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Avaya 1100-Series User's Manual

1. e Group Name Enter a descriptive name e TAC Enter an available trunk access code e Direction Enter two way e Outgoing Display Enter y e Service Type Enter tie e Signaling Group Enter the number of the signaling group added in Step 1 e Number of Members Enter the number of members in the SIP trunk must be within the limits for number of SIP trunks specified in Section 3 1 Note once the add trunk group command is submitted trunk members will be automatically generated based on the value in the Number of Members field add trunk group 10 Page T ot Zil TRUNK GROUP Group Number 10 Group Type sip CDR Reports y Group Name SIP trunk to ASM 1 CORGI IN 2 TAC 10 Direction two way Outgoing Display y Dial Access n Night Service Queue Length 0 Service Type tie Mulvian Coyle ia Signaling Group 10 Number of Members 10 On Page 3 fill in the indicated fields as shown below Default values can be used for the remaining fields e Numbering Format Enter private e Show ANSWERED BY on Display Enter y add trunk group 10 Page 3 of 21 TRUNK FEATURES ACA Assignment n Measured none Maintenance Tests y Numbering Format private UUI Treatment service provider Replace Restricted Numbers n aR Replace Unavailable Numbers n Show ANSWERED BY on Display y DJH Reviewed Solution amp Interoperability Test Lab Application Note
2. 2 G 729 n 2 20 E Media Encryption 1 none DJH Reviewed Solution amp Interoperability Test Lab Application Notes 10 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 4 Configure IP Network Region Use the change ip network region n command where n is an available network region Enter the following values and use default values for remaining fields e Authoritative Domain e Name e Codec Set e Intra region IP e Inter region IP Enter the correct SIP domain for the configuration For the sample configuration avaya com was used Enter a descriptive name Enter the number of the IP codec set configured in Section 3 3 IP Direct Audio Enter yes IP Direct Audio Enter yes Name SIP Trunk MEDIA PARAMETERS Codec Set 1 UDP Port Min 204 UDP Port Max 165 change ip network region 1 Page iL IP NETWORK REGION Region 1 HOC cite tne Authoritative Domain avaya com Intra region IP IP Direct Audio Inter region IP IP Direct Audio 8 IP Audio Hairpinning 3 5 Add Node Names and IP Addresses Use the change node names ip command to add the node name and IP Addresses for the procr interface on Communication Manager and the SIP signaling interface of Session Manager if not previously added yes yes For the sample configuration the node name of the SIP Signaling Interface for Session Manager is ASM1 6_1 with an
3. 2011 CM ES 6 0 1 10 80 111 111 11 01 35 PM 06 00 March 15 2011 CM FS 6 0 1 10 80 111 116 11 01 35 PM 06 00 Incremental Completed RO16x 00 1 510 1 10 00 pm TUE MAR 15 2011 Incremental Completed RO16x 00 1 510 1 10 00 pm TUE MAR 15 2011 iv s Select All None O Initialize data for selected devices Incremental Sync data for selected devices Now Schedule Cancel Launch Element Cut Through Click to select Incremental Sync data for selected devices option Click Now to start the synchronization Use the Refresh button in the table header to verify status of the synchronization Verify synchronization successfully completes by verifying the status in the Sync Status column is Completed Note Depending on the number of administration changes made synchronization might take several minutes to complete DJH Reviewed Solution amp Interoperability Test Lab Application Notes 32 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 5 Configure Avaya 1100 Series and 1200 Series IP Deskphones This section describes the basic configuration of the Avaya 1100 Series and 1200 Series IP Deskphones used in the sample configuration For additional information on configuring these types of endpoints see References 8 9 in Section 9 5 1 Configure Initial Network Parameters Network configuration of the telephone can be accomplished either manually or
4. Lime data 16 04 08 TRACE STARTED 03 17 2011 CM Release String cold 00 1 510 1 18777 16 04 17 SIP lt INVITE sip 4441000 avaya com user phone SIP 2 0 GSO silly Suess 2 0 WSO Eemere 16 04 17 dial 4441000 16 04 17 ring station 4441000 cid 0x92 16 04 17 G729A ss off ps 20 wemel 10 60 48 194 e270 comel IO S0 LLL LOS s20o2 Leick xoip options fax Relay modem off tty US uid 0x50001 KOLD Zeg LOO Li LOS 22054 16 04 17 SIP lt PRACK sip 4441000 10 80 111 111 transport tcp SIP 2 On Communication Manager use the SAT command list trace station xxx where xxx is a valid extension number for a SIP telephone For example the trace below illustrates a second call from the same SIP telephone used in the previous trace to the same IP station list trace station 4443120 Page il LIST TRACE time data 16 06 39 TRACE STARTED 03 17 2011 CM Release String cold 00 1 510 1 18777 16 06 44 active station 4443120 cid 0x93 16 06 44 dial 4441000 16 06 44 ring station 4441000 cid 0x93 16 06 44 CI AIAMS Si mt 1988 20 ron 10 80 48 194 2704 Eemol 10 60 11 5 LO SAOSO 16 06 44 xoip options fax Relay modem off tty US uid 0x50001 sation dos LILO Eet IL LOS s20 72 16 06 46 active station 4441000 cid 0x93 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 42 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 6 3 Call Scenarios Verified
5. t 300 d d DIGITMAP 12 digits starting with 9 followed by an initial 1 9 1 x 10 9 1 x 10 amp amp sip Sn user phone amp amp t 300 DIGITMAP 7 Digit Extensions beginning with 444 444x 4 444x 4 amp amp sip n user phone End of Dial Plan DJH Reviewed Solution amp Interoperability Test Lab Application Notes 37 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 6 Verification Steps 6 1 Verify Avaya Aura Session Manager Operational Status Step 1 Verify overall system status of Session Manager Expand Elements gt Session Manager and select Dashboard to verify the overall system status for both Session Managers Specifically verify the status of the following fields as shown below e Tests Pass e Security Module Up e Service State Accept New Service Session Manager Dashboard This page provides the overall status and health summary of each administered Session Manager Session Manager Instances Service State _ Shutdown System As of 4 21 PM 2 Items Refresh Show ALL x Filter Enable Session Tests Security Service Entity Active Call S S S O Manager Type Alarms Pass Module State Monitoring Count ELSA E ASM1 Core 6 0 38 v up oo KT o 7 6 1 0 0 610023 Expand Elements gt Session Manager gt System Status and select Security Module Status to view more detailed status information on the st
6. 1100 Series and 1200 Series IP Deskphones nnnnnnnnnnnnnnnenna 33 5 1 Configure Initial Network Parameter 33 5 2 Configure Local Telephone Features 34 5 3 Gomigure Local Dial E EE 37 Bic Venfication Steps eg tee Eed 38 6 1 Verify Avaya Aura Session Manager Operational Status seeeeseseseeeeeeeeeee 38 6 2 Verify Avaya Aura Communication Manager Operational Gtaius 41 6 3 Call Scenarios Verified EE 43 6 4 te EIER 44 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 2 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 De VAGKOMYING ccsctetia tees tct ttc t nhc te heelen eebe iere 45 EE Te 46 9 Additional EE 47 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 3 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 1 Introduction These Application Notes describe a solution comprised of Avaya Aura Session Manager Avaya Aura Communication Manager Avaya Aura Messaging and Avaya 1100 Series and 1200 Series IP Deskphones with SIP software As shown in Figure 1 Avaya 1100 Series and 1200 Series IP Deskphones configured as SIP endpoints utilize the Avaya Aura Session Manager User Registration feature and are supported by Avaya Aura Communication Manager Evolution Server Since these telephones were originally developed under the Nortel brand they do not currently support the Avaya Advanced SIP Telepho
7. Series IP Deskphones AST Device Notifications Reboot l Reload l Failback As of 12 36 PM 7 Items Refresh Reset Show ALL ze Filter Enable D AST Registered Details Address Login Name SE ec Location IP Address Device e 7 Prim Sec Surv Show 4443000 avaya com 4443000 avaya com Ted Forth Subnet 10 80 48 201 5060 KR mM E o AAV b Earn 10 80 48 x E AC e Si A Cisco Subnet Show 446002 avaya com SIP saan Seene o E o E Show 4443109 avaya com 4443109 avaya com Apple Branch See 192 160 112 101 5060 M Si D o Show _4443115 avaya com__4443115 avaya com__ Pine Branch__ 192 160 192 160 112 102 5060 MW m Oo oO o Oo o Show 4443122 avaya com 4443122 avaya com Oak Branch 192 160 192 160 112 106 5060 KR oi a oO EE i ee Subnet SE Ac Show 4443100 avaya com 4443100 avaya com Olive Branch 432 160 192 160 112 100 5060 KR Ba EL Ip Select All None DJH Reviewed Solution amp Interoperability Test Lab Application Notes 40 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 6 2 Verify Avaya Aura Communication Manager Operational Status Verify the status the SIP trunk group on Communication Manager Evolution Server by using the status trunk n command where n is the trunk group administered in Section 3 6 Verify that all trunks in the trunk group are in the in service idle state as shown below status trunk 10 TRUNK GROUP STA
8. Verification scenarios for the configuration described in these Application Notes included the following call scenarios Basic Calls Place calls from Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager to either digital or IP H 323 stations Answer the call and verify talkpath Place calls from Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager to either digital or IP H 323 stations Answer the call and place the call on Hold Return to the held call and verify talkpath Verify calls can be transferred from Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager to other stations on Communication Manager Verify calls can be forwarded from Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager to other stations on Communication Manager Verify Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager can create conferences with other SIP Deskphones and non SIP stations on Communication Manager Evolution Server Repeat the above scenarios with calls originating from non SIP stations on Communication Manager Evolution Server to Avaya 1100 Series or 1200 Series SIP Deskphones registered to Session Manager Basic Messaging Features Use Pilot Number to access Avaya Aura Messaging and verify Avaya 1100 Series or 1200 Series SIP subscribers are properly recognized and can login without entering their mailbox number Verify
9. Verify y is specified e ARS AAR Dialing without FAC Verify y is specified display system parameters customer options OPTIONAL FEATURES A D Grp Sys List Dialing Start at 01 n Answer Supervision by Call Classifier n ARS y Computer Tel ARS AAR Partitioning y Cio Omue ale ARS AAR Dialing without FAC y ASAI Link Core Capabilities y Step 4 Verify Private Networking feature is Enabled Page 3 ii CAS Main n Change COR by FAC n ephony Adjunct Links y s Redirected Off net y DIGS Beis 2 Si DCS Call Coverage n On Page 5 of display system parameters customer options command verify the Private 66599 Networking feature is set to y display system parameters customer options OPTIONAL FEATURES Processor and System MSP y Processor Ethernet y Port Network Support y Time of Day Routing n Posted Messages n TN2501 VAL Maximum Capacity y Uniform Dialing Plan y Private Networking y Usage Allocation Enhancements y Page ZS emt iil Wideband Switching 3 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 9 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 2 Configure Trunk to Trunk Transfers Use the change system parameters features command to enable trunk to trunk transfers This feature is needed when an incoming call to a SIP station is transferred to a different telephony system
10. the conditions under which calls will be routed to non SIP stations on Communication Manager Evolution Server or to Avaya Aura Messaging Note Since the SIP stations are registered on Session Manager a routing policy does not need to be defined for calls to SIP stations To add a routing policy expand Elements gt Routing and select Routing Policies Click New not shown In the General section enter the following values e Name Enter an identifier to define the routing policy for making calls to non SIP stations on Communication Manager Evolution Server Leave unchecked Enter a brief description Optional e Disabled e Notes In the SIP Entity as Destination section click Select The SIP Entity List page opens not shown e Select the SIP Entity associated with Communication Manager Evolution Server and click Select e The selected SIP Entity displays on the Routing Policy Details page Use default values for remaining fields Click Commit to save Routing Policy definition Note the routing policies defined in this section are examples and were used in the sample configuration Other routing policies may be appropriate for different customer networks The following screen shows the Routing Policy for Communication Manager Evolution Server Routing Pi Home Elements Routing Routing Policies Routing Policy Details Domains Help Locations Routing Policy Details Adaptations SIP Entities G
11. via DHCP Once network configuration is finished configuration files are used to configure other settings To manually configure the telephone access the Device Settings screen on the telephone and enter the appropriate password Enter the appropriate values for IP address mask default gateway file server address and file server access type fields For the sample configuration HTTP was selected as the type of file server When the telephone boots it locates the lt ModelNumber gt SIP cfg file from the root directory of the HTTP server where lt ModelNumber gt is the model number for the specific telephone For example for the 1165E Deskphone the file name would be 1165eSIP cfg This configuration file contains the following three sections e DEVICE_CONFIG Main device configuration file for configuring local features e FW Firmware Release e DIALING PLAN Local dial plan Each section specifies the FILENAME to be accessed and the PROTOCOL to be used for downloading the file from the file server One of the configuration files used in sample configuration for configuring 1165E Deskphone is shown below Note A value of FORCED for the DowNLoaD_mopE for each section ensures explicit control for when files will be downloaded DEVICE_CONFIG DOWNLOAD_MODE FORCED VERSION 000100 PROTOCOL HTTP FILENAME 1165DeviceConfig dat FW DOWNLOAD_MODE FORCED VERSION SIP1165e04 0
12. 0 04 00 PROTOCOL HTTP FILENAME S1TP1165e04 00 04 00 bin DIALING PLAN DOWNLOAD_MODE FORCED VERSION 000020 PROTOCOL HTTP FILENAME dialplan txt DJH Reviewed Solution amp Interoperability Test Lab Application Notes 33 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 5 2 Configure Local Telephone Features After the configuration file in the previous section has been downloaded the telephone will download the files referenced and will automatically upgrade to the firmware version specified After upgrading the firmware the telephone reboots and downloads the specified main device configuration and local dial plan files An annotated copy of the main device configuration file used in the sample configuration is shown below The important parameters their use and the values used for the sample configuration are shown in bold Note the file shown below has been abbreviated for clarity and does not contain many of the default settings SIP Proxy Server Domain information Note Multiple domains can be defined The first domain corresponds to the domain used in the sample configuration and should match the domain configured in Communication Manager and Session Manager SIP_DOMAIN1 avaya com SIP_DOMAIN3 abc com SIP_DOMAIN4 xyz com SIP_DOMAIN5 test com DNS domain Should match domain specified in Section 3 4 DNS_DOMAIN avaya com Specifies Session Manag
13. 120 Page Leg 3 STATIONS WITH OFF PBX TELEPHONE INTEGRATION Application Dial CC Phone Number Trunk Con sug Dual Extension Prefix Selection Set Mode 444 3120 OPS 4443120 aar 1 On Page 2 verify the following fields were correctly populated e Call Limit Verify 3 is assigned Note if more than 3 call appearances were assigned to the station as described in Section 3 11 modify this field to match the number of call appearances e Mapping Mode Verify both is assigned e Calls Allowed Verify all is assigned change off pbx telephone station mapping 4443120 Page 2 Oe 3 STATIONS WITH OFF PBX TELEPHONE INTEGRATION Station Appl Call Mapping Calls Bridged Location Extension Name Limit Mode Allowed SELLS 444 3120 OPS 3 both all none Configuration of Communication Manager is complete Use the save translation command to save these changes Note After making a change on Communication Manager which alters the dial plan or numbering plan synchronization between Communication Manager and System Manager needs to be completed and SIP telephones must be re registered See Section 4 8 for more information on how to perform an on demand synchronization DJH Reviewed Solution amp Interoperability Test Lab Application Notes 20 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 Configure Avaya Aura Session Manager This section provide
14. 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 6 Define Application Sequence The second step in defining an Application to support SIP stations registered to Session Manager is to define an Application Sequence Expand Elements gt Session Manager gt Application Configuration and select Application Sequences from the left navigation menu Click New not shown In the Application Sequence section enter the following values e Name Enter name for the application sequence e Description Enter description Optional In the Available Applications table click icon associated with the Application for Communication Manager Evolution Server defined in Section 4 5 to select this application Verify a new entry is added to the Applications in this Sequence table and the Mandatory column is l as shown below Session Manager H Home Elements Session Manager Application Configuration Application Sequences Application Sequences Dashboard Help See Application Sequence Editor Administration Communication Profile 3 Application Sequence Editor Network Configuration Name CM 6 0 1 Evolution Server Device and Location ZE Description Configuration Application e 8 A S S Applications in this Sequence Configuration Applications 1 Item Sequences Implicit Users SEH tetto iame SIP Entity Description NRS Proxy Users SS EE aag CM _ 6 0 1 Evolution S
15. 5 Advanced Feature Support for Avaya 1100 and 1200 Series IP Deskphones R3 2 with Avaya Aura Communication Manager 6 0 and Avaya Aura Session Manager 6 0 16 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 17 Configuring SIP Trunks among Avaya Aura Session Manager R6 1 Avaya Communication Server 1000E R7 5 and Avaya Aura Messaging R6 0 1 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 47 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 2011 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 The configurations technical data and recommendations provided in these Application Notes are believed to be accurate and dependable but are presented without express or implied warranty Users are responsible for their application of any products specified in these Application Notes Please e mail any questions or comments pertaining to these Application Notes along with the full title name and filename located in the lower right corner directly to the Avaya Solution a
16. AVAYA Avaya Solution amp Interoperability Test Lab Configuring Avaya 1100 Series and 1200 Series IP Deskphones running R4 0 SIP software with Avaya Aura Session Manager Release 6 1 Avaya Aura Communication Manager Release 6 0 1 and Avaya Aura Messaging Release 6 0 1 Issue 1 0 Abstract These Application Notes describe a solution comprised of Avaya Aura Session Manager Avaya Aura Communication Manager Avaya Aura Messaging and Avaya 1100 Series and 1200 Series IP Deskphones with SIP software e Avaya Aura Session Manager provides SIP proxy routing functionality routing SIP sessions across a TCP IP network with centralized routing policies and adaptations to resolve SIP protocol differences across different telephony systems Avaya Aura Communication Manager serves as an Evolution Server within the Avaya Aura Session Manager architecture and supports SIP endpoints registered to Avaya Aura Session Manager and other types of endpoints including Avaya 9600 Series and Avaya 9601 Series IP Deskphones and 2420 Digital Telephones Avaya Aura Messaging provides a centralized voice mail system for all Communication Manager users During testing Avaya 1100 Series and 1200 Series SIP Deskphones successfully registered with Session Manager placed and received calls to and from SIP and non SIP telephones and executed other telephony features such as conference transfer hold and transfer to Avaya Aura Messaging Th
17. IP address of 10 80 111 107 Note The solution is extensible to configurations using CLAN interface For these configurations enter the node name and IP address of the CLAN interface instead of using the procr interface change node names ip Page Leg 2 IP NODE NAMES Name HP ACLLESS ASM1 6_1 10 80 111 107 default BERT procr 10 80 111 111 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 11 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 6 Configure SIP Signaling Group and Trunk Group In the sample configuration trunk group 10 and signaling group 10 were used for connecting to Session Manager Step 1 Add Signaling Group for SIP Trunk Use the add signaling group n command where n is an available signaling group number to create SIP signaling group Enter the following values and use default values for remaining fields e Group Type Enter sip e IMS Enabled Enter n e Transport Method Enter tep e Peer Detection Enabled Enter y e Peer Server Use default value Note default value is replaced with SM after SIP trunk to Session Manager is established e Near end Node Name Enter procr node name from Section 3 5 e Far end Node Name Enter node name for Session Manager defined in Section 3 5 e Near end Listen Port Verify 5060 is used e Far end Listen Port Verify 5060
18. Notes 21 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 1 Define SIP Domain Expand Elements gt Routing and select Domains from the left navigation menu Click New not shown Enter the following values and use default values for remaining fields e Name Enter the Authoritative Domain Name specified in Section 3 4 In the sample configuration avaya com was used e Type Verify SIP is selected e Notes Add a brief description Optional Click Commit to save The screen below shows the SIP Domain defined for the sample configuration AVAYA Avaya Aura System Manager 6 1 Help About Change Password Log off Routing Home Routing Help WEE Domain Management Adaptations SIP Entities Entity Links Time Ranges 1 Item Refresh Filter Enable Routing Policies Name Type Default Notes Dial Patterns avaya com i pn 7 C Regular Expressions Defaults DJH Reviewed Solution amp Interoperability Test Lab Application Notes 22 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 2 Define Locations Locations are used to identify logical and or physical locations where SIP Entities reside for purposes of bandwidth management or location based routing Expand Elements gt Routing and select Locations from the left navigation menu Click New not shown In the General section enter the following values and use default
19. SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 9 Additional References Avaya Product documentation relevant to these Application Notes is available at http support avaya com Avaya Aura Session Manager 1 Avaya Aura Session Manager Overview Doc ID 100068105 2 Installing and Configuring Avaya Aura Session Manager Doc ID 03 603473 3 Avaya Aura Session Manager Case Studies Doc ID 03 603478 4 Maintaining and Troubleshooting Avaya Aura Session Manager Doc ID 03 603325 5 Administering Avaya Aura Session Manager Doc ID 03 603324 Avaya Aura Messaging 6 Administering Avaya Aura Messaging document dated February 2011 7 Implementing Avaya Aura Messaging document dated December 2010 Avaya Aura Communication Manager 8 SIP Support in Avaya Aura Communication Manager Running on Avaya S8xxx Servers Doc ID 555 245 206 9 Administering Avaya Aura Communication Manager Doc ID 03 300509 10 Administering Avaya Aura Communication Manager Server Options Doc ID 03 603479 11 Avaya Extension to Cellular and Off PBX Station OPS Installation and Administration Guide Doc ID 210 100 500 12 Avaya Toll Fraud Security Guide Doc ID 555 025 600 Avaya IP Deskphones SIP 13 SIP Software for Avaya 1100 Series IP Deskphones Administration Release 4 0 NN43170 600 14 SIP Software for Avaya 1200 Series IP Deskphones Administration Release 4 0 NN43170 601 Avaya Application Notes 1
20. TUS Member Port Service State Mtce Connected Ports Busy 0010 001 TO0006 in service idle no 0010 002 TOO0007 in service idle no 0010 003 TO0008 in service idle no 0010 004 T00009 in service idle no 0010 0005 T00014 in service idle no 0010 006 TO0015 in service idle no 0010 007 TO00043 in service idle no 0010 008 T00044 in service idle no 0010 009 TO0045 in service idle no 0010 010 TO0046 in service idle no Verify the status the SIP signaling group by using the status signaling group command where n is the signaling group number administered in Section 3 6 Verify the signaling group is in service as indicated in the Group State field shown below status signaling group 10 STATUS SIGNALING GROUP Ee et TEDE O Active NCA TSC Count Group Type sip AEH CAS LS GmC Otic O Signaling Type facility associated signaling Group State in service DJH Reviewed Solution amp Interoperability Test Lab Application Notes 41 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Use the SAT command list trace tac where is the trunk access code the trunk group defined in Section 3 6 to trace trunk group activity for the SIP trunk between Session Manager and Communication Manager For example the trace below illustrates a call from an Avaya 1100 Series SIP Deskphone using Extension 444 3120 to an IP H 323 station using Extension 444 1000 list trace tac 10 Page il JS DOEN
21. VERY_LEVEL 2 End For more information describing other configuration settings for Avaya 1100 Series and 1200 Series SIP Deskphones see References 13 and 14 in Section 9 DJH Reviewed SPOC 05 05 2011 Solution amp Interoperability Test Lab Application Notes 2011 Avaya Inc All Rights Reserved 36 of 48 11xx12xx_SM6 1 5 3 Configure Local 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 DIALING_PLAN file is downloaded from the file server at boot time as specified in the Configuration file described in Section 5 1 An annotated copy of the local dial plan file used in the sample configuration is shown below In the sample configuration since users dial 444xxxx to call other stations or the Pilot Number for Avaya Aura Messaging an entry was added to local dial plan file This entry corresponds to the dial plan configuration in Communication Manager There is also an entry for long distance dialing using the FAC 9 for ARS routing Note each entry also allows for the telephone user to press the key to indicate that dialing is complete af ZS AY Avaya 1100 Series and 1200 Series IP Deskphone Dial Plan KE SS Domain used in the dialed URL of the SIP INVITE message Sn avaya com
22. agement Manage Users New User Profile Manage Users Public Contacts Shared Addresses New User Profile System Presence ACLs amp status Ae de tity Communication Profile ce Membership Contacts Identity s Middle Name Description Login Name 4443120 avaya com Authentication Type Basic yj Password eeecee Confirm Password eeecee Localized Display Name Example SIP User Endpoint Display Name DJH Reviewed Solution amp Interoperability Test Lab Application Notes 28 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 2 Select Communication Profile tab on New User Profile page and enter numeric value used to logon in the Communication Profile Password and Confirm Password fields Note Password should match the Security Code field defined in Section 3 11 Verify there is a default entry identified as the Primary profile as shown below New User Profile Identity z Communication Profile Membership Contacts Communication Profile Communication Profile Password eeeesee Confirm Password eegen Primary Select None Name Primary Default If an entry does not exist select New and enter values for the following required attributes e Name Enter Primary e Default Verify M Step 3 In the Communication Address sub section select New to define a Communication Address for the new SIP user Enter values for the follow
23. atus of Security Module for the Session Manager Verify the Status column displays Up as shown below Security Module Status This page allows you to view the status of each Session Manager s Security Module and to perform certain actions Reset Synchronize Update Installed Certificates Connection Status 2Items Refresh Show ALL E Filter Enable Z Session a Default Entity Links Details EE Type Status Connections IP Address VLAN ese NIC Bonding ted actual Show ASM1 SM Up 33 10 80 111 107 24 10 80 111 1 Disabled 9 9 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 38 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 2 Verify status of the SIP Trunks between Session Manager and either Communication Manager or Avaya Aura Messaging Expand Elements gt Session Manager gt System Status and select SIP Entity Monitoring to view more detailed status information for one of the SIP Entity Links Select the SIP Entity for Communication Manager Evolution Server from the All Monitored SIP Entities table to open the SIP Entity Entity Link Connection Status page In the All Entity Links to SIP Entity CM ES 6 0 1 table verify the Conn Status for link is Up Click Show in the Details column to view additional status information for the selected link as shown below SIP Entity Entity Link Connection Status This page displays detailed connection status
24. ay name for user e Security Code Enter the number used to log in station Note this number should match the Communication Profile Password field defined when adding this user in System Manager See Section 4 7 for more information e Coverage Path 1 Enter the coverage path number previously defined for coverage to Avaya Aura Messaging add station 4443120 Page lt 6 STATION Extension 444 3120 Lock Messages n BCS 0 Type 9630SIP Security Code 123456 TENS Ik Port Coverage Path 1 1 COR 2 Name SIP Station User Coverage Path 2 COS el EE STATION OPTIONS Time of Day Lock Table Loss Group 19 Message Lamp Ext 666 4029 Display Language english Button Modules 0 Survivable COR internal Survivable Trunk Dest y IP SoftPhone n IP Video n On Page 2 enter the following values and use defaults for remaining fields e MWI Served User Type Enter sip adjunct add station 4443120 Page 2 On 6 STATION FEATURE OPTIONS Bl sO Eeer ia Per Station CPN Send Calling Number y EC500 State enabled MWI Served User Type sip adjunct DJH Reviewed Solution amp Interoperability Test Lab Application Notes 18 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 On Page 4 add the desired number of call appr entries in the BUTTON ASSIGNMENTS section This governs how many concurrent calls can be supported Avaya 1100 Series IP Deskphones have the capability of ha
25. calls between Avaya 1100 Series or 1200 Series SIP subscribers are forwarded to the correct Avaya Aura Messaging mailbox in both No Answer and Busy conditions Verify calls between Avaya 1100 Series or 1200 Series SIP subscribers are successfully forwarded to Avaya Aura Messaging and the correct Personal Greetings are played in both No Answer and Busy conditions Verify Avaya 1100 Series or 1200 Series SIP subscribers can leave voice mail messages for other subscribers Verify Avaya Aura Messaging sends appropriate Message Waiting Notification messages when Avaya 1100 Series or 1200 Series SIP subscribers leave or retrieve messages Supplemental Features Use Auto Attendant Number to access Avaya Aura Messaging and verify Avaya Aura Messaging can successfully transfer calling party to correct Avaya 1100 Series or 1200 Series SIP subscriber When Reach Me is activated for Avaya 1100 Series or 1200 Series SIP subscribers verify Avaya Aura Messaging can successfully call the Reach Me destination After subscriber accepts call verify calling party is connected to subscriber DJH Reviewed Solution amp Interoperability Test Lab Application Notes 43 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 e Verify Avaya 1100 Series or 1200 Series SIP subscribers could use Reply Forward and Call Sender features with other subscribers e Verify Avaya Aura Messaging sends appropriate Message Waiting Notification me
26. d verify the limit specified for number of Maximum Off PBX Telephones OPS is sufficient as shown below display system parameters customer options Page Le iil OPTIONAL FEATURES G3 Version V16 Software Package Enterpris t 2 EE TD Sens al USED Platform Maximum Ports 6400 45 Maximum Stations 2400 2 Maximum Off PBX Telephones TOSCO SIEA 0 Maximum Off PBX Telephones OPS 9600 8 Maximum Off PBX Telephones PBFMC 9600 0 Step 2 Verify SIP Trunk Capacity is sufficient for the expected number of calls On Page 2 of the display system parameters customer options command verify the limit specified for number of Maximum Administered SIP Trunks is sufficient as shown below display system parameters customer options Page Ze iil OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H 323 Trunks 4000 0 Maximum Concurrently Registered IP Stations 2400 Maximum Administered Remote Office Trunks 4000 J S Maximum Video Capable IP Softphones 2400 1 Maximum Administered SIP Trunks 4000 10 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 8 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 3 Verify AAR ARS Routing features are Enabled on system On Page 3 of system parameters customer options command verify the following features are enabled e ARS Verify y is specified e ARS AAR Partitioning
27. eneral Time Ranges bled Dial Patterns Regular Expressions E S A SIP Entity as Destination Defaults Name FQDN or IP Address Type Notes CM ES 6 0 1 10 80 111 111 CH Evolution Srvr 6 0 1 Time of Day Add Remove View Gaps Overlaps 1Item Refresh Filter Enable 0O Ranking 1 Name 2 Mon Tue Wed Thu Fri Sat Sun Start Time End Time Notes o 24 7 00 00 23 59 Time Range 24 7 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 24 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 4 Define Dial Pattern This section describes the steps to define a dial pattern to route calls to non SIP stations on Communication Manager Evolution Server In the sample configuration 7 digit extensions beginning with 4441 are assigned to non SIP stations on Communication Manager Note Since the SIP stations are registered on Session Manager a dial pattern does not need to be defined for SIP stations supported by Communication Manager Evolution Server To define a dial pattern expand Elements gt Routing and select Dial Patterns Click New not shown In the General section enter the following values and use default values for remaining fields e Pattern Add dial pattern associated with non SIP stations e Min Enter the minimum number digits that must to be dialed e Max Enter the maximum number digits that may be dialed e SIP Domain Select the SIP Domain defined in Section 4 1 e Notes Enter a brie
28. er as the SIP registrar for domain avaya com A second address parameter could be specified as a backup registrar for failover not tested SERVER_IP1_1 10 80 111 107 SERVER_IP1_2 10 80 111 107 UDP Port numbers Note UDP was not used in the sample configuration SERVER_PORT1_1 5060 SERVER_PORT1_2 5060 SERVER_PORT2_1 5060 SERVER_PORT2_2 5060 TCP Port numbers enter 0 to disable TCP is used in the sample configuration SERVER_TCP_PORT1_1 5060 SERVER_TCP_PORT1_2 5060 SERVER_TCP_PORT2_1 0 SERVER_TCP_PORT2_2 0 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 34 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 en TLS Port numbers 0 to disable If enabled 5061 is typically used Note TLS was not used in the sample configuration SERVER_TLS_PORT1_1 0 SERVER_TLS_PORT1_2 0 S S ERVER_TLS_PORT2_1 0 ERVER_TLS_PORT2_2 0 Listening ports SIP_UDP_PORT 5060 SIP_TCP_PORT 5060 SIB PLS PORT 0 _RETRIES1 3 ERVER_RETRIES2 3 RETRIES3 3 Server retries SERVER S S ERVER Recovery amp Log levels RECOVERY_LEVEL 2 LOG_LEVEL 255 Firmware update AUTO_UPDATE YES AUTO_UPDATE_TIME 0 Service Package Not supported in this con
29. erver CM ES 6 0 1 System Tools Select All None Available Applications 2Items Refresh Filter Enable Name P Entity Description Note The Application Sequence defined for Communication Manager Evolution Server can only contain a single Application Click Commit to save the new Application Sequence DJH Reviewed Solution amp Interoperability Test Lab Application Notes 27 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 7 Add SIP Users Add new SIP users for each Avaya 1100 Series or 1200 Series SIP station defined in Section 3 11 Alternatively use the option to automatically generate the SIP station after adding a new SIP user To add new SIP users expand Users gt User Management and select Manage Users Step 1 Click New not shown Enter values for the following required attributes for a new SIP user in the Identity section and use default values for remaining fields e Last Name Enter last name of user e First Name Enter first name of user e Login Name Enter extension number lt domain gt where lt domain gt matches domain defined in Section 4 1 e Authentication Type Verify Basic is selected e Password Enter password to be used to log into System Manager e Confirm Password Repeat value entered above e Localized Display Name Enter display name for user The screen below shows results from Step 1 for a new SIP user User Management Dome Users User Man
30. ese Application Notes provide information for the setup configuration and verification of the call flows tested on this solution DJH Reviewed Solution amp Interoperability Test Lab Application Notes 1 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Table of Contents 1 vleit 4 2 Equipment and Software Valdated 6 3 Configure Avaya Aura Communication Manager 7 3 1 Verify System Capacities and Licensimg ANEN 7 3 2 Configure Trunk to Trunk Transfers EE 10 3 3 Configure IP Codec EE 10 3 4 Configure IP Network Region sssssssssssssssssnnnnnnnseserrrrrnnnrnssserrrnrnnnnnnsserrnnrnnnnnnenet 11 3 5 Add Node Names and IP Addresses EE 11 3 6 Configure SIP Signaling Group and Trunk Group 12 3 7 Configure Route Pattern aires ci teins fd deesce anit EE 14 3 8 Administer Numbering EE 15 3 9 Administer Locations EEN 16 3 10 Administer AAR Digit Analysis seco ee oie ee ee he 17 3 We ele E en E 17 3 12 Verify Off PBX Telephone Station Mapping ssssssssseneesessserrrnrrnesserrrrrnnnnnesee 20 4 Configure Avaya Aura Session Manager 21 4 1 DEMME SIP Domas nenin nen EES eg 22 42 Define Be le EE 23 4 3 Define GEIER Aon acon sin eA e 24 4 4 Define Dial FS ett Seet tei Rate et See ee 25 4 5 Define Applcaton ee 26 4 6 Define Application Sequence asec cena ainauenaiienann aman 27 APs Add SIP EE 28 4 8 Synchronize Changes with Avaya Aura Communication Manager 32 5 Configure Avaya
31. extension on Communication Manager SIP telephones register with Session Manager and use Communication Manager for call origination and termination services This section describes the administration of Communication Manager Evolution Server using a System Access Terminal SAT Some administration screens have been abbreviated for clarity The following administration steps will be described Verify System Capacities and Communication Manager Licensing Configure Trunk to trunk Transfers Configure IP Codec Set Configure IP Network Region Configure IP Node Names and IP Addresses Configure SIP Signaling Groups and Trunk Groups Configure Route Pattern Administer Numbering Plan Administer Locations Administer AAR Analysis Configure Stations After completing these steps the save translation command should be performed 3 1 Verify System Capacities and Licensing This section describes the procedures to verify the correct system capacities and licensing have been configured If there is insufficient capacity or a required features is not available contact an authorized Avaya sales representative to make the appropriate changes DJH Reviewed Solution amp Interoperability Test Lab Application Notes 7 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 1 Verify Off PBX Telephone Capacity is sufficient for the expected number of endpoints On Page 1 of the display system parameters customer options comman
32. f description Optional In the Originating Locations and Routing Policies section click Add The Originating Locations and Routing Policy List page opens not shown e In Originating Locations table select ALL e In Routing Policies table select the appropriate Routing Policy defined for Communication Manager Evolution Server in Section 4 3 e Click Select to save these changes and return to Dial Pattern Details page Click Commit to save the new definition The following screen shows the Dial Pattern defined for routing calls to non SIP stations on Communication Manager Evolution Server Routing Pu Home Elements Routing Dial Patterns Dial Pattern Details Domains Help Locations Dial Pattern Details Adaptations SIP Entities General Entity Links Pattern 4441 Time Ranges Min Routing Policies ge Emergency Call Regular Expressions Defaults SIP Domain ALL v Notes to CM ES stations Originating Locations and Routing Policies 1Item Refresh Filter Enable Routing Policy Notes Select All None DJH Reviewed Solution amp Interoperability Test Lab Application Notes 25 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 5 Define Application To support SIP stations registered to Session Manager an Application must be defined for Communication Manager Evolution Server To define a new Application expand Elements gt Session Manager gt Appl
33. figuration ENABLE_SERVICE_PACKAGE NO Service Package http or https SERVICE_PACKAGE_ PROTOCOL HTTP sano se ee Banner FORCE _BANNER YES BANNER Avaya Fb Autologin AUTOLOGIN_ENABLE YES LEE Enable Disable SIP ping SIP_PING YES VMAIL Settings Voice mail extension dialed when messages button is pressed Enter Pilot Number for Avaya Aura Messaging VMAIL 4445000 VMATL_DELAY 600 Specify Transfer Hold and Conference settings TRANSFER TYPE STANDARD HOLD TYPE RFC3261 ENABLE _3WAY_CALL YES REDIRECT TYPE RFC3261 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 35 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Maximum MAX LOGINS 5 number of Multi user logins en Enable UPDATE method ENABLE _ UPDATE YES ENABLE_PRACK YES Fb PROXY Checking PROXY_CHECKING YES Zen File Manager FM_CONFIG_ENABLE YES FM_CERTS_ENABLE YES FM _CONFIG_ENABLE YES Local Storage Limits MAX_INBOX_ENTRIES 100 MAX_OUTBOX_ENTRIES 100 MAX _REJECTREASONS 20 MAX_CALLSUBJECT 20 MAX_PRESENCENOTE 20 MAX_IM_ENTRIES 999 MAX_ADDR_BOOK_ENTRIES 100 tereta Session Timer Setttings SESSION_TIMER_ENABLE NO RECO
34. for all entity links from all Session Manager instances to a single SIP entity All Entity Links to SIP Entity CM ES 6 0 1 1 Item Refresh Filter Enable Details Session Manager Name SIP Entity Resolved IP Port Proto Conn Status Reason Code Link Status Hide ASM1 10 80 111 111 5060 TCP Up 200 OK Up Time Last Down Time Last Up Last Message Sent Last Message Response Last Response Latency ms Mar 3 2011 3 57 50 PM MST Mar 4 2011 9 48 32 AM MST Mar 17 2011 4 03 00 PM MDT 9 Repeat the steps to verify the Entity Link status for SIP Trunk to Avaya Aura Messaging DJH Reviewed Solution amp Interoperability Test Lab Application Notes 39 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 3 Verify Registrations of SIP Endpoints Expand Elements gt Session Manager System Status and select User Registrations to verify the SIP endpoints have successfully registered with Session Manager For example the screen below highlights an Avaya 1100 Series SIP Deskphone successfully registered to Session Manager Note As previously mentioned Avaya 1100 Series and 1200 Series SIP Deskphones do not currently support the Avaya Advanced SIP Telephony AST protocol However Communication Manager and Session Manager have the capability to extend some advanced telephony features to non AST telephones See References 15 and 16 in Section 9 for more information on configuring these features on Avaya 1100 Series and 1200
35. for the appropriate subscriber Avaya Aura Session Manager is managed by Avaya Aura System Manager For the sample configuration Avaya Aura System Manager and Avaya Aura Session Manager each run on an Avaya S8800 Server Avaya Aura Communication Manager Evolution Server runs on an Avaya S8800 server with an Avaya G650 Media Gateway DJH Reviewed Solution amp Interoperability Test Lab Application Notes 4 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Avaya Core Site Avaya Aura Communication Manager R6 0 1 Avaya S8800 Server w We CM 10 80 111 111 Avaya Digital Solution and Telephones Interoperability Test Lab Avaya 96xx A ZC H 323 Deskphones Avaya 96x1 Avaya 11xx 12xx H 323 Deskphones SIP Deskphones mm zm e mm om mm S ll ee eee eee Sal Avaya Aura Messaging R6 0 1 Avaya Aura Session Manager R6 1 10 80 111 102 SIP Signaling 10 80 111 107 Ethernet sey rer td daa Manager R6 1 Domain avaya com Other Servers Active Directory ATE 10 80 111 30 SSS ons Data Center Figure 1 Sample Configuration In general a SIP endpoint 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 SIP endpoint Communication Manager routes the call back over the SIP trunk to Session Manager for delivery to the dest
36. ication Configuration and select Applications from the left navigational menu Click New not shown In the Application Editor section enter the following values e Name Enter name for the application e SIP Entity Select SIP Entity associated with Communication Manager Evolution Server e CM System for SIP Entity Select name of Managed Element associated with Communication Manager In the sample configuration CM ES 6 0 1 was used e Description Enter description Optional Leave fields in the Application Attributes optional section blank Click Commit to save application The screen below shows the Application defined for Communication Manager Evolution Server in the sample configuration Session Manager UH Home Elements Session Manager Application Configuration Applications Applications Dashboard Session M e Center Application Editor Administration Communication Profile Application Editor PP Network Configuration Name CM_6 0 1 Evolution Server Device and Location GE SIP Entity CM ES 6 0 1 x Application CM System View Add for SIP CM ES 6 0 1 Refresh CM Configuration Entity Systems Application Sequences Application Attributes optional Implicit Users NRS Proxy Users Name Valua System Status Application Handle URI Parameters System Tools DJH Reviewed Solution amp Interoperability Test Lab Application Notes 26 of
37. ination SIP endpoint If the call is destined for an H 323 or Digital telephone Communication Manager terminates the call directly These Application Notes focus on the configuration of the SIP endpoints SIP trunks and call routing These Application Notes assume Avaya Aura Messaging Communication Manager and Session Manager are already installed and basic configuration steps have been performed Only steps relevant to configuration of SIP endpoints will be described in this document For further details on configuration steps not covered in this document consult the appropriate document in Section 9 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 5 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 2 Equipment and Software Validated The following equipment and software were used for the sample configuration Equipment Software Firmware Avaya Aura Session Manager on an Avaya S8800 Release 6 1 server Build 6 1 0 0 610023 Avaya Aura System Manager Release 6 1 on Avaya S8800 server Version 6 1 0 4 5072 6 1 4 11 Avaya Aura Messaging on an Avaya S8800 server Release 6 0 1 Version 6 0 1 8 0 Avaya Aura Communication Manager Evolution Release 6 0 1 SP1 Server Version R16x 00 1 510 1 18777 e 1100 Series and 1200 Series IP Deskphone SIP FW R4 00 04 e Digital Telephones DCP N A e 9600 Series IP Deskphone H 323 FW R3 1 SP1 e 9601 Series IP De
38. ing required attributes e Type Select Avaya SIP from drop down menu e Fully Qualified Address Enter same extension number as used for Login Name in Step 1 Note value is shown in Handle field after address is added e Domain Verify Domain matches Domain name defined in Section 4 1 Click Add not shown to save the Communication Address for the new SIP user The screen below shows results from Step 3 Communication Address New Type Handle Domain C Avaya SIP 4443120 avaya com Select All None DJH Reviewed Solution amp Interoperability Test Lab Application Notes 29 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 4 In the Session Manager Profile section enter to expand section Enter values for the following fields e Primary Session Manager Select Session Manager e Secondary Session Manager Select None from drop down menu e Origination Application Sequence Select Application Sequence defined in Section 4 6 for Communication Manager e Termination Application Sequence Select Application Sequence defined in Section 4 6 for Communication Manager e Survivability Server Select None from drop down menu e Home Location Select Location defined in Section 4 2 The screen below shows results from Step 4 V Session Manager Profile Primary Secondary Maximum Primary Session Manager 9 0 9 e Primary Secondary Maximum Secondary Session Manage
39. is used e Far end Network Region Enter network region defined in Section 3 4 e Far end Domain Enter domain name for Authoritative Domain field defined in Section 3 4 e DTMF over IP Verify rtp payload is used Note TCP was used for the sample configuration However TLS would typically be used in production environments add signaling group 10 Page iL Or 1 SIGNALING GROUP Group Number 10 Group Type sip IMS Enabled n Transport Method tcp CHS 2 io SP Enabled hse 2a mn IP Video n Peer Detection Enabled y Peer Server Others Near end Node Name procr Far end Node Name ASM1 6_1 Near end Listen Port 5060 Far end Listen Port 5060 Far end Network Region 1 Far end Domain avaya com Bypass If IP Threshold Exceeded n DTMF over IP rtp payload Drrece EE EE EE Session Establishment Timer min 3 IP Audio Hairpinning n Enable Layer 3 Test n Direct IP IP Early Media o H 323 Station Outgoing Direct Media n Alternate Route Timer sec 6 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 12 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 2 Add SIP Trunk Group Add the corresponding trunk group controlled by the signaling group defined Step 1 using the add trunk group n command where n is an available trunk group number Fill in the indicated fields as shown below Default values can be used for the remaining fields e Group Type Enter sip
40. l Sender or Auto Attendant Use the change locations command to identify a default proxy route Set the Proxy Rte field to use the Route Pattern defined in Section 3 7 change locations Page Lor 16 LOCATIONS ARS Prefix 1 Required For 10 Digit NANP Calls y Loc Name Timezone Rule NPA ARS Atd Disp Prefix Proxy Sel No EE E Parm Rte Pat eee een 00 00 0 il 10 Ze 3 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 16 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 10 Administer AAR Digit Analysis This section provides the configuration of the AAR Automatic Alternate Routing pattern used in the sample configuration for routing calls between SIP users and other stations on Communication Manager Evolution Server In the sample configuration extension numbers starting with digits 444 3xxx are assigned to SIP stations supported by Communication Manager Evolution Server Note Other methods of routing may be used Use the change aar analysis n command where n is the first digit of the extension numbers used for SIP stations in the system Fill in the indicated fields as shown below and use default values for remaining fields e Dialed String Enter leading digit s of extension numbers assigned to SIP Stations e Min Enter minimum number of digits that must be dialed e Max Enter maximum number of digits that may be dialed e R
41. mp Interoperability Test Lab at interoplabnotes list avaya com DJH Reviewed Solution amp Interoperability Test Lab Application Notes 48 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1
42. ndling 11 call appearances while 1200 Series can handle 10 call appearances In the sample configuration three call appearances were configured here to support conferencing scenarios Note Avaya 1100 Series IP Deskphones display only one local call appearance button when idle So the number of entries shown below is not required to match the number of appearances displayed on the telephone add station 4443120 Page A or 6 STATION BUTTON ASSIGNMENTS 1 call appr 2 call appr 3 call appr 4 Or en CO On Page 6 enter the following values and use defaults for remaining fields e SIP Trunk Enter aar to use Route Pattern defined in Section 3 7 add station 4443120 Page Dog STATION SIP FEATURE OPTIONS Type of 3PCC Enabled None SIP Trunk aar DJH Reviewed Solution amp Interoperability Test Lab Application Notes 19 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 12 Verify Off PBX Telephone Station Mapping Use the change off pbx telephone station mapping xxx command where xxx is an extension assigned to an 1100 Series or 1200 Series SIP telephone to verify an Off PBX station mapping was automatically created for the SIP station On Page 1 verify the following fields were correctly populated e Application Verify OPS is assigned e Trunk Selection Verify aar is assigned change off pbx telephone station mapping 4443
43. nt Editor Template DEFAULT_9630SIP_CM_6_0 Nj Set Type Security Code seess Port Qup Voice Mail Number 4445000 Delete Endpoint on Unassign of Endpoint from User or on Delete User Click Commit not shown to save definition of new user DJH Reviewed Solution amp Interoperability Test Lab Application Notes 31 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 8 Synchronize Changes with Avaya Aura Communication Manager After completing these changes in System Manager perform an on demand synchronization Expand Elements gt Inventory gt Synchronization and select Communication System On the Synchronize CM Data and Configure Options page expand the Synchronize CM Data Launch Element Cut Through table and select the row associated with Communication Manager Evolution Server as shown below Ui Home Elements Inventory Synchronization Communication System Synchronize CM Data and Configure Options Inventory Manage Elements Help Discovered Inventory Synchronize CM Data and Configure Options Discovery Management Synchronization Synchronize CM Data Launch Element Cut Through Configuration Options Communication Expand All Collapse All Synchronize CM Data Launch Element Cut Through Messaging System 2 Items Refresh Show ALL Filter Enable gO Element Name FQDN IP Address Last Sync Time Last Translation Time Sync Type Sync Status Location Software Version March 15
44. ny AST protocol implemented in Avaya 9600 Series or Avaya 9601 Series SIP Deskphones However Communication Manager and Session Manager have the capability to extend some advanced telephony features to non AST telephones See References 15 and 16 in Section 9 for more information on configuring these features on Avaya 1100 Series and 1200 Series IP Deskphones Note although Avaya 1100 Series and 1200 Series IP Deskphones support the ability to failover to a secondary SIP Registrar this functionality was not tested in the sample configuration and will not be described in these Application Notes Avaya Aura Communication Manager Evolution Server supports Avaya 2420 Digital telephones and Avaya 9600 Series IP Deskphones and is connected over a SIP trunk to Avaya Aura Session Manager Release 6 1 using the SIP Signaling network interface on Session Manager Avaya Aura Messaging consists of an Avaya Aura Messaging Application Server MAS and Avaya Message Storage Server MSS running on a single Avaya S8800 server Avaya Aura Messaging is also connected over a SIP trunk to Avaya Aura Session Manager All inter system calls are carried over these SIP trunks All users have mailboxes defined on Avaya Aura Messaging which they access via a dedicated pilot number Interoperability testing included verifying calls between stations were re directed to Avaya Aura Messaging and the calling party was able to leave a voice mail message
45. oute Pattern Enter Route Pattern defined in Section 3 7 e Call Type Enter unku change aar analysis 6 Page Le 2 AAR DIGIT ANALYSIS TABLE Location all Percent Full al Dialed Torani Route Gage Node ANI SC ELANG Min Max Pattern Type Num Reqd 443 7 7 10 unku n 4443 7 7 10 unku n 4445 H 7 10 unku n 778 7 y 1O unku n 3 11 Configure Stations For each SIP user defined in Session Manager add a corresponding station on Communication Manager The extension number defined for the SIP station will be the login ID the user enters to register to Session Manager The configuration is the same for all of the 1100 Series or 1200 Series IP Deskphones except for the desired number of call appearances Note Instead of manually defining each station using the Communication Manager SAT interface an alternative option is to automatically generate the SIP station when adding a new SIP user using System Manager See Section 4 7 for more information on adding SIP users DJH Reviewed Solution amp Interoperability Test Lab Application Notes 17 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Use the add station n command where n is a valid extension number defined in the system On Page 1 enter the following values and use defaults for remaining fields e Phone Type Enter 9630SIP e Port Leave blank Once the command is submitted a virtual port will be assigned e g S0000 e Name Enter a displ
46. r Origination Application Sequence CM 6 0 1 Evolution Server Termination Application Sequence CM 6 0 1 Evolution Server Survivability Server si Home Location 192 160 Subnet vi DJH Reviewed Solution amp Interoperability Test Lab Application Notes 30 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 5 In the Endpoint Profile section enter to expand section Enter values for the following fields e System Select Managed Element associated with Communication Manager Evolution Server e Use Existing Endpoints Leave unchecked to automatically create new endpoint when new user is created Else enter if endpoint was already defined in Section 3 11 e Extension Enter same extension number used for Login Name in Step 1 e Template Select DEFAULT_9630SIP_CM_6_0 e Security Code Enter numeric value used to log on to SIP telephone Note this field must match the value entered for the Communication Profile Password field in Step 2 e Port Select IP from drop down menu e Voice Mail Number Enter Pilot Number for Avaya Aura Messaging e Delete Station on Unassign of Endpoint Enter to automatically delete station when Endpoint Profile is un assigned from user The screen below shows the results from Step 5 when adding a new SIP user in the sample configuration Endpoint Profile System Profile Type Use Existing Endpoints Extension Q 4443120 Endpoi
47. rface SIL Solution Interoperability and Test Lab SIP Session Initiation Protocol SM Avaya Aura Session Manager SMGR System Manager used to configure Session Manager SSH Secure Shell SSL Secure Socket Layer TAC Trunk Access Code Communication Manager Trunk Access TCP Transmission Control Protocol TCP IP Transmission Control Protocol Internet Protocol TLS Transport Layer Security URL Uniform Resource Locator DJH Reviewed Solution amp Interoperability Test Lab Application Notes 45 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 8 Conclusion These Application Notes describe how to configure Avaya Aura Session Manager and Avaya Aura Communication Manager Evolution Server to support Avaya 1100 Series or 1200 Series SIP Deskphones Interoperability testing included making bi directional calls between SIP telephones and other types of stations on Communication Manager Evolution Server In addition various features including hold transfer and conference were tested Interoperability testing also included verification that calls from Avaya 1100 Series or 1200 Series SIP subscribers were successfully forwarded to Avaya Aura Messaging in both busy and no answer scenarios and Avaya 1100 Series or 1200 Series SIP subscribers could use supplemental Avaya Aura Messaging features such as Auto Attendant and Reach Me DJH Reviewed Solution amp Interoperability Test Lab Application Notes 46 of 48
48. s 13 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 On Page 4 fill in the indicated fields as shown below Default values can be used for the remaining fields e Support Request History Enter y e Telephone Event Payload Type Enter 120 add trunk group 10 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone y Prepend to Calling Number n Send Transferring Party Information n Network Call Redirection n Send Diversion Header n Support Request History y Telephone Event Payload Type 120 3 7 Configure Route Pattern This section provides the configuration of the route pattern used in the sample configuration for routing calls between SIP stations and other stations supported by Communication Manager Evolution Server Use change route pattern n command where n is an available route pattern Fill in the indicated fields as shown below and use default values for remaining fields e Grp No Enter a row for the trunk group defined in Section 3 6 e FRL Enter 0 e Numbering Format Enter lev0 pvt In the sample configuration route pattern 10 was created as shown below change route pattern 10 Page Leg 3 Pattern Number 10 Pattern Name SIP to ASM SCCAN n Secure SIP n Gro Hine PAS El HO Osh ONO Eeer DES Exe No Mrk Lmt List Del Digits OSIG Dgts Intw 1 10 0 n user Ae n user Be n user BC CV Ali Ohaus Ca CAS oe ITC BCIE Service Fea
49. s the procedures for configuring Avaya Aura Session Manager to support registrations of SIP endpoints These instructions assume other administration activities have already been completed such as defining the SIP entities for Avaya Aura Messaging Avaya Aura Communication Manager and Session Manager defining the network connection between System Manager and Session Manager defining Communication Manager as a Managed Element and defining the Entity Links for the SIP trunks between each SIP entity and Session Manager For more information on configuring SIP Trunks see Reference 17 in Section 9 For more information on other aspects of administering Session Manager see References 2 through 5 in Section 9 The following administration activities will be described e Define SIP Domain and Locations Define Routing Policies and Dial Patterns which control routing between SIP Entities Define Applications and Application Sequences supporting SIP Users Add new SIP Users Synchronize changes with Avaya Aura Communication Manager Note Some administration screens have been abbreviated for clarity Configuration is accomplished by accessing the browser based GUI of Avaya Aura System Manager using the URL http lt ip address gt SMGR where lt ip address gt is the IP address of Avaya Aura System Manager Log in with the appropriate credentials DJH Reviewed Solution amp Interoperability Test Lab Application
50. skphone H 323 FW R6 0 SP1 Table 1 Equipment and Software Firmware Note Avaya 9608 and 9641G IP Deskphones H 323 were tested in the sample configuration Avaya 9601 IP Deskphone was not tested since this device does not support H 323 protocol Note The following field updates were also installed on Avaya Aura Messaging See http support avaya com for more information on installing these field updates o W16007rf ab o C16007rf ad o A14007rf ac o M6104rf ab DJH Reviewed Solution amp Interoperability Test Lab Application Notes 6 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 Configure Avaya Aura Communication Manager This section describes the steps needed to configure the SIP trunk between Communication Manager and Session Manager to support calls between SIP telephones and other stations on Communication Manager These instructions assume the G450 Media Server is already configured on Communication Manager For information on how to administer these other aspects of Communication Manager see References 8 through 12 in Section 9 Avaya and third party SIP telephones are configured as Off PBX Stations OPS in Communication Manager Communication Manager does not directly control an OPS endpoint but its features and calling privileges can be applied by associating a local extension with the OPS endpoint Similarly a SIP telephone in Session Manager is associated with an
51. ssages when Avaya 1100 Series or 1200 Series SIP subscribers use Reply or Forward features e Verify Avaya 1100 Series or 1200 Series SIP subscribers were able to create 3 party conferences when call was forwarded or re directed to Avaya Aura Messaging Long Duration Scenarios e Verify Avaya 1100 Series or 1200 Series SIP subscribers can remain on active call with other stations for at least 30 minutes e Verify Avaya 1100 Series or 1200 Series SIP subscribers can place a call on hold to other stations for at least 30 minutes e Verify Avaya 1100 Series or 1200 Series SIP subscribers can leave long voice mail messages for other subscribers 6 4 Known Limitations Since Avaya 1100 Series and 1200 Series IP Deskphones with SIP software have not implemented Presence features testing with Avaya Presence Services has been deferred until a future time DJH Reviewed Solution amp Interoperability Test Lab Application Notes 44 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 7 Acronyms AAR Automatic Alternate Routing Routing on Communication Manager ARS Automatic Route Selection Routing on Communication Manager CLAN Control LAN Control Card in Communication Manager DCP Digital Communications Protocol DTMF Dual Tone Multi Frequency IMS IP Multimedia Subsystem IP Internet Protocol SAT System Access Terminal Communication Administration Inte
52. such as when calls are transferred to Avaya Aura Messaging For simplicity the Trunk to Trunk Transfer field on Page 1 was set to all to enable all trunk to trunk transfers on a system wide basis Note Enabling this feature poses significant security risk by increasing the risk of toll fraud and must be used with caution To minimize the risk a COS could be defined to allow trunk to trunk transfers for specific trunk group s For more information regarding how to configure Communication Manager to minimize toll fraud see Reference 12 in Section 9 change system parameters features Page Leg i FEATURE RELATED SYSTEM PARAMETERS Self Station Display Enabled n Trunk to Trunk Transfer all pack with Called Party Queuing n wer Timeout Interval rings 3 3 3 Configure IP Codec Set Use the change ip codec set n command where n is the number used to identify the codec set Enter the following values e Audio Codec Enter G 711MU and G 729 as supported types e Silence Suppression Retain the default value n e Frames Per Pkt Enter 2 e Packet Size ms Enter 20 e Media Encryption Enter the value based on the system requirement For the sample configuration none was used change ip codec set 1 Page L e 2 LE Cogkee Ze Code er 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size ms 1 G 711MU n 2 20
53. te prefix e Total Length Enter 7 since a private prefix was not defined change private numbering 7 Page Lox 2 NUMBERING PRIVATE FORMAT dpe lt E Trk Private Total Len Code Grp s Prefix Len 7 444 10 7 Total Administered 1 Maximum Entries 540 DJH Reviewed Solution amp Interoperability Test Lab Application Notes 15 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 Step 2 Administer Uniform Dialplan Use the change uniform dialplan n command where n is the first digit of the extension numbers used for SIP stations in the system In the sample configuration 7 digit extension numbers starting with 444 3xxx are used for extensions associated with Avaya 1100 Series or 1200 Series SIP Deskphones Fill in the indicated fields as shown below and use default values for remaining fields e Matching Pattern Enter digit pattern of extensions associated with SIP stations e Len Enter extension length e Net Enter aar change uniform dialplan 6 Page Leg 2 UNIFORM DIAL PLAN TABLE Penae emits Jop Matching Insert Node Pattern Je Det DOMES Net Conv Num 4443 7 0 aar n 4445 7 0 aar n Ge 7 0 Sree ial Wis 2 0 ciek im 3 9 Administer Locations This section provides the configuration of the Locations screen Configuring a default route is necessary to enable stations on Communication Manager to use Avaya Aura Messaging features such as Cal
54. ture PARM No Numbering LAR 012M A W Request Dgts Format Subaddress l yyyyyn rest levO pvt none Se r e ae mm TSSE lev0 pvt none SYR TE AWA WE AV Ale Gal ial TESE none DJH Reviewed Solution amp Interoperability Test Lab Application Notes 14 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 3 8 Administer Numbering Plan Extension numbers used for SIP Users registered to Session Manager must to be added to either the private or public numbering table on Communication Manager For the sample configuration private numbering was used and all extension numbers were unique within the private network However in many customer networks it may not be possible to define unique extension numbers for all users within the private network For these types of networks additional administration may be required as described in Reference 9 in Section 9 Step 1 Administer Private Numbering Plan Use the change private numbering n command where n is the length of the private number Fill in the indicated fields as shown below e Ext Len Enter length of extension numbers In the sample configuration 7 was used e Ext Code Enter leading digit s from extension number In the sample configuration 444 was used e Trk Grp s Enter row for trunk group defined in Section 3 6 e Private Prefix Leave blank unless an enterprise canonical numbering scheme is defined in Session Manager If so enter the appropria
55. values for remaining fields e Name Enter a descriptive name for the location e Notes Add a brief description Optional In the Location Pattern section click Add and enter the following values e IP Address Pattern Enter the logical pattern used to identify the location For the sample configuration 192 160 112 was used e Notes Add a brief description Optional Click Commit to save The screen below shows the Location defined for the Avaya 1100 Series and 1200 Series SIP Deskphones used in the sample configuration Routing i Home Elements Routing Locations Location Details Domains Help Adaptations a Call Admission Control has been set to ignore SDP All calls will be counted using the Default Audio Bandwidth SIP Entities See Session Manager gt Session Manager Administration gt Global Setting Entity Links Time Ranges General Routing Policies Dial Patterns Regular Expressions San Overall Managed Bandwidth Managed Bandwidth Units Kbit sec e Total Bandwidth Per Call Bandwidth Parameters Default Audio Bandwidth 80 Kbit sec Location Pattern 1 Item Refresh Filter Enable IP Address Pattern Notes 192 160 112 Select All None DJH Reviewed Solution amp Interoperability Test Lab Application Notes 23 of 48 SPOC 05 05 2011 2011 Avaya Inc All Rights Reserved 11xx12xx_SM6 1 4 3 Define Routing Policy Routing policies describe

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