Cisco CUCM and CUC Softswitch Integration
Introduction
As a part of the FP29.1 program, the Tellabs voice ONTs were tested with the Cisco Unified Call Manager (CUCM) and the Cisco Unity Connection (CUC) version 10.5 for voice services. This document covers how to configure the Tellabs system to be compatible with Cisco Call Manager. It also covers some key CUCM/CUC settings that are important for interoperability with the Cisco Call Manager for SIP-based lines. Tellabs does not sell or support Cisco products and the configuration shown here is likely one of many that will work. This document simply documents one working configuration used during testing.
Document Number
ENG-010547
Applies To
All Tellabs OLTs (1150/1134/1131)
Tellabs 142R ONT
Releases Tested
- Tellabs SR29.1 Software Release, revisions 015708 and above.
- Cisco CUCM 10.5.2.10000-5
- Cisco CUC 10.5.2.10000-5
Testing Exceptions Noted
The following issues/exceptions were noted during the testing of the system:
- CUCM only supports a maximum of 6 lines and so is not practical to use with the 729GP even though it is compatible.
Assumptions
This document only covers the configuration needed for proper interoperability with the Cisco Unified Call Manager Softswitch. The document assumes the following is true:
- TConfig is Set Up: TConfig is the Tellabs Configuration tool for voice and is a part of the EMS software. TConfig should configured properly and operational on the OLT prior to starting this AppNote. If TConfig has not been set up, or you are unsure of its status see ENG-010473 Configuring Softswitch Voice found on the portal. Execute only the portions associated with TConfig Setup and ONT VoIP tab setup. Prior to starting this procedure, you should have verified that the ONT has an IP address and can successfully download configuration data from the EMS.
- EMS is reachable from the ONT: The EMS TConfig tool must be able to communicate with the EMS either via having a routed path to the ONT, or a secondary NIC on the voice VLAN. This is necessary for XML file download.
- DHCP Server Present on the Network: There must be an operational DHCP server on the VLAN designated for voice. The ONTs only support DHCP.
- Optional DNS Server: If you intend to use hostnames, then a DHCP server must be defined on the voice network, and it must have entries to resolve the needed hostnames. If only IP addresses are used for reaching the Softswitch then the DNS Server is required. The DNS server entries if used must be configured in DHCP to hand the DNS entries to the ONT. DHCP is the only mechanism supported for communicating the DNS server IPs to the ONT.
Tellabs ONT Configuration
This section will give a sample configuration for the Tellabs ONT that is known to work with the Cisco Call Manager for all tested features.
The Tellabs ONT is primarily configured via three standard profiles which define the connectivity to the switch and standard policies for voice lines. Most installations only require one or two sets of profiles based on usage.
The three profiles used by the system are:
- Equipment Profile - This profile defines TConfig refresh interval, SIP timer settings and country code/region.
- Service Profiles - Service profiles define how to connect to the switch and the registrar. It also allows configuration of the RTP and SIP layer behaviors.
- Call Features Profile - Defines the switch type, dialing plan, Call features supported, hotline/warmline, etc.
Equipment Profile
The Equipment profile defines basic connectivity settings for the system. The default profile in most instances will work well for most users. The Equipment Profile defines the following settings:
- TConfig Refresh Interval - How often the voice line should poll the server for changes in the profiles. Default is once per day. (This only affects global changes to profiles when no line changes have been performed).
- SIP Timers - Allows configuration of Key SIP timers. Typically changes to this are not required.
- Country Code - Allows configuration of the call progress tones and local impedance for the voice line to match other countries and regions. The default is US.
Detailed explanations for each field can be found in Tellabs documentation if modifications are required. Most installations only have a single Equipment Profile for all lines. Typically, the default equipment profile can be used without modification.
The Default profile has the following characteristics:
- Expects UDP operation on standard SIP port 5060.
- Refreshes profiles once per day (only affects global changes to profiles where no line change is made.
- Sets of lines for US tones and impedances.
Services Profile
Only a few modifications are needed for proper operation of the system with the Cisco Softswitch.
- Registrar URI - The Registrar URI should be set to the IP Address of the primary CUCM.
- Proxy Address - The Proxy Address should be set to the IP address of the primary CUCM.
- Realm - The realm defines the security realm to which the SIP line will be authenticated. It is critical that this match the Softswitch or authentication will fail and no voice service will be possible. The ONT strictly enforces checking of the realm and will discard all messages without a matching realm when challenged for authentication. The default realm for CUCM if not modified, is ccmsipline. This is the most common issue with authentication.
- Enable RFC2833 - It is recommended that RFC2833 be enabled. This tells the ONT to send dialed digits as messages rather than audio. Typically this works better with PBX style switches such as CUCM that are using audio feedback and/or using low bandwidth codecs.
- Register Headstart - For proper operation, it is critical that the registration interval on the Cisco Softswitch SIP Profile "Timer Subscribe Expires (seconds)Required Field" be several times the value of the registrar headstart. If this is not true it can cause problems with registration with the switch. The register period configured in the ONT is only a suggestion to the switch and the switch will define the registration interval. It is recommended that the minimum registration interval on the Cisco Call Manager be set to 3x the register headstart. Please note the Cisco CUCM does not properly implement the Timer Register Expires setting and it is ignored by the switch.
The following changes should typically be made to the profile:
- Interdigit Timer - This timer defines how long the system will wait to consider dialing complete before sending the digits. A value of 8 seconds is recommended, but can be tuned to the sensibility of users. Some dial slowly, others want more responsivity.
- Critical Timer - Defines the time to wait when there is a partial match on multiple patterns in the dial plan (such as when 7 digit and 10-digit dialing is supported). Digits are sent after a partial match is found, and the Critical timer expires with no other digits dialed.
- Timed Release - This defines the amount of time the line must be onhook before the call is terminated. Three seconds is the recommended setting.
Call Features Profile
It is recommended that two Call Features profiles be created, one with Call Waiting On, the Other with Call Waiting Off. This allows control over whether the users can use these features. Most other features should be enabled and allow the switch to control them. Since Call Waiting is an endpoint function, it should be controlled at the ONT.
If the Cisco Call Manager has not been added to the list of switches in your EMS it may be necessary to add it on the TConfig Profiles->General Tab.
- Softswitch Name Full Form - CISCO-CALL-MANAGER
- Softswitch Short Form - CUCM
- Add + Apply - the user MUST press both Add followed by Apply so that the new Softswitch will be added. If Apply is not pressed, the entry will NOT be added.
The following fields should be modified:
- Softswitch Name - Select CISCO-CALL-MANAGER
- Digit Map - Load the appropriate digit map for your installation. Typically one of two profiles will be used:
- Standard-NorthAmerican-Telco-Dialplan - Defines a standard dial plan used for lines where it has a typical telco dialing plan for North America. Defines 7 digit dialing, 10 Digit Dialing, 911 with emergency headers, info numbers (211,311, etc), and standard Vertical Service Codes.
- Standard-NorthAmerican-PBX-Dialplan - Defines additional patterns necessary for 4 digit extensions not beginning with 9, and all standard North American dialing plans when prefaced with a 9 to get out to the PSTN.
- Dial Plan Changes - Dial plans can be quite complicated and Tellabs only supports changes to dial plans when any changes are made with the support of Tellabs TAC. Customer changes made without Tellabs TAC support must be supported by the customer.
- Enable CW - Recommend creating two Call Features Profiles, one with Call Waiting enabled (with -CW at the end of the profile name) and one with Call Waiting Disabled (with -NoCW at the end of the profile name). This allows control over which users have call waiting and which have no call waiting.
- Hold - Recommend Hold is allowed on all profiles.
- Enable 3WC - Enable Three-Way Calling for users. Typically can also be configured on the switch side.
- Message Waiting - If the message waiting light or stutter dial tone is needed, it is controlled by this attribute. Message waiting MUST be properly configured on the switch for proper operation.
- Visual - Enables the message waiting light on phones that support FSK based message waiting lights. NEON and Voltage Based PBX message waiting lights are NOT supported.
- Audible - Enables the stutter dial tone indication for message waiting.
- Enable Caller ID - Typically it should be set to Enabled. It can also be controlled on the switch so enabling it allows it to be driven by the switch side.
Voice Line Configuration
Each voice line must be configured with the per-user configuration.
This is done via the Ports View->Voice Tab, and right-click / properties Menu.
With the SR29.1 load, the entire configuration can be made from the gridded display for all lines:
The following fields should be entered:
- Description - Name given to the line. Typically used to identify users name.
- AOR URI - For simple installations, just enter the users phone number or extension. The EMS upon apply will construct the AOR as follows: sip:<phone_num>@<proxy_address> which is taken from the Services Profile. If desired, the AOR can be explicitly entered to match local needs. The Cisco Call manager by default appears to accept the IP address of the Call Manager for registrations. Hostname/Domain can be used in the proxy address part of the AOR if needed.
- Contact URI User - will be auto populated and typically should not be modified.
- Softswitch User - The username used to log into the switch. On most systems it is the directory number, but varies by installation.
- Softswitch Password - The password sent to the switch as part of digest authentication of the switch line. The password will only be used if authentication is on and the switch challenges registrations or outgoing calls. The system requires that a password be entered, so enter a dummy password if authentication is off on the switch.
- Services Profile - Select the services profile defined in the previous steps.
- Call Features - Select the Call Features profile defined in the previous steps.
Cisco Call Manager/CUC Configuration Overview
These sections will overview at a high level the configuration of the Cisco Call Manager. There may be other configurations that will work, but this is the configuration used for all qualification testing.
In the testing, Cisco Unity Connection was used to provide voice mail services and to determine MWI status. Message Waiting Interface is how analog lines indicate a message waiting via a Visual Light indication, or an Audible stutter dial tone. Cisco supports the Unity to Cisco Call Manager connection to be set up in one of two modes, SIP or Skinny. SIP was used in qualification testing as the SIP interface uses the standard NOTIFY messages for MWI and they are passed through the CUCM directly to the SIP endpoints. This is the preferred configuration for SIP interoperability when MWI is being used.
Several of the profiles are modified from their defaults and so are commended to be copied to preserve system defaults.
At a high level, the system configuration tasks are:
- Creation of SIP profile on CUCM
- Voice Mail Ports and Pilot creation.
- Voice Mail Profile Creation
- Phone Security Profile Creation
- Creation of SIP Trunk Profile on CUCM
- Creation of SIP Trunk on CUCM
- SIP Trunk Route Pattern Creation
- CUCM Users Creation
- CUCM Phones Creation
- Creation of Phone System on CUC
- Creation of Phone System Port Group on CUC
- Creation of Phone System Ports on CUC
- Creation of Users on CUCM
- Creation of Phones on CUCM
- Creation of CUC Phone System
- Creation of CUC Port Group
- Creation of CUC ports.
- Creation of CUC Users for Voicemail
CUCM SIP Profile Configuration
A copy of the standard profile needs to be made for the SIP profile so that modifications can be made without affecting defaults:
The registration interval needs to be at least 2x to 3x of the register head start configured in the Services profile of the Tellabs ONT voice line. If it is not, it can cause problems with registration. In this example, I moved the registration time up to every 6 minutes. This also obviously affects how quickly a failed voice line is detected, so care should be taken in ensuring the right relationship between register head start on the ONT services profile and the Timer Subscribe Expires. It should be noted that the value configured on the Registration timeout in the Cisco SIP profile seems to have no relationship to the registration behavior.
CUCM Voice Mail Port Creation
The voice mail port creation is standard and does not have any special characteristics. Screens are included for completeness.
In this example I created two VM ports but it should be governed by the size of the installation and number of active voice mail sessions could be active at any given time. The entire configuration below is standard practice and there is no ONT specific configuration.
The directory number defined is later added to a hunt group which is used as the voice mail pilot.
It should also be noted that all the examples use the default partition and calling space which in any sizable installation, multiple partitions and search spaces would be utilized.
The voice mail pilot must be created to use as the pilot for the voice mail hunt group.
CUCM Voice Mail Profile
The Voice mail profile defines the binding of a voice mail system to the voice mail pilot number and is configured on the lines.
CUCM Phone Security Profile
It is recommended that Digest authentication be turned on for the "Phone" devices that represent the ONTs. This will cause the Cisco Call Manager to challenge the ONT on registration and invites and helps to secure the installation. The username and password used to authenticate is the User associated with the phone and the digest credentials specified.
CUCM SIP Trunk Configuration
The SIP Trunk has an associated SIP profile and needs to have several settings set up to allow refers and to allow for notifications or NOTIFY messages associated with MWI to flow across the trunk.
Accept Out of Dialog Refer should be enabled.
Accept Unsolicited Nofity should be enabled.
Accept Replaces header should be accepted.
This profile does not enable authentication across the trunk since both CUC and CUCM were both on the same machine in a non production environment, but it is recommended that production environments use authentication and secures the turn between CUC and CUCM to maintain proper security.
Redirecting Diversion Header Delivery Inbound is needed due to the routing of voicemail through this interface.
Redirecting Diversion Header Delivery Outbound is needed due to voicemail being routed through this trunk.
The SIP Information section should be used to define the IP address of the Cisco Unity Server that the Call Manager will connect to.
The SIP Trunk security profile that was created previously should be selected.
For the SIP profile, the standard SIP profile can be utilized.
SIP Trunk Routing
A trunk route is needed to route all the voice mail calls toward the CUC. The configuration is all standard for any Cisco CUCM/CUC using a SIP trunk to communicate.
- Route Pattern: A route pattern should be created using the Voice Mail hunt pilot as the route pattern.
- Gateway / Route List: The Gateway / Route list should point at the SIP trunk that runs between the CUCM and CUC.
- Call Classification: The Call Classification should be OnNet.
User Configuration
For each ONT, a User must be created so that it can own the device.
Both lines of an ONT will have the same Softswitch user ID on the Cisco CUCM to mimic the behavior of a 2 line SIP phone. Using the same login minimizes the number of licenses consumed on the Cisco switch.
- User ID: Enter the user ID, any user ID can be used. In this example it was the same as the phone number but that is not required. This is used in all authentication of the user including digest authentication on SIP.
- Password: Enter the Password and Confirm it. Password is not used by the SIP interface but used on other interfaces.
- Pin: Enter the PIN and confirm it. (PIN is not used by the voice line, only digest credentials.
- Digest Credentials: Enter and confirm the Digest Credentials, these will be used for authenticating the voice line to the switch on the SIP register and invite messages.
- Controlled Devices: Do not associate the controlled device, this will happen for you automatically once you select the Owner User in the Phone configuration screens.
Phone Configuration
For each ONT a phone needs to be created and directory numbers associated with each of the lines on the ONT. Use the Device->Phone->Add New to create the new phone device to represent the ONT.
- MAC Address: While it is not critical that the MAC address match the ONT, it is good practice to enter the MAC address of the ONT for tracking purposes.
- Description: Enter a Description for the line for tracking purposes.
- Phone Button Template: Select Third-Party SIP Device (Advanced)
- Owner User ID: The ONT has to be owned by some user. Licenses are based on users.
- Device Security Profile: Select the Device Security Profile created previously.
- Digest User: Select the user name that was created under End User and typically would be the same Owner User selected above.
Click on each of the line links which correspond to voice lines on the ONT and create directory numbers for each.
- Directory Number: Directory number should be the number chosen for the line.
- Associated Device: Associated Device will be automatically populated.
- Voice Mail Profile: Voice Mail Profile should select the profile previously created.
In most instances, if you are providing voice mail you will need to forward on busy or unregistered. Forward on unregistered ensures that if the ONT is down or offline that messages will be collected from any callers.
- Display (Caller ID): Enter the Display Caller ID. This will be shown on the called party’s caller ID when this line calls out.
- ASCII Display (Caller ID): Auto filled by the switch.
- Caller Name: Check this if you want the Caller Name to be displayed on called party’s caller ID.
- Caller Number: Check this if you want the caller’s phone number to be displayed on the called party’s caller id.
CUC Configuration
The Cisco Unity Connection Server needs to have the SIP interface configured to connect to the Cisco Unified Call Manager (CUCM). The following shows an example of the configuration to do this.
Phone System Config
A phone system needs to be created from the Telephony integrations menu, Phone System option to communicate from the CUC to CUCM. The system created needs to be designated as the Default Trap phone system and Enable for Forwarded Message Notification Calls needs to be enabled.
Once the phone system is created, there need to be a port group created and phone system ports created and associated with that group. The number of ports created will be based on the size of the installation and is outside the scope of this document.
While I don’t think it is strictly necessary I enabled Registration with the SIP server so that a true SIP session is established between CUC and CUCM.
Message waiting indicators need to be enabled on the port group.
The phone system ports have to be created and associated with the group.
Press the Add Ports button at the top right and continue with the config. This document does not show that step but only shows the configured ports. In this
The port needs to be configured to be Enabled, point to the proper server, answer calls, perform message notifications, and allow trap connections.
When you have configured all the ports, you should use the link in the upper right corner to Check Telephony Configuration. If both CUC and CUCM are running, this should come back with no errors.
You should also be able to see the Trunk connection as up.
CUC User Configuration
The settings in the User Configuration are all standard. A user needs to be configured for each voice line that needs voicemail.
If all of the above steps are followed, you should have a functioning CUC/CUCM setup with phone service and voice mail.