Avaya Solution & Interoperability Test Lab Application Notes for Avaya 1100- and 1200-Series IP Deskphones R3.2 with Avaya Aura™ Communication Manager R6, Avaya Aura™ Session Manager R6, and Avaya Modular Messaging R5.2 – Issue 1.0 Abstract These Application Notes describe a solution comprised of Avaya Aura™ Communication Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and 1200-Series IP Deskphones with SIP software.
1. Introduction These Application Notes describe a solution comprised of Avaya Aura™ Communication Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and 1200-Series IP Deskphones with SIP software (formerly known as Nortel 1100- and 1200-Series SIP Phones). These telephones were originally developed under the Nortel brand, and as such do not currently support the Avaya Advanced SIP Telephony (AST) protocol implemented in Avaya 9600 Series IP Telephones (SIP).
In general, a SIP telephone originates a call by sending a call request (SIP INVITE message) to Session Manager, which then routes the call over a SIP trunk to Communication Manager for origination services. If the call is destined for another local SIP telephone, Communication Manager routes the call back over the SIP trunk to Session Manager for delivery to the destination SIP telephone. If the call is destined for an H.323 or Digital telephone, then Communication Manager terminates the call directly.
. Calling Features 4.1. Overview Table 2 below shows the calling features successfully tested. Notes on specific feature operations are included in Section 4.2. In addition to basic calling capabilities, the Internet Engineering Task Force (IETF) has defined a supplementary set of calling features in RFC 5359 [13], previously referred to as the SIPPING features. This provides a useful framework to describe product capabilities and compare features supported by various equipment vendors.
Some supported features shown in Table 2 can be invoked by dialing a Feature Name Extension (FNE). Or, a speed dial button on the telephone can be programmed to an FNE. Communication Manager automatically handles many other standard features such as call coverage, trunk selection using Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS), Class Of Service/Class Of Restriction (COS/COR), and voice messaging.
5. Configure Avaya Aura™ Communication Manager This section describes a procedure for setting up a SIP trunk between Communication Manager serving as an Evolution Server and Session Manager. This includes steps for setting up IP codecs, an IP network region, SIP signaling group, SIP trunk group, dial plan, class of service, class of restriction, and call routing. Also, a procedure is described here to configure SIP telephones and features available with OPS in Communication Manager.
2. Proceed to Page 2 of the OPTIONAL FEATURES form. Verify that the number of Maximum Administered SIP Trunks supported by the system is sufficient for the number of SIP trunks needed. If not, contact an authorized Avaya account representative to obtain additional licenses. Note: Each SIP call between two SIP endpoints requires two SIP trunks for the duration of the call. The license file installed on the system controls the maximum permitted.
5.2. IP Codec Set This section describes the steps for administering an IP codec set, which is used in the IP network region when Communication Manager communicates with the SIP telephones via Session Manager. Step Description 1. Enter the change ip-codec-set n command, where n is a number between 1 and 7, inclusive. IP codec sets are used in Section 5.3 for configuring an IP network region to specify which codec sets may be used within and between network regions. For the compliance testing, G.722-64K, G.
5.3. IP Network Region This section describes the steps for administering an IP network region, which is used when Communication Manager communicates with the SIP telephones via Session Manager. Step Description 1. Enter the change ip-network-region n command, where n is a number between 1 and 250 inclusive and configure the following as shown in the display screen below: Authoritative Domain – Set to avaya.com for the sample configuration.
5.5. SIP Signaling Group This section describes the steps for administering a signaling group for communication between Communication Manager and Session Manager. Step Description 1. Enter the command add signaling-group n, where n is an available signaling group and configure the following as shown in the display screen below: Group Type – Set to sip. Transport Method – Set to tls. IMS Enabled – Set to n. Near-end Node Name - Set to procr.
5.6. SIP Trunk Group This section describes the steps for administering a trunk group for communication between Communication Manager and Session Manager. Step Description 1. Issue the command add trunk-group n, where n is an unallocated trunk group and configure the following as shown in the display screen below: Group Type – Set to the Group Type field to sip. Group Name – Enter any descriptive name. TAC (Trunk Access Code) – Set to any available trunk access code. Service Type – Set to tie.
5.7. Define System Features This section describes the steps for administering system wide call features and options related to OPS in Communication Manager. Step Description 1. Use the change system-parameters features command and navigate to Page 18 to administer system wide features for the SIP telephones. Those related to features listed in Table 2 are shown outlined in red. These are all standard Communication Manager features.
5.8. Define the Dial Plan This section describes the steps for administering the dial plan, including overall dial plan format, Feature Access Codes (FACs), and Feature Name Extensions (FNEs). Step Description 1. Use the change dialplan analysis command to define the dial plan formats used in the system. This includes all telephone extensions, Feature Name Extensions (FNEs), and Feature Access Codes (FACs).
3. Use change feature-access-codes to define the access codes for the FNEs highlighted in red. The following screens have been abbreviated to highlight those FACs involved in supporting the FNEs and the AAR FAC.
4. FNEs are defined using the change off-pbx-telephone feature-name-extensions set n command, where n is a number between 1 and 99 and will default to 1 if n is not specified. This command is used to support both SIP telephones and Extension to Cellular. The highlighted fields correspond to those features listed as supported in Table 2. The fields that have been left blank correspond to those more appropriate for Extension to Cellular.
5.9. Specify Class of Service (COS) and Class Of Restriction (COR) This section describes the steps for administering the COS and COR, which affects what calling features and feature options are permitted for defined groups of telephone users. Step Description 1. Use the change cos-group n command, where n is a class of service group number, to set the appropriate service permissions to support the corresponding features (shown in outlined in red). For the sample configuration, COS group 1 was used.
2. Use the change cor n command, where n is a class of restriction number, to enable applicable calling features. To use the Directed Call Pickup feature, the Can Use Directed Call Pickup and Can Be Picked Up By Directed Call Pickup fields must be set to “y” for the affected stations. In the sample configuration, the telephones were assigned to COR 1. Note that Page 4 can be used to implement a form of centralized call screening for groups of stations and trunks.
5.10. SIP Stations This section describes the steps for administering OPS stations in Communication Manager and associating the OPS station extensions with the telephone numbers of the Avaya 1100-and 1200Series IP Deskphones. The configuration is the same for all phones except for the desired number of call appearances as detailed in Step 3. Note that the corresponding users must be configured in Session Manager. There are two methods to sequence these steps: 1.
Step Description 1. Enter the add station n command, where n is an available extension in the dial plan, to administer an OPS station. On Page 1 of the form configure the following fields as shown in the display screen below: Type – Set to 9630SIP. Port – Leave blank. (Once the form is submitted, a virtual port is assigned, e.g., S00022) Name – Enter any descriptive name. Coverage Path – Enter the coverage path number defined for this telephone (e.g., for coverage to voice mail).
2. Proceed to Page 2 of the form. Set MWI Served User Type to sip-adjunct. add station 30043 Page 2 of 6 STATION FEATURE OPTIONS LWC Reception: spe LWC Activation? y Coverage Msg Retrieval? Auto Answer: Data Restriction? Idle Appearance Preference? Bridged Idle Line Preference? CDR Privacy? n Per Button Ring Control? n Bridged Call Alerting? n Active Station Ringing: single H.
4. Enter the change off-pbx-telephone station-mapping command and configure the following as shown in the screen below: Station Extension – Set the extension of the OPS station as configured above. Application – Set to OPS. Phone Number – Enter the number that the SIP telephone will use for registration and call termination. In the sample configuration, the Phone Number is the same as the Station Extension. Trunk Selection – Set to aar.
2. Enter the change route-pattern n command, where n is the route-pattern to be configured, in this case 60. On Page 1 of the form configure the following fields as shown in the screen below: Pattern name – Set to an appropriate name. Grp No – Set to the trunk group being used, in this case 60 (see Section 5.6). FRL – Set to 0 (lowest restriction, or a higher number if appropriate). No. Del Dgts - Set to 0 (all digits are being sent). LAR – Set to next for the first row.
6. Configure Avaya Aura™ Session Manager This section describes the administration of SIP telephones in Session Manager. It is assumed that a trunk has already been provisioned that matches the Communication Manager configuration in Sections 5.5 and 5.6. For additional references in configuring SIP trunking between Communication Manager and Session Manager see References [4-6, 11-12]. The following screens show a sample configuration for an Avaya 1165E IP Deskphone whose extension is 30043.
In the Identity section, enter a Login Name, for example 30043@avaya.com, and the required passwords. Note that the Shared Communication Profile Password is the one the telephone is required to use when registering to Session Manager. It is also recommended to enter the display names. The Localized Display Name is what is displayed on a telephone when a call is made.1 SMGR Login Password, while required, was not used in this sample configuration, and can be any value.
In the Communication Profile section, there are three sub-sections that need to be filled in: Communication Address, Session Manager Profile, and Endpoint Profile. Clicking on the arrow next to Communication Profile reveals the other sections. Click New under Communication Address. FS; Reviewed: SPOC 09/07/2010 Solution & Interoperability Test Lab Application Notes ©2010 Avaya Inc. All Rights Reserved.
Set Type to Avaya SIP, and fill in the extension portion of the Fully Qualified Address, e.g., 30043. Select the SIP domain configured in Session Manager from the drop-down menu to the right of @. In the sample configuration, the domain is avaya.com. Then click Add. This will move the entry to the table as shown in the next screen. FS; Reviewed: SPOC 09/07/2010 Solution & Interoperability Test Lab Application Notes ©2010 Avaya Inc. All Rights Reserved.
Click on the Session Manager Profile checkbox to expand that section. Click on the pull-down menu next to Primary Session Manager, and select the appropriate Session Manager instance from the list. Select the appropriate Origination and Termination Application Sequence. In the sample configuration, these sequences are those associated with the Communication Manager Evolution Server. Select the desired Home Location. The screen below shows what was used for extension 30043.
Click on the Endpoint Profile checkbox to expand that section, and enter the appropriate System, which is the Communication Manager Evolution Server supporting the telephone. Check Use Existing Endpoints if using Method 1 (See Section 5.10), causing Session Manager to use the station previously entered in Communication Manager. Note that leaving this field unchecked will force System Manager to attempt to create the station in Communication Manager, and is used in Method 2.
7. Configure Avaya 1100- and 1200-Series IP Deskphones This section describes the basic configuration of the Avaya 1100- and 1200-Series IP Deskphones. For additional details, see References [8, 9] available at http://www.avaya.com/. Five models were tested: Avaya 1120E, 1140E, 1165E, 1220, and 1230. The configuration was done using configuration files and the local telephone screen interface, as shown in these Application Notes. The steps below show the configuration screens for the 1165E model.
7.2. Configure Local Telephone Features After the configuration file in the previous section has been downloaded, the telephone will attempt to download the files referenced. It will automatically upgrade to the firmware version specified if the firmware files are available at the file server. After that, the telephone will reboot and attempt to download the specified main device configuration and dial plan files.
SERVER_PORT3_2 SERVER_PORT4_1 SERVER_PORT4_2 SERVER_PORT5_1 SERVER_PORT5_2 5060 0 0 5060 5060 #------TCP Port numbers, 0 to disable SERVER_TCP_PORT1_1 0 SERVER_TCP_PORT1_2 0 # TCP is used in the sample configuration SERVER_TCP_PORT2_1 5060 SERVER_TCP_PORT2_2 5060 SERVER_TCP_PORT3_1 0 SERVER_TCP_PORT3_2 0 SERVER_TCP_PORT4_1 5060 SERVER_TCP_PORT4_2 5060 SERVER_TCP_PORT5_1 0 SERVER_TCP_PORT5_2 0 #------TLS Port numbers, 0 to disable, typically 5061 for TLS enabled.
SFTP Y SFTP_READ_PATTERNS *.log, *.
MAX_CALLSUBJECT 5 #------Instant Messaging MAX_IM_ENTRIES 50 IM_MODE ENCRYPTED #------- Enable IM blue LED IM_NOTIFY YES #------Bluetooth ENABLE_BT YES # Local Privacy feature disabled in favor of Calling Number Block FNE # (see Section 4.2.3) DISABLE_PRIVACY_UI Yes #------VQMON configuration -------------VQMON_PUBLISH NO VQMON_PUBLISH_IP 10.1.1.
USB_MEMORY_STICK UNLOCK #------Enable UPDATE method ENABLE_UPDATE YES ENABLE_PRACK YES #------SRTP_MODE can be (BE-2MLines/SecureOnly/BE-Cap Neg) SRTP_ENABLED NO SRTP_MODE BE-2MLines SRTP_CIPHER_1 AES_CM_128_HMAC_SHA1_80 SRTP_CIPHER_2 AES_CM_128_HMAC_SHA1_32 #------Audio Codecs AUDIO_CODEC1 G722 AUDIO_CODEC2 PCMU AUDIO_CODEC3 G729 AUDIO_CODEC4 PCMA AUDIO_CODEC5 AUDIO_CODEC6 AUDIO_CODEC7 AUDIO_CODEC8 G729_ENABLE_ANNEXB YES # G723_ENABLE_ANNEXA YES #------PROXY Checking PROXY_CHECKING YES #------File Manager
#------Login banner LOGIN_BANNER_ENABLE NO #------IPV6 IPV6_ENABLE_GUI NO PREFER_IPV6 NO IPV6_ENABLE NO #------Connection Keep Alive #CONN_KEEP_ALIVE 120 #KEEP_ALIVE_TYPE CRLF #------NAT signaling NAT_SIGNALLING SIP_PING #------Login Notify - Notifies user of previous logins LOGIN_NOTIFY YES LOGIN_NOTIFY_WITH_TIME YES #------Screen Saver & Background image SCRNSVR_ENABLE YES SCRNSVR_UNPRTCTD_ENABLE YES SCRNSVR_UPASS_ENABLE YES SCRNSVR_MODE NO_PASS SCRNSVR_IMAGE screensaver3.
7.3. Configure Local Telephone Dial Plan The telephone will use a local dial plan configuration file to determine when enough digits have been pressed to complete dialing, so that the user need not press an additional key to launch the call. The file is downloaded from the file server at boot time, and was specified as “dialplan.txt” in Section 7.1. An annotated copy of the file used in the sample configuration is shown below.
each individual phone. For mass deployments, Section 7.4.2 shows how the device configuration file and a speed dial list file can be used to support automatic configuration. Note that manually configured buttons will override automatically configured buttons at the same position. See References [8, 9] for more details. 7.4.1. Manual Configuration Steps Description 1. Press the More… soft key twice and select Prefs (not shown). Navigate to Feature Options -> Feature Keys.
Steps Description 3. Access a Communication Manager feature via speed dial button by pressing the appropriate line button. FS; Reviewed: SPOC 09/07/2010 Solution & Interoperability Test Lab Application Notes ©2010 Avaya Inc. All Rights Reserved.
7.4.2. Automatic (Mass) Configuration Steps Description 1. Add the following line to the device configuration file for the corresponding phone type (e.g., 1165DeviceConfig.dat), where SpeedDials.txt will contain the speed dial button configuration data: DEFAULT_CUSTOMKEYSFILE 2. SpeedDials.txt Create the file SpeedDials.txt with an entry for each speed dial button that is to be programmed. Set index to the key position number (see layout for the 1165E in Step 1 in Section 7.4.
8. Verification Steps All features shown in Table 2 were tested using the sample configuration. The following steps can be used to verify and/or troubleshoot installations in the field. Step Description 1. After rebooting the telephone, use the More and Prefs soft keys at the phone to verify that the parameters set in the phone configuration file have been loaded. Verify registration with Session Manager by the appearance of the idle screen.
4. Enter status trunk n, where n is the member in the active state as noted in the previous step for verification of codec used and shuffling status: Codec Type – The codec used for Audio is G.722-64k in this example. Shuffling - If the Near-end and Far-end IP addresses for Audio belong to the Avaya 1100- and/or 1200-Series IP Deskphones and the Audio Connection Type is ip-direct, it signifies that shuffling was successful. In this example, shuffling was successful.
10. Additional References Avaya documentation may be found at http://support.avaya.com/. Avaya Aura™ Session Manager [1] Avaya AuraTM Session Manager Overview, Doc # 03-603323, Issue 2 [2] Administering Avaya AuraTM Session Manager, Doc # 03-603324, Issue 2 [3] Maintaining and Troubleshooting Avaya AuraTM Session Manager, Doc # 03-603325, Issue 2 Avaya Aura™ Communication Manager [4] Administering Avaya Aura™ Communication Manager Server Options, Doc # 03-603479, Issue 2, June 2010.
FS; Reviewed: SPOC 09/07/2010 Solution & Interoperability Test Lab Application Notes ©2010 Avaya Inc. All Rights Reserved.
©2010 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice.
