco Call Manager.

In the example to the right the directory number 12345678 is defined as a global number.
When dialing this number a call setup message is sent to the Cisco Call Manager, instructing it to call this phone number.

Update the exchange

Log on to the exchange and update the exchange by pressing the SendAll button. Reset the exchange when the transfer is finnished.

Verify the configuration

You should now be able to dial from an AlphaCom station to a Cisco IP station and from a Cisco IP station to an AlphaCom station.

Troubleshooting:

Wireshark - capture data

Cisco Call Manager 6 configuration guide

From Zenitel Wiki

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