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Avaya 9600 Telephone User Manual

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1. IP NETWORK REGION Region n Location Authoritative Domain Name Intra region IP IP Direct Audio n MEDIA PARAMETERS Inter region IP IP Direct Audio n Codec Set 1 IP Audio Hairpinning n UDP Port Min 2048 UDP Port Max 3028 RTCP Reporting Enabled y RTCP MONITOR SERVER PARAMETERS DIFFSERV TOS PARAMETERS Use Default Server Parameters y Call Control PHB Value Server IP Address 3 Audio PHB Value Server Port 5005 Video PHB Value 802 1P Q PARAMETERS RTCP Report Period secs 5 Call Control 802 1p Priority 7 Audio 802 1p Priority 6 Video 802 1p Priority 7 AUDIO RESOURCE RESERVATION PARAMETERS H 323 IP ENDPOINTS RSVP Enabled y H 323 Link Bounce Recovery y RSVP Refresh Rate secs 15 Idle Traffic Interval sec 20 Retry upon RSVP Failure Enabled y Keep Alive Interval sec 6 RSVP Profile Keep Alive Count 5 RSVP unreserved BBE PHB Value 40 Issue 2 December 2007 125 Sample Station Forms Figure 10 IP Network Region page 2 change ip network region n Page 2 of x IP NETWORK REGION INTER GATEWAY ALTERNATE ROUTING DIAL PLAN TRANSPARENCY Incoming LDN Extension Conversion to Full Public Number Delete __ Insert Maximum Number of Trunks to Use for IGAR Dial Plan Transparency in Survivable Mode n 323 SECURITY PROCEDURES BACKUP SERVERS IN PRIORITY ORDER DoF WN FE Allow SIP URI Conversio
2. CALL CENTER SYSTEM PARAMETERS Expert Agent Selection Minimum Agent LoginID P EAS Enabled n assword Length Direct Agent Announcement Extension Delay Message Waiting Lamp Indica VECTORING Converse Fi Converse Signaling Prompting Interflow qpos Reverse Star Pound Digit Fo Available Agent Adjus BSR tes Status For station rst Data Delay 0 Second Data Delay 2 Tone msec 100 Pause msec 70_ Timeout secs 10 EWT Threshold 2 r Collect Step n tments for BSR n Tie Strategy l1st_found Store VDN Name in Station s Local Call Log n n ERVICE OBSERVING Service Observing Warning Tone n or Conference Tone n Service Observing Allowed with Exclusion n Allow Two Observer s in Same Call n 124 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Figure 8 IP Address Mapping Form change ip network map IP ADDRESS MAPPING Subnet FROM IP Address TO IP Address or Mask Region Le Zita 3 10 Drie 2a 35259 24 1 E o diy eg 3 32 2 E T 5 Tor Die AS 25 3 Tai a As SG Tey 0 Ain 9 4 Figure 9 IP Network Region Form change ip network region n Page 1 of X Emergency 802 10 Location VLAN Extension 3 i 0 _ en 0 4 Page 1 of x
3. SKINS SNMPADD SNMPSTRING SNTPSRVR SPEAKERSTAT SUBSCRIBE_SECURITY SUPPORT_GIGABIT SYSTEM_ LANGUAGE TCP_KEEP_ALIVE INTERVAL TCP_KEEP_ALIVE STATUS Nu Null Null Null Null 10 Applicable to the SIP 9640 IP Telephone only Represents a list of skin information tuples Each skin information is a pair of skin label skin URL data Each skin tuple is delimited by commas Each skin tuple contains skin label verbatim label displayed on the screen and skin URL Skin label and URL are separated by a The URL may be specified in an absolute or relative path format for next higher directory level in relative path format origin is the directory specified by HTTPDIR or TLSDIR depending on download via http or https String maximum is 1023 characters Example Yankees Color http svn avaya com drop skins yankees_color boohisscolor xml Text string containing zero or more allowable source IP Addresses for SNMP queries in dotted decimal or DNS format separated by commas with up to 255 total ASCII characters including commas and no intervening spaces Text string containing the SNMP community name string up to 32 ASCII characters no spaces Used to retrieve date and time via SNTP in case of several entries first address always first etc Zero to 255 characters zero or more IP Addresses in dotted decimal or DNS name format separated by commas without any intervenin
4. 2 0082 e eee 110 Chapter 9 Administering Applications and Options 111 Customizing Telephone Applications and Options 8 111 Avaya A Menu Administration 1 2 2 ee 112 Administering Standard Avaya Menu Entries 0 28585 112 Administering the WML Browser 2 2 eee eee ee ee es 112 Appendix A Glossary of Terms 2 20852228 82 s 115 Appendix B Related Documentation 2 2 5 119 IETF DOCUMOIIS 44 4 4 4 8 0 Re ee ee 119 IMU Documents 2 04 264A DEES ANODE Ae Aw ORS 119 ISO IEC ANSI IEEE Documents 0 00002 ee eee eee 119 Appendix C Sample Station Forms 000 5885 ae 121 ee ee Se bebe Ge oe ees Ge eee eee oe Be ees 135 6 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 1 Introduction About This Guide This guide is for personnel who administer Avaya Communication Manager DHCP HTTP HTTPS servers for 9600 Series SIP IP Telephones a Local Area Network LAN SIP Enablement Services SES or a Network Time server The 9600 Series IP Telephones use Internet Protocol IP technology with Ethernet line interfaces and support both SIP and H 323 protocols The 9600 Series IP Telephones provide support for DHCP HTTP and HTTPS over IPv4 UDP which enhance the administration and servicing of the telephones These telephones use DHCP to obtain dynamic IP Addresses and HTTPS or HTTP to download new versions of softwar
5. Flag to define whether or not menu item s for MS Exchange Calendar integration are provided to the user Values are O off 1 on If disabled the menu item in Options amp Settings sub menu to select access to MS Exchange Calendar is hidden to the user If PROVIDE_EXCHANGE_CONTACTS is also disabled the complete sub branch for MS Exchange integration is hidden Values are O Off 1 On Flag to define whether or not menu item s for MS Exchange Contacts integration are provided to user If disabled the menu item in Options amp Settings sub menu to select access to MS Exchange Contacts is hidden If PROVIDE_EXCHANGE_CALENDAR is also disabled the complete subbranch for MS Exchange integration is hidden Values are O Off 1 On Flag to define whether or not logout function is provided to user If disabled and phone is operating in user mode hide Logout item in option menu Values are O off 1 on Flag to define whether or not Network Information menu is provided to user If disabled and phone is operating in user mode hide complete Network Information Values are O off 1 0n Flag to define whether or not Options amp Settings menu is provided to user If disabled and phone is operating in user mode hide complete Option amp Settings menu tree Values are O off 1 on Flag to determine whether user can select a Transfer Type Attended Unattended via the Avaya A Menu Call Settings options Applies to 3rd pa
6. Default Server Port Default RTCP Report Period secs AUTOMATIC TRACE ROUTE ON Link Failure y H 248 MEDIA GATEWAY Link Loss Delay Timer min H 323 IP 5 Link Loss Primary Periodic Registration High High sec Figure 17 IP Options System Parameters Form page 2 change system parameters ip options IP OPTIONS SYST Always use G 711 IP DTMF TRANSMISSION MODE 30ms no SS Intra System IP DTMF Transmission Mode Inter System IP DTMF HYPERACTIVE MEDIA GATEWAY R EGISTRATIONS Enable Detection and Alarms EM PARAMETERS in band g711 Page 1 of x 800 Low 400 40 Low 15 20 10 5005 5 ENDPOINT Delay Timer min Search Time sec Timer min Page 2 of x for intra switch Music On Hold See Signaling Group Forms Issue 2 December 2007 129 Sample Station Forms Figure 18 Class of Restriction screen page 1 change cor n Page 1 of x CLASS OF RESTRICTION COR Number n COR Description supervisor FRL 0 APLT y Can Be Service Observed n Calling Party Restriction none Can Be A Service Observer y Called Party Restriction none Partitioned Group Number 1 Forced Entry of Account Codes n Priority Queuing n Direct Agent Calling y Restriction Override none Facility Access Trunk Test n Restricted Call List n Can Change Coverage n Unrestricted Call L
7. IEEE 8021X d cee ae Ba ae a es ee E aa IEEE ANSI Documents IETF Documents 2 a eee Initialization Process for 9600 Series IP Telephones 2 1 2 ee a Installation Network Information Required before installing e eor ee a ea Interface administeringthe IP Address Mapping Form IP Addresses administering IP Codec Set Form 2005 IP Interface and Addresses IP Options System Parameters Form ISO IEC ANSIJEEE Documents ITU Documents 2 4 k M oa N Sa gl o B o aak o o L Language Selection Link Layer Discovery Protocol LLDP LLDP Data Units Transmitted Local Administrative Options N Network Assessment 2 200 Network Audio Quality Display Network Considerations Other Network Information Required Network Region Form Network Requirements 25 Network Time Protocol Server Network Time Server 2 205 NTP S NVCR fa ae ga ae ae ee ew we Numbering Public Unknown Format Form O Options and Applications Administering Options Administering Options Customizing Options entering using the Telephone Dialpad Options for 9600 Serie
8. Internal extension telephone number length Specifies the number of digits associated with internal extension numbers by the algorithm that dials calls from the incoming Call Log or from Web pages Range 1 or 2 digits from 3 to 13 Telephone international access code The maximum number of digits if any dialed to access public network international trunks by the algorithm that dials calls from the incoming Call Log or from Web pages Range 0 4 digits Telephone long distance access code The digit if any dialed to access public network long distance trunks Range 1 digit 0 to 9 or Null Needed to for Enhanced Local Dialing Algorithm Length of national telephone number The number of digits in the longest possible national telephone number Range 5 to 15 Needed to for Enhanced Local Dialing Algorithm Outside line access code The character s dialed including and if any to access public network local trunks Range 0 2 dialable numeric digits including Null 12 of 21 Issue 2 December 2007 385 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range PHNNUMOFSA PHY1STAT PHY2PRIO PHY2STAT PHY2VLAN POE_CONS_SUPPORT Nu i When ENABLE_AVAYA_ENVIRONMENT 0 this value sets the number of Session Appearances Ethernet line interface
9. SIP IP Telephone continued Destination Port Source Port Use UDP or TCP 514 CNAPORT The port number specified in the test request message System specific FEPORT or the port number specified in a CNA RTP test request FEPORT 1 if FEPORT is even or FEPORT 1 if FEPORT is odd or the port number specified ina CNA RTP test request plus or minus one as with FEPORT above Any unused port number Any otherwise unused port number 50000 Any unused port number PORTAUD which must be in the range specified by the RTP_PORT _LOW and RTP_PORT _RANGE parameters or the port number reserved for CNA RTP tests PORTAUD 1 if PORTAUD is even or PORTAUD 1 if PORTAUD is odd or the port number reserved for CNA RTP tests plus or minus one as for PORTAUD above Transmitted Syslog UDP messages Transmitted CNA registration messages TCP Transmitted CNA test results UDP messages Transmitted signaling protocol packets Transmitted RTP and SRTP packets TCP UDP RTCP and SRTCP packets transmitted to the far end of the audio connection UDP 2of3 Issue 2 December 2007 33 Network Requirements Table 6 Transmitted Packets Source SIP IP Telephone continued Destination Port Source Port Use UDP or TCP RTCPMONPORT PORTAUD RTCP packets transmitted to UDP 1 if an RTCP monitor PORTAUD is even or PORTAUD 1 if PORTAUD is odd
10. Interface number subtype 3 system port Interface number 1 OID SNMP MIB II sysObjectID of the telephone IEEE 802 3 MAC PHY Reports autonegotiation status and speed of Organization Configuration Status the uplink port on the telephone Specific TIA LLDP MED LLDP MED Media Endpoint Discovery capabilities 00 33 Capabilities Inventory Power via MDI Network Policy MED Caps TIA LLDP MED Network Policy Tagging Yes No VLAN ID for voice L2 Priority DSCP Value TIA LLDP MED Inventory Hardware MODEL Full Model Name Revision TIA LLDP MED Inventory Firmware BOOTNAME Revision TIA LLDP MED Inventory Software APPNAME Revision TIA LLDP MED Inventory Serial Telephone serial number Number TIA LLDP MED Inventory Avaya Manufacturer Name TIA LLDP MED Inventory Model MODEL with the final Dxxx characters Name removed 2of3 102 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Link Layer Discovery Protocol LLDP Table 13 LLDPDU Transmitted by 9600 Series SIP IP Telephones continued Category TLV Name Type TLV Info String Value Avaya Proprietary Avaya Proprietary Avaya Proprietary Avaya Proprietary Avaya Proprietary Avaya Proprietary Basic Mandatory PoE Conservation Level Support Call Server IP Address IP Phone Addresses CNA Server IP Address File Server 802 1Q Framing End of LLDPDU Provides Power Conservation abilities se
11. PPM Proxy Server PSTN QoS RSVP RTCP RTP SCEP SDP A secure version of HTTP Internet Engineering Task Force the organization that produces standards for communications on the internet Local Area Network Link Layer Discovery Protocol All IP telephones with an Ethernet interface support the transmission and reception of LLDP frames on the Ethernet line interface in accordance with IEEE standard 802 1AB SIP Software Release 2 0 and up supports LLDP Media Access Control ID of an endpoint Encryption of the audio information exchanged between the IP telephone and the call server or far end telephone Network Address Port Translation Network Address Translation Off PBX Station Personal Profile Manager part of the SIP Enablement Services SES platform PPM is responsible for maintaining and managing end users personal information in the system An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients A proxy server primarily plays the role of routing meaning its job is to ensure that a request is sent to another entity closer to the targeted user Proxies are also useful for enforcing policy for example making sure a user is allowed to make a call A proxy interprets and if necessary rewrites specific parts of a request message before forwarding it Public Switched Telephone Network the network used for traditional telephony Q
12. language files and settings files from either an HTTP or HTTPS server The telephone queries the file server which transmits a script file to the telephone This script file ata minimum tells the telephone which binary file the telephone must use The binary file is the software that has the telephony functionality The telephone uses the script file to determine if it has the proper binary file If the telephone determines the proper binary file is missing the telephone requests an binary file download from the file server The file server then downloads the file and conducts some checks to ensure that the file was downloaded properly If the telephone determines it already has the proper file the telephone proceeds as described in the next paragraph without downloading the binary file again The telephone checks and loads the binary file then uses the script file to look for a settings file if appropriate The optional settings file can contain settings you have administered for any or all of the 9600 Series SIP IP Telephones in your network For more information about this download process and settings file see Chapter 7 Telephone Software and Binary Files Step 5 Telephone and the SES Server In this step the telephone might prompt the user for an extension and password The telephone uses that information to exchange a series of messages with SES which in turn communicates with Avaya Communication Manager CM For a new inst
13. 9600 Series IP Telephones are supported by Avaya Communication Manager CM Release 4 0 and later Be sure to administer 9600 Series SIP IP Telephones as 4620S IP telephones on Avaya Communication Manager Note The 9620 only supports a total of 12 call appearances and administered feature buttons The 9630 9630G and 9640 9640G can be administered for a total of 24 call appearances and feature buttons For specific administration instructions about the 9600 Series SIP IP Telephones see Administering Stations on page 48 Communication Manager Administrative Requirements There are several initial CM provisioning tasks that must be performed before administering SIP users These tasks are described in S P Support in Avaya Communication Manager Running on Avaya S8XXX Server s Document Number 555 245 206 the latest release of which is Issue 8 January 2008 The tasks to administer Communication Manager for SIP Enablement Services SES and fall into three categories e system level preparation e SIP trunk administration and e call routing administration The sections that follow describe each of these tasks Issue 2 December 2007 37 Communication Manager Administration System Level Preparation Tasks The system level preparation tasks include e Setting the SIP Trunk capacity on the System Capacity screen e Verifying that the IP Trunks field is set to y on the System Parameters Customer Options screen page 4 e Verifying th
14. Applications and Options provide administrative control of telephone functions and options Appendix A Glossary of Terms Provides a glossary of terms used in this document or which can be applicable to 9600 Series SIP IP Telephones Appendix B Related Provides references to Web sites with external documents that Documentation relate to telephony in general and can provide additional information about specific aspects of the telephones Appendix C Sample Station Provides examples of Avaya Communication Manager forms Forms related to system wide and individual telephone administration Other Documentation See the Avaya support site at http Awww avaya com support for 9600 Series SIP IP Telephone technical and end user documentation See Appendix B Related Documentation for Web sites that list related non Avaya documents such as those published by the Internet Engineering Task Force IETF and the International Telecommunication Union ITU Issue 2 December 2007 13 Introduction 14 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 2 Administration Overview and Requirements 9600 Series IP Telephones The 9600 Series IP Telephones currently support the H 323 signaling protocol and the SIP signaling protocol The H 323 standard provides for real time audio video and data communications transmission over a packet network An H 323 telephone protocol stack comprises several pro
15. Code form change off pbx station mapping xxxx change system parameters features change system parameters features change system parameters features change system parameters features change dialplan analysis change feature access codes RTCP Report Period secs Authoritative Domain Music Tone on Hold Directed Call Pickup Extended Group Call Pickup Whisper Page Tone Given To Call Type Various fields on pages 1 5 of the form SIP telephones have a fixed reporting period Note that this parameter is only displayed if Use Default Server Parameters is set to n Make sure that the Authoritative Domain is set to the same value as SIP Domain for Solution Bridged call items on this form MUST be none or orig In CM Release 5 0 default is none This CM setting controls the music on hold capability for all endpoints including SIP telephones This CM setting controls the availability of directed call pickup This CM setting allows a user to answer calls that were directed to another call pickup group This CM setting controls who hears the whisper page Includes all telephone extensions and OPS Feature Name Extensions FNEs To define the FNEs for the OPS features listed in Table a FAC must also be specified for the corresponding feature In a sample configuration telephone extensions are five digits in length and begin with 3 or 4 FNEs are fiv
16. Customizeable System Parameters continued Parameter Name Default Value Description and Value Range MUSICSRVR MWISRVR Null MYCERTCAID CAldentifier MYCERTCN SERIALNO MYCERTDN Null MYCERTKEYLEN 1024 MYCERTRENEW 90 MYCERTURL Null List of Music on Hold Server IP or DNS address es Used to retrieve Music on hold for audio streams of sessions put on hold in case of several entries first address always first etc In some third party proxy environments the SIP proxy registrar might be different from the Music on Hold server In this case the Music on Hold server is set via this parameter If both functions are provided by the same server it is not necessary to set MUSICSRVR and the SIP proxy server is used for Music on Hold Zero to 255 characters zero or more IP addresses in dotted decimal or DNS name format separated by commas without any intervening spaces if operating in a non Avaya environment this value is set via a SET command in the settings file otherwise the address of SIP Proxy server is used List of Message Waiting Indicator Event Server IP or DNS address es Used to register for MWI event notifications in case of several entries first address always first etc In some third party proxy environments the SIP proxy registrar may be different than the MWI server In this case the MWI server is set via this parameter If both functions are provided by the same server it is not nece
17. Network icon 3 Verify that Microsoft DHCP Server is listed as one of the Network Services on the Services tab 4 If it is listed continue with the next section If it is not listed install the DHCP server Issue 2 December 2007 59 Server Administration Creating a DHCP Scope for the IP Telephones Use the following procedure to create a DHCP scope for the IP telephones 1 2 Select Start gt Programs gt Admin Tools gt DHCP Manager Expand Local Machine in the DHCP Servers window by double clicking it until the sign changes to a sign 3 Select Scope gt Create 4 Using information recorded in Table 3 Required Network Information Before Installation Per DHCP Server Define the Telephone IP Address Range Set the Subnet Mask To exclude any IP Addresses you do not want assigned to IP telephones within the Start and End addresses range a In the Exclusion Range Start Address field enter the first IP Address in the range that you want to exclude b In the Exclusion Range End Address field enter the last IP Address in the range that you want to exclude c Click the Add button d Repeat steps a through c for each IP Address range to be excluded Note Avaya recommends that you provision the 9600 Series IP Telephones with sequential IP Addresses Also do not mix 9600 Series IP Telephones and PCs in the same scope Under Lease Duration select the Limited To option and set the lease du
18. PROVIDE EXCHANGE CONTACTS e QKLOGINSTAT e RICPCONT Issue 2 December 2007 11 Introduction RTCPMON RTCPMONPORT SIP_MODE SIPCONFERENCECONTINUE TLSSRVRID VU_MODE VU_TIMER WMLEXCEPT WMLHOME WMLIDLETIME WMLIDLEURI WMLPORT WMLPROXY The following parameters have been modified or renamed Parameters PHYxDUPLEX and PHYxSPEED were combined PHY1SPEED has been renamed to PHY1_ OPERATIONAL_MODE This parameter now includes the current duplex mode PHY2SPEED has been renamed to PHY2 OPERATIONAL_MODE This parameter now includes the current duplex mode The OUTBOUND_SUBSCRIPTION_REQUEST_DURATION default value has been changed from 17280000 to 86400 seconds This parameter can now be set through the settings file The dimensions for SNTP_SYNC_INTERVAL and SNTP_SYNC_RANDOMIZATION_INTERVAL have changed from seconds to minutes EXCHANGE_CONTACTS_ENABLED has been renamed to USE_EXCHANGE_CONTACTS EXCHANGE_CALENDAR_ENABLED has been renamed to USE _EXCHANGE_CALENDAR The default value definition of ENABLE _G726 has changed from 0 to 1 The default values and sidetone definitions of the audio parameters AUDIOSTHD and AUDIOSTHS have been modified WAIT_FOR_REGISTRATION_TIMER can now be set through the settings file The following configuration parameters are no longer valid and have been removed PHY1DUPLEX PHY2DUPLEX INTER_DIGIT_DIALING_TIMEOUT_DURATION 12 9
19. System specific Any unused Transmitted signaling UDP port number protocol packets 3 of 3 Security For information about toll fraud see the respective call server documents on the Avaya support Web site The 9600 Series SIP IP Telephones cannot guarantee resistance to all Denial of Service attacks However there are checks and protections to resist such attacks while maintaining appropriate service to legitimate users 9600 Series SIP IP Telephones support Transport Layer Security TLS for signaling and for secure communications SRTP This standard allows the telephone to establish a secure connection to a HTTPS server in which the upgrade and settings file can reside This setup adds security over another alternative Communications between the 9600 Series SIP IP telephone and the Personal Profile Manager PPM can also be secured by setting the CONFIG_SERVER_SECURE_MODE parameter You also have a variety of optional capabilities to restrict or remove how crucial network information is displayed or used These capabilities are covered in more detail in Chapter 6 Server Administration and include e Depending on the SIGSIGNAL parameter supporting signaling channel encryption while registering and when registered with appropriately administered Avaya Communication Manager e Restricting the response of the 9600 Series SIP IP Telephones to SNMP queries to only IP Addresses on a list you specify e Specifying an SNMP com
20. alternate file may be included depending on which software bundle you download e Binary files for all current 9600 Series SIP IP Telephones e Other useful information such as a ReadMe file and the latest binary code In addition to the upgrade script binary files and Read Me file you need the latest binary code the Avaya SIP IP Telephones use which is part of the software bundle you choose for your site All these files are in self extracting executable file comes in both zipped and unzipped format When the majority of your IP telephones are SIP based select the software bundle identified as SIP from the Avaya Support Web site The binary files in this SIP software bundle are the same as in the H 323 bundle The difference is a modified upgrade script file that assumes SIP is the default protocol for your 9600 Series IP Telephones and that H 323 is the exception For more information on SIP centric environments see Converting Software on 9600 Series IP Telephones in the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide 68 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 9600 Series SIP IP Telephone Scripts and Binary Files Upgrade Script File An upgrade script file named 96xxupgrade txt tells the IP telephone whether the telephone needs to upgrade software The 9600 Series SIP IP Telephones attempt to read this file whenever they reset The upgrade scri
21. and Avaya does not necessarily endorse the products services or information described or offered within them We cannot guarantee that these links will work all of the time and we have no control over the availability of the linked pages Warranty Avaya Inc provides a limited warranty on this product Refer to your sales agreement to establish the terms of the limited warranty In addition Avaya s standard warranty language as well as information regarding support for this product while under warranty is available through the following Web site http Awww avaya com support Copyright Except where expressly stated otherwise the Product is protected by copyright and other laws respecting proprietary rights Unauthorized reproduction transfer and or use can be a criminal as well as a civil offense under the applicable law Avaya support Avaya provides a telephone number for you to use to report problems or to ask questions about your product The support telephone number is 1 800 242 2121 in the United States For additional support telephone numbers see the Avaya Web site http Awww avaya com support Software License USE OR INSTALLATION OF THE PRODUCT INDICATES THE END USER S ACCEPTANCE OF THE TERMS SET FORTH HEREIN AND THE GENERAL LICENSE TERMS AVAILABLE ON THE AVAYA WEBSITE AT http support avaya com Licenselnfo GENERAL LICENSE TERMS IF YOU DO NOT WISH TO BE BOUND BY THESE TERMS YOU MUST RETURN TH
22. be download A substring specifies the completed URL to the language file including protocol identifier http or https server and path Flag to enable disable LLDP Link Layer Discovery Protocol Valid values are 0 disabled the telephone will not support LLDP 1 enabled the telephone will support LLDP 2 auto the telephone will support LLDP but the transmission of LLDP frames will not begin until or unless an LLDP frame is received Numerical code of severity level Store entries to the local event log if event occurs with a severity level whose numerical code is equal to or less than the LOCAL_LOG_LEVEL value Values are 0 emergencies 1 alerts 2 critical 3 errors 4 warning 5 notice 6 informational 7 debug 9 of 21 82 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range LOG_CATEGORY LOGOS LOGSRVR MEDIAENCRYPTION MSGNUM MTU_SIZE Null Null Null 1500 Comma separated list of keywords in standard string format representing logging categories SW modules or functions to be included in lower level logging Logging implementation blocks all traces at level Warning or lower unless the category corresponding to a given
23. connection as the transport protocol for SCEP List of IP Address es or DNS Name s of HTTP file server s used to download telephone files HTTP server addresses can be in dotted decimal or DNS format and must be separated by commas 0 255 ASCII characters including commas Controls whether ICMP Destination Unreachable messages will be processed Values are 0 DU messages not transmitted 1 DU messages not transmitted in response to specific events 2 DU message with code 2 will be transmitted in case of specific events Controls whether ICMP Redirect messages will be processed Values are 0 Redirect messages will neither be transmitted nor received Redirect messages will be supported 1 Redirect messages will not be transmitted but received Redirect messages will be supported per RFC 1122 This is the timeout that takes place when user stops inputting digits The timeout is treated as digit collection completion and when it occurs the application sends out an invite Range in seconds of 1 to 10 IP Address of the telephone Range is 7 to 15 ASCII characters less than the default string length defining one IP Address in dotted decimal format Requests 802 1Q tagging mode auto on off Values are 0 auto 1 on 2 off Layer 2 audio priority value Range from 0 to 7 Layer 2 signaling priority value Range from 0 to 7 8 of 27 Issue 2 December 2007 81 Administering Telephone Options Table 11 9600 Serie
24. disabled 1 Supplicant operation enabled but responds only to received unicast EAPOL messages 2 Supplicant operation enabled responds to received unicast and multicast EAPOL messages Differentiated Services Code Point for audio Values range from 0 to 63 Differentiated Services Code Point for signaling Values range from 0 to 63 Used for daylight saving time calculation in hours Values range from 0 to 2 Used to identify start date for automatic change to Daylight Saving Time Default string length with a format of either odddmmmht or Dmmmht where o one character representing an ordinal adjective of 1 first 2 second 3 third 4 fourth or L last ddd 3 characters containing the English abbreviation for the day of the week mmm 3 characters containing the English abbreviation for the month h one numeric digit representing the time to make the adjustment exactly on the hour at hAM 0h00 in military format where valid values of h are 0 through 9 t one character representing the time zone relative to the adjustment where L is local time and U is universal time D one or two ASCII digits representing the date of the month from 1 or 01 to 31 or the character L which means the last day of the month 4 of 21 Issue 2 December 2007 77 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name D
25. enabled dialing rules also apply for dialing from Contacts Note that If CTDC_SUPPORT is enabled Enhanced Local Dialing is automatically disabled independent of the actual setting of ENHDIALSTAT If CTDC_SUPPORT is disabled Enhanced Local Dialing is processed as defined by ENHDIALSTAT EXCHANGE_SERVER_ Null Used to connect to Microsoft Exchange server to LIST access calendar data Zero to 255 characters zero or more IP Addresses in dotted decimal or DNS name format separated by commas without any intervening spaces EXCHANGE_USER_ Null String of 0 to 255 characters representing user domain DOMAIN for Microsoft Exchange Server FAILED SESSION _ 30 Timer to automatically remove a failed call session REMOVAL_TIMER Range in seconds is 5 to 999 G726_PAYLOAD_TYPE 110 a eae payload used for G726 Range is 96 to 127 GMTOFFSET 0 00 Offset used to calculate time from GMT reference time Default string length positive or negative number of hours and minutes less than 13 hours GROUP 0 Specific user group as tested in configuration files Valid values are 0 to 999 HEADSYS 1 Headset operational mode One ASCII numeric digit Valid values are 0 or 2 General Operation where a disconnect message returns the telephone to an idle state 1 or 3 Call Center Operation where a disconnect message does not change the state of the telephone HTTPDIR Null HTTP server directory path The path name prepended to all file names
26. exclude c Click the Add button d Repeat steps a through c for each IP Address range that you want to exclude Note You can add additional exclusion ranges later by right clicking the Address Pool under the newly created scope and selecting the New Exclusion Range option Click the Next button after you enter all the exclusions The Lease Duration dialog box displays For all telephones that obtain their IP Addresses from the server enter 30 days in the Lease Duration field This is the duration after which the IP Address for the device expires and which the device needs to renew Click the Next button The Configure DHCP Options dialog box displays Click the No I will activate this scope later button The Router Default Gateway dialog box displays For each router or default gateway enter the IP Address and click the Add button When you are done click the Next button The Completing the New Scope Wizard dialog box displays Click the Finish button The new scope appears under your server in the DHCP tree The scope is not yet active and does not assign IP Addresses Highlight the newly created scope and select Action gt Properties from the menu Under Lease duration for DHCP clients select Unlimited and then click the OK button AX CAUTION IP Address leases are kept active for varying periods of time To avoid having calls terminated suddenly make the lease duration unlimited 64 9600 Ser
27. file the telephone must use The binary file is the software that has the telephony functionality and is easily updated for future enhancements In a newly installed telephone the binary file might be missing In a previously installed telephone the binary file might not be the proper one In both cases the telephone requests a download of the proper binary file from the file server The file server downloads the file and conducts some checks to ensure that the file was downloaded properly If the telephone determines it already has the proper file the telephone proceeds to the next step without downloading the binary file again After checking and loading the binary file the 9600 Series SIP IP Telephone if appropriate uses the script file to look for a settings file The settings file contains options you have administered for any or all of the IP Telephones in your network For more information about the settings file see Contents of the Settings File on page 70 Software As part of installation a conversion from H 323 to SIP signaling protocol is done as described in Converting Software on 9600 Series IP Telephones of the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide When the telephone is first plugged in a software download from an HTTP or HTTPS server starts to give the phone its proper functionality For software upgrades SIP Enablement Services SES provides the capabi
28. from LLDP For more information see Link Layer Discovery Protocol LLDP on page 101 The Administrative Process The following list depicts administration for a typical 9600 Series SIP IP Telephone network Your own configuration might differ depending on the servers and system you have in place 1 Avaya Communication Manager 4 0 or greater administered for 9600 Series IP Telephones All 9600 Series SIP IP Telephones must be administered with the 4620SIP station type 2 SES SIP Enablement Services administered 3 LAN and applicable servers file servers Network Time server administered to accept the telephones 4 Telephone software downloaded from the Avaya support site 5 46xxsettings file updated with site specific and SIP specific information as applicable 6 9600 Series Telephones installed For more information see the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide 7 Individual 9600 Series IP Telephones updated using Craft procedures as applicable For more information see Local Administrative Procedures in the Avaya one X Deskphone Edition for 9600 SIP P Telephones Installation and Maintenance Guide Administrative Checklist Use the following checklist as a guide to system and LAN administrator responsibilities This high level list helps ensure that all telephone system prerequisites and requirements are met prior to telephone installation Not
29. is zero or one URL TCP port number to be used to access the HTTP proxy server by the WML browser application if defined by WMLPROXY Valid value is 0 65535 20 of 21 Issue 2 December 2007 93 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Description and Value Range Value WMLPROXY Null Address of WML proxy server WMLPROXY is used as the HTTP proxy server by the WML browser application If WMLPROXyY is null or if WMLPROXY cannot be resolved into a valid IP address an HTTP proxy server is not used Value is zero or one IP address in dotted decimal or DNS name format Note that WMLPROXY defines the HTTP proxy server for WML browser application and HTTPPROXY to perform SCEP certificate enrollment 21 of 27 Note Table 11 applies to all 9600 Series SIP IP Telephones Certain 9600 SIP IP Telephones might have additional optional information that you can administer For more information see Chapter 8 Administering Telephone Options VLAN Considerations This section contains information on how to administer 9600 Series SIP IP Telephones to minimize registration time and maximize performance in a Virtual LAN VLAN environment If your LAN environment does not include VLANs set the system parameter L2Q to 2 off to ensure correct operation VLAN Tagging IEEE 802 1Q tagging VLAN is a useful method of m
30. messages sent to closed ports so as not to reveal information to potential hackers The default is 1 sends Destination Unreachable messages for closed ports used by traceroute ICMPRED Controls whether ICMP Redirect messages are processed The default is 0 redirect messages are not processed L2Q 802 1Q tagging mode The default is 0 automatic L2QVLAN VLAN ID of the voice VLAN The default is 0 LOGSRVR Syslog server IP or DNS address MTU_SIZE Maximum transmission unit size Used to accommodate older Ethernet switches that cannot support the longer maximum frame length of tagged frames since 802 1Q adds 4 octets to the frame PHY1STAT Controls the Ethernet line interface speed The default is 1 auto negotiate PHY2STAT Controls the secondary Ethernet interface speed The default is 1 auto negotiate PROCPSWD Security string used to access local procedures The default is 27238 PROCSTAT Controls whether local procedures are enabled The default is 0 enabled SIPPROXYSRVR SNTPSRVR TLSDIR TLSPORT TLSSRVR VLANTEST SIP proxy registrar server IP or DNS address 0 to 255 characters zero or one IP Address in dotted decimal or DNS name format separated by commas without any intervening spaces The default is null List of SNTP server IP or DNS address es u sed to retrieve date and time via SNTP Used as path name that is prepended to all file names used in HTTPS get operations during initialization 0 127 character
31. no message is received registration is retried Range is 1 60 seconds Exceptions domains for the WML browser proxy server If WMLPROXY is resolved and WMLEXCEPT is null the HTTP proxy server defined by WMLPROXyY is used for all transactions of the WML browser application If WMLEXCEPT is not null the HTTP proxy server is only used for the URLs whose domains are not on the WMLEXCEPT list Format is zero or more strings in DNS format separated by commas without any intervening spaces Home page for WML browser If this parameter is null the telephone will not display the browser option under the A Avaya Menu If non null the URL specified is retrieved via HTTP and rendered in the Web page display area when the WML browser application is initially accessed Value is zero or one URL Number of minutes of inactivity until the Web browser will display the idle URL When the Web idle timer reaches the number of minutes equal to this parameter the telephone sends an HTTP GET for the URI specified by WMLIDLEURI Valid value is 1 999 Note that the web idle timer starts only when access to the WML browser is provided by an application line under the A Avaya Menu and the parameter WMLIDLEURI is non null URL of web page displayed after idle timer expires Note that the web idle timer will only be started when access to the WML browser is provided by an application line under the A Avaya Menu and the parameter WMLIDLEURI is non null Value
32. performance while on a call For more information see the telephone user guide While on a call the telephones display network audio quality parameters in real time as shown in Table 4 Table 4 Parameters in Real Time Parameter Possible Values Received Audio Coding G711 G722 G726A or G 729 Packet Loss No data or a percentage Late and out of sequence packets are counted as lost if they are discarded Packets are not counted as lost until a subsequent packet is received and the loss confirmed by the RTP sequence number Packetization Delay No data or an integer number of milliseconds The number reflects the amount of audio data in each RTP packet One way Network Delay No data or an integer number of milliseconds The number is one half the value RTCP or SRTCP computes for the round trip delay Network Jitter No data or an integer number of milliseconds reporting the Compensation Delay average delay introduced by the jitter buffer of the telephone The implication for LAN administration depends on the values the user reports and the specific nature of your LAN like topology loading and QoS administration This information gives the user an idea of how network conditions affect the audio quality of the current call Avaya assumes you have more detailed tools available for LAN troubleshooting SIP Station Number Portability The 9600 Series SIP IP Telephones provide station number portability On startup or a reboot
33. procedure Intended to facilitate restricted access to local procedures even when command sequences are known Password is viewable not hidden PROCSTAT 0 Controls access to local dialpad administrative procedures Values are 0 Full access to craft local procedures 1 restricted access to craft local procedures PROVIDE_EDITED__ 2 Controls whether edited dialing is allowed and whether DIALING on hook dialing is disabled Valid values are 0 Disable edit dialing Dialing Options is not displayed to the user so the user cannot change edit dialing the telephone defaults to on hook dialing 1 Disable on hook dialing and do not display Dialing Options to the user so the user cannot change edit dialing the telephone defaults to edit dialing 2 Display Dialing Options to allow user to change from on hook to edit dialing This is the default 3 Display Dialing Options to allow user to change from edit dialing to on hook dialing the telephone defaults to edit dialing 14 of 21 Issue 2 December 2007 87 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range PROVIDE_EXCHANGE _ CALENDAR PROVIDE_EXCHANGE__ CONTACTS PROVIDE_ LOGOUT PROVIDE_ NETWORKINFO__ SCREEN PROVIDE_OPTIONS_ SCREEN PROVIDE_TRANSFER_ TYPE QKLOGINSTAT REGISTERWAIT ROUTER 1 3600 0 0 0 0
34. setting 1 auto negotiate 2 10Mbps half duplex 3 10Mbps full duplex 4 100Mbps half duplex 5 100Mbps full duplex and 6 1000Mbps full duplex if supported by the hardware Layer 2 priority value for frames received on or forwarded to the secondary Ethernet interface Set this parameter only when VLAN separation is 1 enabled Values are from 0 7 and correspond to the drop down menu selection Secondary Ethernet interface setting 0 Secondary Ethernet interface off disabled 1 auto negotiate 2 10Mbps half duplex 3 10Mbps full duplex 4 100Mbps half duplex 5 100Mbps full duplex and for post Release S1 0 use 6 1000Mbps full duplex if supported by the hardware VLAN identifier used by frames received on or forwarded to the secondary Ethernet interface Set this parameter only when VLAN separation is 1 enabled Value is 1 4 ASCII numeric digits from 0 to 4094 Null is not a valid value nor can the value contain spaces Flag to activate Power over Ethernet conservation mode Valid values are 0 the telephone does not support power conservation mode 1 the telephone indicates support of power conservation mode by transmission of LLDP frames with appropriate indication in Avaya Extreme proprietary PoE Conservation Support Level TLV The telephone supports power conservation mode if requested by reception of an LLDP frame with Avaya Extreme proprietary PoE Conservation Level Request 13 of 21 86 9
35. station being configured change off pbx telephone station mapping XXXXXX where XXXXXX represents the extension number of the station being configured Phone Number Trunk Selection Configuration Set Call Limit the other appropriate field values for example the Trunk Selection value indicates the SIP trunk group The Configuration Set value can reference a set that has the default settings in Communication Manager Change the call limit to match the number of call appr entries in the Add Station form 4 of 4 Administering Stations This section refers to Communication Manager CM administration on the Switch Administration Terminal SAT or by Avaya Site Administration Administer the following items on the Station form sample screens of which are provided in Figure 1 through Figure 4 Avaya recommends setting the features covered in this section because they optimize the user interface 48 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Stations Administering Features The following buttons can be administered for a 9600 Series SIP IP Telephone unless otherwise noted Administrable Station Features Feature Administration Notes Audix One Touch Recording Auto Callback Autodial Bridged Call Appearances Busy Indicator Call Appearances Call Forward all Call Forwarding busy don t answer Call Park Call Unpark Call Pickup CPN Blo
36. string Destination TCP port used for requests to https server 0 65535 The default is 443 IP Address es or DNS name s of Avaya file server s used to download configuration files Note Transport Layer Security is used to authenticate the server Number of seconds to wait fora DHCPOFFER on a non zero VLAN The default is 60 seconds Issue 2 December 2007 55 Server Administration DHCP Generic Setup This document is limited to describing a generic administration that works with the 9600 Series SIP IP Telephones Three DHCP software alternatives are common to Windows operating systems e Windows NT 4 0 DHCP Server e Windows 2000 DHCP Server e Windows 2003 DHCP Server Any other DHCP application might work It is the responsibility of the customer to install and configure the DHCP server correctly DHCP server setup involves 1 Installing the DHCP server software according to vendor instructions 2 Configuring the DHCP server with e IP Addresses available for the 9600 Series SIP IP Telephones e The following DHCP options Option 1 Subnet mask As described in Table 3 item 3 Option 3 Gateway router IP Address es As described in Table 3 item 1 If using more than one address the total list can contain up to 127 total ASCII characters You must separate IP Addresses with commas with no intervening spaces Option 6 DNS server s address list If using more than one address the tota
37. trace is enabled If the LOCAL_LOG_ LEVEL is set to Warning or lower this parameter would enable low level traces from the adaptors or manager as indicated Applies to all logging mechanisms syslog and local log Example ALSIP SESSION enables debug level traces from the ALSIP adaptor and Session manager List of custom logo definitions used as background on display Each logo tuple is delimited by commas Each logo tuple contains logo label verbatim label displayed on the screen and logo URL Logo label and URL are separated from one another by a String maximum of 1023 characters Syslog server IP or DNS address 0 to 255 characters zero or one IP Addresses in dotted decimal or DNS name format This parameter sets the cryptosuite and session parameters for SRTP The parameter can have one or two of the following nine values Separated by commas without any intervening spaces 1 aescm128 hmac80 2 aescm1 28 hmac32 3 aescm1 28 hmac80 unauth 4 aescm128 hmac32 unauth 5 aescm1 28 hmac80 unenc 6 aescm1 28 hmac32 unenc 7 aescm1 28 hmac80 unenc unauth 8 aescm1 28 hmac32 unenc unauth 9 none Voice mail system telephone extension number Specifies the number to be dialed automatically when the telephone user presses the Message button Maximum Transmission Unit size Range is 1496 or 1500 only octets 10 of 21 Issue 2 December 2007 83 Administering Telephone Options Table 11 9600 Series SIP IP Telephones
38. used in HTTP and HTTPS get operations during initialization Value 0 127 ASCII characters no spaces Null is a valid value Leading or trailing slashes are not required The command syntax is GET HTTPDIR myhttpdir where myhttpdir is your HTTP server path HTTPDIR is the path for all HTTP operations 7 of 21 80 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Default Value Parameter Name Description and Value Range wo Null DOMAINS HTTPPORT 80 HTTPPROXY Null HTTPSRVR 0 0 0 0 ICMPDU 1 ICMPRED 0 INTER_DIGIT_TIMEOUT 5 IPADD 0 0 0 0 L2Q 0 L2QAUD 6 L2QSIG 6 Domains to be excluded for SCEP String representing zero or one domains in a URL of 0 to 255 characters in dotted decimal or DNS name format with multiple domains delimited by commas Destination TCP port used for requests to the HTTP server during initialization Range is 0 65535 Note For SIP Release 1 0 there should be no need to set this parameter to values other than default value Zero or one IP or DNS address of the HTTP server for SCEP 0 to 255 characters in dotted decimal or DNS name format followed by a colon and port number The colon and port number are optional If this parameter is not null this proxy transport address is used to set up the HTTP
39. used in the operation of 9600 Series IP Telephones Knowing these ports or ranges helps you administer your networking infrastructure Note In many cases the ports used are the ones called for by IETF or other standards bodies Many of the explanations in Table and Table refer to configuration parameters or options settings For more information about parameters and settings see Administering Options for the 9600 Series SIP IP Telephones Table 5 Received Packets Destination SIP IP Telephone Destination Port Source Port Use UDP or TCP The number used in the Any Received DNS messages UDP Source Port field of the DNS query sent by the telephone The number used in the Any Packets received by the TCP Source Port field of the telephone s HTTP client packets sent by the telephone s HTTP client The number used in the Any TLS SSL packets received TCP Source Port field of the TLS by the telephone s HTTP SSL packets sent by the client telephone s HTTP client 68 Any Received DHCP messages UDP The number used in the Any Received SNTP messages UDP Source Port field of the SNTP query sent by the telephone 161 Any Received SNMP messages UDP 50000 Any Received CNA test request UDP messages 1 of 2 Issue 2 December 2007 31 Network Requirements Table 5 Received Packets Destination SIP IP Telephone continued Destination Port Source Port Use UDP or TCP The number used in the Any Receiv
40. 00 Series SIP IP Telephone Release S2 0 Feature Support Delivered by CM Feature Avaya Communication Feature Telephone Button FNU Manager 3 Way Conferencing X non Avaya environment 6 way Conference X Bridge Automatic Call Back FNU Cancel Call Forward All Calls FNU Call Forward Busy X FNU Don t Answer non Avaya environment 1 of 3 42 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Telephone Administration Table 7 9600 Series SIP IP Telephone Release S2 0 Feature Support continued Delivered by CM Feature Avaya Communication Feature Telephone Button FNU Manager Call Forward FNU Deactivation Call Forward X Unconditional non Avaya environment Call Hold X Call Management X incoming outgoing call screening Call Park and Unpark FNU Call Pick Up Group FNU Call Pickup Directed FNU Call Pickup Extended FNU Group Calling Party Number FNU Block Unblock Consultation Hold Directed Call Pick Up X FNU Distinctive Alerting EC500 Enable FNU EC500 Disable FNU Extend Call for EC500 FNU Available a i Release Extended Group Call Pickup Find Me X Group Call Pickup FNU 2of3 Issue 2 December 2007 43 Communication Manager Administration Table 7 9600 Series SIP IP Telephone Release S2 0 Feature Support continued Delivered by CM Feature Avaya Communication Feature Telephone Button FNU Manager Last Number Dialed X Redial Malicious Call Trace No FNU Message W
41. 00222 eee Chapter 6 Server Administration 0 00002 wee Sotware Checklist crs 64 oe 2 OS SOS OS OS CES SS SE OSE SSH DHCP and File Servers 2 dd ea ee wh a a EE a DHCP Server Administration 1 0 ee e Configuring DHCP for 9600 Series SIP IP Telephones DHCP Generlie Setup lt s cacan eee bee ee dec ee eee eee 4 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 31 34 35 37 37 37 37 38 38 39 39 39 40 40 40 40 40 41 41 41 42 42 44 48 49 51 51 51 53 53 53 54 54 56 Contents Windows NT 4 0 DHCP Server 2 0 eee ee 59 Verifying the Installation of the DHCP Server 2 2 59 Creating a DHCP Scope for the IP Telephones 5 5 60 Editing Custom Options lt 24 4082 eee es eae ke we ee 61 Adding the DHCP Option 2 2 0 2 eee ee ee es 61 Activating the Leases 2 ee 62 Verifying Your Configuration 2 2 2 ee 62 Windows 2000 DHCP Server 2 2 2 ee eee ee es 63 Verifying the Installation of the DHCP Server 2 63 Adding DHCP Options 2 bss s ssa sados esas ease edes 65 Activating the New Scope sanaaa 65 HTTP Generic Seb 66 oer ch ese Gees one wre a a eens see de 66 Chapter 7 Telephone Software and Binary Files 67 General Download Process 0022 eee eee eee ee ees 67 BONNE 644K RENEE ESR EECA ESE EERE EH ER SESS 67 9600 Series SIP IP Telephone Scripts and Bi
42. 2Q 0 the telephone sets L2QVLAN 0 and transmits DHCP messages without tagging e If VLANTEST is 0 the timer will never expire Note Regardless of the setting of L2Q VLANTEST or L2QVLAN you must have DHCP administered so that the telephone will get a response to a DHCPDISCOVER when it makes that request on the default 0 VLAN After VLANTEST expires if a 9600 Series SIP IP Telephone receives a non zero L2QVLAN value the telephone will release the IP Address and send DHCPDISCOVER on that VLAN Any other release will require a manual reset before the telephone will attempt to use a VLAN on which VLANTEST has expired See the Reset procedure in Chapter 3 of the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide The telephone ignores any VLAN ID administered on the Communication Manager call server VLAN Default Value and Priority Tagging The system value L2ZQVLAN is initially set to 0 and identifies the 802 1Q VLAN Identifier This default value indicates priority tagging as defined in IEEE 802 IQ Section 9 3 2 3 Priority tagging specifies that your network closet Ethernet switch automatically insert the switch port default VLAN without changing the user priority of the frame cf IEEE 802 1D and 802 1Q The VLAN ID 0 zero is used to associate priority tagged frames to the port native VLAN of the ingress port of the switch But some switches do not understand a VLA
43. 600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Description and Value Range Value PRESENCE_SERVER Null List of Presence Server IP or DNS address es This value is used to access the server for presence indications in case of several entries first address always first etc In some environments the SIP proxy registrar may be different than the presence server In this case the presence server is set via this parameter If both functions are provided by the same server it is not necessary to set PRESENCE_SERVER the SIP proxy server is accessed for server based presence indications Zero 0 to 255 characters zero or more IP addresses in dotted decimal or DNS name format separated by commas without any intervening spaces When operating in a non Avaya environment this value is set via a SET command in the settings file If this value is not set the SIP Proxy server address is used When not set via settings file this value is retrieved via PPM PROCPSWD 27238 Text string containing the local dialpad procedure password Null or 1 7 ASCII digits If set password must be entered immediately after accessing the Craft Access Code Entry screen either during initialization or when Mute or Contacts for the 9610 is pressed to access a craft
44. 600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Document Organization Document Organization The guide contains the following sections Chapter 1 Introduction Provides an overview of this document Chapter 2 Administration Provides an overview of the administrative process and describes Overview and Requirements general hardware software and operational requirements Chapter 3 Network Describes administrative requirements for your Local Area Requirements Network Chapter 4 Communication Describes how to administer Avaya Communication Manager to Manager Administration operate with 9600 Series SIP IP Telephones Chapter 5 SIP Enablement Covers SIP Enablement Services SES configuration for 9600 Services SES Administration Series SIP IP Telephones Chapter 6 Server Administration Describes DHCP and HTTP HTTPS administration for the 9600 Series IP Telephones Chapter 7 Telephone Software Describes telephone software covers software downloads and and Binary Files provides information about the configuration file Chapter 8 Administering Describes how to use file parameters and options to administer Telephone Options 9600 Series SIP IP Telephones Covers backup and restoration of telephone data Also describes how to use local procedures to customize a single telephone from the dialpad Chapter 9 Administering Describes customizeable application specific parameters to
45. AVAYA Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Administrator Guide Release 2 0 16 601944 Issue 2 December 2007 2007 Avaya Inc All Rights Reserved Notice While reasonable efforts were made to ensure that the information in this document was complete and accurate at the time of printing Avaya Inc can assume no liability for any errors Changes and corrections to the information in this document may be incorporated in future releases For full legal page information please see the complete document Avaya Legal Page for Hardware Documentation Document number 03 600759 To locate this document on our Web site simply go to http www avaya com support and search for the document number in the search box Documentation disclaimer Avaya Inc is not responsible for any modifications additions or deletions to the original published version of this documentation unless such modifications additions or deletions were performed by Avaya Customer and or End User agree to indemnify and hold harmless Avaya Avaya s agents servants and employees against all claims lawsuits demands and judgments arising out of or in connection with subsequent modifications additions or deletions to this documentation to the extent made by the Customer or End User Link disclaimer Avaya Inc is not responsible for the contents or reliability of any linked Web sites referenced elsewhere within this documentation
46. Administration Administration Enhancements SIP Software Release 2 0 supports functionality introduced on Avaya Communication Manager Release 5 0 and SIP Enablement Services SES Release 5 0 Visiting User Support Visiting user support allows users to easily move between geographic locations while retaining their telephone extension and settings 9600 Series SIP IP Telephones can be provisioned through the settings file VU_MODE configuration parameter to one of three modes e No Visiting User the telephone operates normally and has no user interface impact for normal operation The telephone can be forced to a registered Inactive state when a visiting user registers elsewhere e Optional Visiting User the telephone prompts the user at registration time if they are visiting or not e Forced Visiting User the telephone allows only visiting user registrations For more information see Visiting User Administration Link Layer Discovery Protocol LLDP 9600 Series SIP IP Telephones now support link layer discovery protocol See Link Layer Discovery Protocol LLDP for information 802 1X 9600 Series SIP IP Telephones now support IEEE standard 802 1X for increased security The new configuration parameter DOT1X defines the 802 1X operational mode The new parameter DOT1XSTAT enables disables 802 1X The new parameter DOT1XEAPS specifies the authentication method to use with 802 1X These parameters can be set throu
47. Chapter 2 Administration Overview and Requirements 15 9600 Series IP Telephones 2 2 2 c25 22284 eee eS Sb ee eerewe ces 15 Parameter Data Precedence 2 2 2 eee eee ee eee ee es 18 The Administrative Process 1 eet 19 Administrative Checklist 1 2 ee 19 Telephone Initialization Process 2 ee 21 Step 1 Telephone to Network 2 0 0 eee eee ee ee 21 Step 2 Telephone to LLDP Enabled Network 2588585 21 Step 3 Telephone to DHCP Server 2 00 ee eee eee ees 22 Step 4 Telephone and File Server 2 0 02 2 eee eee eee 22 Step 5 Telephone and the SES Server 2 08 ee ee eee 22 Error GOMniOne s s e 6 cas 4h E ee 6 HG 6S ee a ES eS Se 23 Chapter 3 Network Requirements 2 20 22280885 25 Network Assessment aoaaa aa ee 25 Hardware Requirements 2 2 eee 25 Server Requirements 2 6 ee ee ee 26 DHCP Sevels cuce a amp amp rel Og Roe oe ares So eons 6 5 aces S ice 6 Sea 26 HTIPHTIPS SEVE ss bbe GS SOS Ch OOO SE SSS YS EEE OCDE 27 Network Time Protocol NTP Server 1 0 ce eee eee es 27 Required Network Information 2 6 eee 27 Other Network Considerations 1 2 ee 28 SNMP s aa ranea ee De ee a Se Oe 28 Registration and Authentication 2 002282 ee eee 28 Reliability and Performance 2 6 eee ee ee 29 ee aes See aA ae ce he ks th OS a a ea 29 IEEE 802 1D and G02 1G gt o kk cde Oe HVE REL O
48. DED EROS 29 Network Audio Quality Display on 9600 Series SIP IP Telephones 30 SIP Station Number Portability 2 2 30 Issue 2 December 2007 3 Contents TCP UDP Port Utilization 2 22 ce ec cee wee ee eee ee RO a Soar 15s et oe a e a E R Seti Se ec ae Gee we ae Registration and Authentication 2 eee eee ee es Chapter 4 Communication Manager Administration Call Server Requirements 2 26 ee et switch Compatibility s aos 44664 44 4056s 624 ea OS Communication Manager Administrative Requirements System Level Preparation Tasks 1 2 eee eee ee ee es SIP Trunk Administration 6 444 424 eee HR ORR ERD RES Call Routing Administration 2 2 2 ee ee IP Interface and Addresses 2 eee UDP Port Selection 1 41644 44 4 2 OA EMD Aaea RSVP atid RTCP SRTCP basses hoe OR we HED Oe OOS Voice Mail Integration 2266 acc He ee we we Le we wee Aulo Hold o be ea EDA ERS ESTEE TS SOE HE AEG SH KO Call Transfer Considerations 2 6 2 Conferencing Call Considerations 2 2 2 eee ee ee ee es Telephone Administration 6 CM SIP IP Telephone Configuration Requirements 4 Administering Stations 2 2 ee Administering Features 2 ee ee es Chapter 5 SIP Enablement Services SES Administration IMFOQUCUON 64 05 6h ceo SK Ske RDM PRES OES Oe BREE Using the Web Browser to Configure SES 002
49. E PRODUCT S TO THE POINT OF PURCHASE WITHIN TEN 10 DAYS OF DELIVERY FOR A REFUND OR CREDIT Avaya grants End User a license within the scope of the license types described below The applicable number of licenses and units of capacity for which the license is granted will be one 1 unless a different number of licenses or units of capacity is specified in the Documentation or other materials available to End User Designated Processor means a single stand alone computing device Server means a Designated Processor that hosts a software application to be accessed by multiple users Software means the computer programs in object code originally licensed by Avaya and ultimately utilized by End User whether as stand alone Products or pre installed on Hardware Hardware means the standard hardware Products originally sold by Avaya and ultimately utilized by End User License Type s Designated System s License DS End User may install and use each copy of the Software on only one Designated Processor unless a different number of Designated Processors is indicated in the Documentation or other materials available to End User Avaya may require the Designated Processor s to be identified by type serial number feature key location or other specific designation or to be provided by End User to Avaya through electronic means established by Avaya specifically for this purpose Third party Components Certain soft
50. ES Server to register and authenticate it Registration is described in the Initialization process in Step 5 Telephone and the SES Server on page 22 For further information see nstalling and Administering SIP Enablement Services R 4 0 03 600766 available on the Avaya support Web site http Awww avaya com support 28 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Other Network Considerations Reliability and Performance All 9600 Series SIP IP Telephones respond to a ping or traceroute message sent from Avaya Communication Manager or any other network source The telephones do not originate a ping or traceroute The 9600 Series SIP IP Telephones offer and support remote ping and remote traceroute The switch can instruct the telephone to originate a ping or a traceroute to a specified IP Address The telephone carries out that instruction and sends a message to the switch indicating the results For more information see your switch administration documentation If applicable the telephones test whether the network Ethernet switch port supports IEEE 802 1D q tagged frames by ARPing the router with a tagged frame For more information see VLAN Considerations on page 94 If your LAN environment includes Virtual LANs VLANs your router must respond to ARPs for VLAN tagging to work properly QoS For more information about the extent to which your network can support any or all of the
51. IEEE Documents oaoa 119 Applications and Options Administering 11 Applications Customizing 111 Application specific parameters administering 17 Assessment of Network aoaaa aa 25 B Backup Restore aoao aoa a 0 000 110 Binary File and Upgrade Script Choosing 68 Binary Files 2 2 2 2 0 0 0 0 0008 68 Binary Files and Telephone Software 67 Binary Files and Scripts for 9600 Series SIP IP TIGPMONCS see a de ce eee eee ee ia 68 Browser Administering 44 112 C Call Forward administration 2 49 Call Server Requirements ay Call Transfer Considerations 41 Checklist Administrative 19 CM SIP Configuration Requirements 45 Communication Manager Administration 37 Communication Manager Administrative Requirements 0 0505 37 Communication Manager SIP IP Telephone Configuration Requirements 44 Conferencing Call Considerations 42 Configuration Requirements CM SIP 44 45 Configuring SES Using the Web Browser Si Contents of the Settings File 70 Customizeable System Parameters 74 Customizing 9600 Series IP Telephone Applications and Options 2 111 D DHCP and File Servers 53 DHCP Generic Setup a aoao 56 DHCP options ooa a 56 DHCP Parameters Setby 55 DHCP Server so oa
52. N ID of zero and require frames tagged with a non zero VLAN ID If you do not want the default VLAN to be used for voice traffic e Ensure that the switch configuration lets frames tagged by the 9600 Series SIP IP Telephone through without overwriting or removing them e Set the system value L2QVLAN to the VLAN ID appropriate for your voice LAN Issue 2 December 2007 95 Administering Telephone Options Another system value you can administer is VLANTEST VLANTEST defines the number of seconds the 9600 IP Series Telephone waits fora DHCPOFFER message when using a non zero VLAN ID The VLANTEST default is 60 seconds Using VLANTEST ensures that the telephone returns to the default VLAN if an invalid VLAN ID is administered or if the phone moves to a port where the L2QVLAN value is invalid The default value is long allowing for the scenario that a major power interruption is causing the phones to restart Always allow time for network routers the DHCP servers etc to be returned to service If the telephone restarts for any reason and the VLANTEST time limit expires the telephone assumes the administered VLAN ID is invalid The telephone then initiates registration with the default VLAN ID Setting VLANTEST to 0 has the special meaning of telling the phone to use a non zero VLAN indefinitely to attempt DHCP In other words the telephone does not return to the default VLAN Note If the telephone returns to the default VLAN b
53. N is forwarded to the telephone Add commands to the 46xxsettings ixt file to enable VLAN separation The following example assumes the data VLAN ID is yyy and the data traffic oo priority is z SET VLANSEP 1 SET PHY2VLAN yyy SET PHY2PRIO z Note Also configure the network switch so that 802 1Q tags are not removed from frames forwarded to the telephone Issue 2 December 2007 71 Telephone Software and Binary Files The GROUP System Value You might have different communities of users all of which have the same telephone model but which require different administered settings For example you might want to group users by time zones or work activities Use the GROUP system value for this purpose 1 identify which telephones are associated with which group and designate a number for each group The number can be any integer from 0 to 999 with 0 as the default meaning your largest group is assigned as Group 0 2 At each non default telephone instruct the installer or user to invoke the GROUP Craft Local procedure as specified in the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide and specify which GROUP number to use The GROUP System value can only be set on a phone by phone basis 3 Once the GROUP assignments are in place edit the configuration file to allow each telephone of the appropriate group to download its proper settings Here is an example of a se
54. O RTCP disabled 1 RTCP enabled RTCP Monitor IP or DNS address to be used as destination for RTCP monitoring Zero to 255 characters zero or one IP addresses in dotted decimal or DNS name format Note that this value is only set via SET command in settings file if operating in a NON Avaya environment otherwise this value is retrieved via PPM RTCP monitor port number TCP UDP port to be used as destination port for RTCP monitoring Valid range is 0 65535 Note that this value is only set via SET command in settings file if operating in a NON Avaya environment otherwise this value is retrieved via PPM Specifies lower limit of a port range to be used by RTP RTCP or SRTP SRTCP connections for example to adapt to firewall traversal policies Values 1024 65503 Specifies the width of the port range to be used by RTP RTCP or SRTP SRTCP connections for example to adapt to firewall traversal policies The upper limit is calculated by the value of RTP_PORT_LOW plus the value of RTP_PORT_RANGE taking into consideration the overall limit of 65535 Values 32 64511 Number of idle time minutes after which the screen saver is turned on Valid values range from zero disabled to 999 minutes 16 65 hours Defines whether DTMF tones are send in band regular audio or out band negotiation and transmission of DTMF according to RFC 2833 with fallback to send in band DTMF tones if far end does not support RFC2833 Values are 1 in band DTMF 2
55. P Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range MYCERTWATI 1 NETMASK 0 0 0 0 NO_DIGITS_TIMEOUT 20 OUTBOUND _ 86400 SUBSCRIPT ON_ REQUEST DURATION PHNEMERGNUM Null PHNCC 1 PHNDPLENGTH 5 PHNIC 011 PHNLD 1 PHNLDLENGTH 10 PHNOL 9 Flag defining phone s behavior when performing certificate enrollment Values are O wait until a certificate or a denial is received or a pending notification is received 1 periodical check in the background IP subnet mask Range is 7 to 15 ASCII characters defining one IP Address in dotted decimal format Number of seconds of delay after going off hook or getting secondary dial tone before phone automatically plays a warning tone and does not accept dial input any longer Range in seconds is 1 to 60 Number of seconds used in initial SUBSCRIBE messages This is the suggested duration value of the telephone which might be lowered by the server depending on the server configuration Range is 60 31536000 Note that the default value is equal to one day and the maximum value represents one year The number dialed when the Emerg softkey is pressed or when a pop up screen for making an emergency call is confirmed Telephone country code The administered international country code for the location by the algorithm that dials calls from the incoming Call Log or from Web pages Range 1 3 digits from 1 to 999
56. P keep alive message TCP ACK message to the far end The time is controlled by the system s TCP IP stack The timer is restarted after application level data for example a SIP message is sent over the socket When the system is idle this keep alive time expires and results in sending a TCP ACK keep alive packet Valid values are 10 3600 seconds Display time according to defined format in the top line and in the call log Values are 0 am pm format 1 24h format Path name for https downloads Character string of 0 to 127 characters representing a directory name or path to directory Destination TCP port used for requests to https server during initialization Values 0 65535 Flag to indicate if TLS server identification is required Valid values are 0 no certificate match necessary TLS SSL connection will be established anyway 1 certificate match required TLS SSL connection will only be established if the server s identity matches the server s certificate File names of certificates to be used for authentication List of file names separated by commas 0 to 1024 characters Activate deactivate usage of calendar on Microsoft Exchange Server Values are 0 disabled 1 enabled Flag that indicates whether a directional attributes or 0 0 0 0 IP Address is used in the SDP to signal hold operation O use a directional attributes 1 use quad zeros Enables or disables VLAN separation Controls whether frames
57. QoS initiatives see your LAN equipment documentation See QoS on page 40 for QoS implications for the 9600 Series SIP IP Telephones All 9600 Series SIP IP Telephones provide some detail about network audio quality For more information see Network Audio Quality Display on 9600 Series SIP IP Telephones on page 30 IEEE 802 1D and 802 1Q For more information about IEEE 802 1D and IEEE 802 1Q and the 9600 Series SIP IP Telephones see IEEE 802 1D and 802 1Q on page 40 and VLAN Considerations on page 94 Three bits of the 802 1Q tag are reserved for identifying packet priority to allow any one of eight priorities to be assigned to a specific packet oO kf O1 OD O N 7 Voice traffic with less than 10ms latency Voice traffic with less than 100ms latency Controlled load traffic for critical data applications Traffic meriting extra effort by the network for prompt delivery for example executive Network management traffic e mail Reserved for future use The default priority for traffic meriting the best effort for prompt delivery of the network Background traffic such as bulk data transfers and backups Note Priority 0 is a higher priority than Priority 1 Issue 2 December 2007 29 Network Requirements Network Audio Quality Display on 9600 Series SIP IP Telephones All 9600 Series SIP IP Telephones give the user an opportunity to monitor network audio
58. RFC2833 procedure Parameter to allow to download during start up the specific configuration sets for H323 or SIP endpoints Valid values are 0 Default 1 H323 2 SIP Lower limit of port range for signaling to support by the phone Values range from 1024 to 65503 Port range for signaling to support by the phone Values range from 32 to 64511 Determines whether the telephone uses a proxy to receive incoming calls or can receive calls directly from another telephone Values are O proxy mode 1 peer to peer mode 16 of 21 Issue 2 December 2007 89 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range SIP_PORT_SECURE 5061 SIPCONFERENCECONTI 0 NUE SIPDOMAIN Null SIPPORT 5060 SIPREGISTRAR Null SIPROXYSRVR Null SIPSIGNAL 2 Default SIP port for secure message transfer via TLS Values range from 1024 65535 When the ENABLE_AVAYA_ENVIRONMENT parameter is 0 non Avaya environment and the telephone initiating the conference ends the call the other parties will be dropped unless SIPCONFERENCECONTINUE is set to 1 continue conference call without initiator If this parameter is set to 0 the capability is turned off and the phone ends the conference when the initiator hangs up SIP domain name for registration 0 to 255 characters string representing domain name Def
59. S software and Network Time server which controls time related parameters These servers are not necessarily separate hardware units Features amp Functions supported by H 323 9600 Series IP Telephones Not Supported by SIP Button modules are not currently supported by 9600 SIP IP Telephones Backup Restore 9600 Series H 323 IP Telephones use HTTP to store backup files 9600 Series SIP IP Telephones use the Personal Profile Manager PPM functionality within SIP Enablement Services SES for backup and restore functions Settings File amp System Parameters Both SIP and H 323 9600 Series IP Telephones and 4600 Series IP Telephones use the same settings file Some of the same system parameters are used however numerous SIP specific parameters support SIP operation only In H 323 9600 Series IP Telephones the parameters OPSTAT and APPSTAT control all user interface functions whereas 9600 Series SIP IP Telephones use a separate parameter for example ENABLE_CONTACTS ENABLE_CALLLOG for each user interface function 8 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Change History Language Support SIP telephones support the same languages as H 323 telephones with the exception of Hebrew SIP does not support Hebrew or the English Large Text Font for any language Further all SIP language files have xml file extensions whereas H 323 language files have txt file extensions SNMP amp MIBs Although
60. SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range MODE CALLFWDADDR CALLFWDDELAY CALLFWDSTAT CNAPORT CNASRVR CNGLABEL CONFIG_ SERVER CONFIG_SERVER_ SECURE MODE 0 Null 50002 Null Null When ENABLE_AVAYA_ENVIRONMENTS 0 this parameter indicates how transfers are performed 0 attended transfer 1 unattended transfer The URI to which calls are forwarded in 3rd party non Avaya environments only Third party non Avaya environments only Specifies the number of ring cycles generated at the phone before the call is forwarded to the Call Forwarding Address if call forwarding on No answer is selected in 3rd party environments Valid number of ringing cycles are 0 20 Third party non Avaya environments only Specifies the sum of the allowed Call Forwarding permissions This parameter controls which of the Call Forwarding Feature Buttons are made visible and active for the user in 3rd party environments Valid values are 0 no Call Forwarding permitted 1 Call Forward Unconditional only permitted 2 Call Forward Busy only permitted 4 Call Forward No Answer only permitted Others sum of Call Forward types permitted Transport layer port number to be used for registration to CNA server for network analysis Valid range is 0 65535 List of CNA server IP or DNS address es Used to connect to CNA
61. TRICTION CALLING PERMISSION Enter y to grant permission to call specified COR 0 n 15 30 n 44 n 58 1 on 16 n 31 n 45 n 59 2 n 172 n 32 n 46 n 60 3 n 18 n 33 n 47 n 61 4 n 19 n 34 n 48 n 62 5 n 20 n 35 n 49 n 63 6 n 21 n 36 n 50 n 64 TT 227 m 37 n SL cn 65 8 n 23 n 38 n 52 n 66 9 n 24 n 39 n 582 n 67 10 n 2572 40 n 54 n 68 e a 26 n 41 n 552 n 69 12 n 27 n 42 n 56 n 70 13 21 28 n 43 n 57 n 71 14 n 29 n n Depp AasA AA AASB BS Ss 712 713 74 1a 76 TIR 718 79 80 BL 82 83 84 85 Dep pAaA AA PABA AS Page 4 of x 8 672 87 88 89 90 91 92 93 94 95 96 97 98 992 Issue 2 December 2007 n ee e e a e AA AASB Ss Ss 131 Sample Station Forms Figure 22 System Parameters Customer Options Optional Features screen display system parameters customer options G3 Version V12 Location 2 Platform 2 OPTION 123456789012 Maximum Maximum Maximum Maximum Maximum Platfo Maximum Off PBX Te Off PBX Te Off PBX Te Off PBX Te Off PBX Te Pa 4 4 AL FEATURES Software Package RFA System ID SID RFA Module ID MID US rm Maximum Ports 300 7 Maximum Stations 300 17 XMOBILE Stations 30 28 lephones EC500 1200 0 lephones OPS 1200 0 lephones SCCAN 0 0 lephones PBFMC 0 0 lephones PVFMC 0 0 NOTE You must logoff amp login to effe
62. Table 1 indicates that you can administer system configuration parameters in a variety of ways and use a variety of administrative mechanisms like Maintaining the information on the call server Manually entering the information by means of the telephone dialpad Administering the DHCP server Editing the configuration file on the applicable HTTP or HTTPS file server User modification of certain parameters when given administrative permission to do so Note Not all parameters can be administered on all administrative mechanisms 16 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 9600 Series IP Telephones Table 1 Administration Alternatives and Options for 9600 Series SIP IP Telephones Administrative Parameter s Mechanisms For More Information See Telephone Avaya Chapter 4 Communication Manager Administration Administration IP Addresses Tagging and VLAN Network Time Server NTS Quality of Service Interface Application specific parameters Communication Manager and SES DHCP strongly recommended Settings file Manual administration at the telephone LLDP LLDP DHCP Settings file Manual administration at the telephone DHCP Settings file Settings file DHCP Settings file strongly recommended LLDP Manual administration at the telephone DHCP Settings file strongly recommended Chapter 6 Server Administration and Appendix B Related Do
63. Terms Glossary of a a aoaaa AMS TLS i ee ee ee a aoa i 31 34 66 67 90 92 99 TLVs Impact on System Parameter Values 104 U UDP Port Selection 2 2040 39 UDP TCP Port Utilization 2 2 002 231 Upgrade Script and Binary File Choosing the Right 68 Upgrade Script File 0 69 Upgrade Script contents of 70 V Visiting User Administration aoaaa aa 105 VLAN Considerations o oo a 94 VLAN Default Value 2 aaa 95 VLAN Detection aooaa a 95 VLAN Separation aooaa a a 96 VLAN Separation Rules o oo 97 VLAN Tagging a aaa a 94 Voice Mail Integration 41 W What s New 2 2 00040 10 13 WML Browser Administering 112 Issue 2 December 2007 137 Index 138 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0
64. This file is where you identify non default option settings application specific parameters and so on You can download a template for this file from the Avaya support Web site An example of what the file might look like follows 70 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 9600 Series SIP IP Telephone Scripts and Binary Files Note The following is intended only as a simple example Your settings will vary from the settings shown This sample assumes specification of a DNS Server identifying SIP specific settings and setting the time date SET DNSSRVR dnsexample yourco com SET SIPPROXYSRVR 192 168 1 110 SET SIPSIGNAL 1 TCP SET ENABLE_PRESENCE 1 show presence icons SET SIPDOMAIN domain name SET SNTPSRVR 192 168 1 111 SET GMTOFFSET 5 00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L second Sunday in March at 2 am Local time SET DSTSTOP 1SunNov2L first Sunday in November at 2 am Local time Note that the DSTSTART and DSTSTOP parameters reflect the new 2007 Daylight Savings Time values for North America See Chapter 8 Administering Telephone Options for details about specific values You need only specify settings that vary from defaults although specifying defaults is harmless VLAN separation controls whether or not traffic received on the secondary Ethernet interface is forwarded on the voice VLAN and whether network traffic received on the data VLA
65. X 2 the telephone can support authentication from the switch The attached PC in this scenario gains access to the network without being authenticated e Telephone with attached PC PC Only Authenticates When the telephone is configured for Pass Through Mode or Pass Through Mode with Logoff DOT1X 0 or 1 an attached PC running 802 1X supplicant software can be authenticated by the data switch The telephone in this scenario gains access to the network without being authenticated Some switches support authentication of multiple devices connected through a single switch port This is known as multi supplicant or MAC based operation These switches typically send unicast 802 1X packets to authenticating devices These switches support the following two scenarios e Standalone telephone Telephone Only Authenticates When the telephone is configured for Supplicant Mode DOT 1X 2 the telephone can support authentication from the switch When DOT1X is 0 or 1 the telephone is unable to authenticate with the switch e Telephone and PC Dual Authentication Both the telephone and the connected PC can support 802 1X authentication from the switch The telephone may be configured for Pass Through Mode or Pass Through Mode with Logoff DOT1X 0 or 1 The attached PC must be running 802 1X supplicant software 100 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Link Layer Discovery Protocol LLDP Link Layer Discovery P
66. able G711U codec capability of the phone If the parameter is set to 1 the phone includes G711U capability in an outbound INVITE request and accepts G711U when received in an incoming INVITE request Values are O disabled 1 enabled Enable or disable G722 capability of the telephone If the parameter is set to 1 the phone includes G722 capability in an outbound INVITE request and accepts G722 when received in an incoming INVITE request If set to 0 processing of G722 as a capability is disabled Values are O disabled off 1 enabled on Enable or disable G726 capability of the telephone If the parameter is set to 1 the telephone includes G726 capability in an outbound INVITE request and accepts G726 when received in an incoming INVITE request Values are O disabled off 1 enabled on Enable or disable G729A codec capability of the phone Values are 0 G729A disabled 1 The phone includes G729 A without Annex B support capability in an outbound INVITE request and accepts G729 when received in an incoming INVITE request 2 The phone includes G729 A with Annex B support capability in an outbound INVITE request and accepts G729 when received in an incoming INVITE request Enable or disable the ability to modify contacts if the Contact application is enabled Values are O disabled 1 enabled Activate deactivate multiple contacts warning Depending on current value a warning message is displayed explaining to the user there are
67. aiting X Indication Music on Hold X One Touch Recording X Priority Call FNU Send All Calls Enable FNU Disable Transfer attended X non Avaya environment Transfer unattended X one button transfer non Avaya environment Transfer to Voice Mail FNU Whisper Page X 3 of 3 CM SIP IP Telephone Configuration Requirements This section refers to Communication Manager CM administration on the Switch Administration Terminal SAT or by Avaya Site Administration The system wide CM form and the particular page that needs to be administered for each feature are provided These features which already exist are not required but are recommended because they optimize the telephone user interface CM 4 0 or greater is required For sample Station and other pertinent forms see Appendix C Sample Station Forms 44 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Table 8 CM SIP Configuration Requirements Task Form Command CM SIP IP Telephone Configuration Requirements Field s Value s IP Network Region IP Network Region Off PBX Telephones Station Mapping Feature Related System Parameters page 1 Feature Related System Parameters page 4 Feature Related System Parameters page 4 Feature Related System Parameters page 17 Define the dial plan formats on the Dialplan Analysis Table form Define the access codes corresponding to the OPS FNEs on the Feature Access
68. aling is off PHNCC the international country code of the Communication Manager CM call server For example 1 for the United States 44 for the United Kingdom and so on PHNDPLENGTH the length of the dial plan on the CM call server PHNIC the digits the CM call server dials to access public network international trunks For example 011 for the United States PHNLD the digit dialed to access public network long distance trunks on the CM call server PHNLDLENGTH the maximum length in digits of the national telephone number for the country in which the CM call server is located PHNOL the character s dialed to access public network local trunks on the CM call server Note In all cases the values you administer are the values relevant to the location of the CM call server at which the IP telephones are registered If a telephone is in Japan but its CM call server is in the United States set the PHNCC value to 1 for the United States In all cases the digits the telephones insert and dial are subject to standard CM call server features and administration This includes Class of Service COS Class of Restriction COR Automatic Route Selection ARS and so on As indicated in Table 11 you can administer the system parameter ENHDIALSTAT to turn off the Enhanced Local Dialing feature Example A corporate voice network has a 4 digit dialing plan The corporate WML Web site lists a 4 digit t
69. allation and for full service the user can enter the telephone extension and the SES password For a restart of an existing installation this information is already stored on the telephone but the user might have to confirm the information The telephone and SES and SES and CM exchange more messaging The expected result is that the telephone is appropriately registered and CM call server data such as feature button assignments are downloaded For more information about the installation process see the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide 22 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Error Conditions Error Conditions Assuming proper administration most of the problems reported by telephone users are likely to be LAN based Quality of Service server administration and other issues can impact user perception of IP telephone performance The Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide covers possible operational problems that might be encountered after successful 9600 Series SIP IP Telephone installation The User Guides for a specific telephone model also contain guidance for users having problems with specific IP telephone applications Issue 2 December 2007 23 Administration Overview and Requirements 24 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 3 Network Requirement
70. am default ENHANCED CALL FORWARDING Forwarded Destination Active Unconditional For Internal Calls To n External Calls To n Busy For Internal Calls To n External Calls To n No Reply For Internal Calls To n External Calls To n SAC CF Override n 122 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Figure 4 Station Form Site Data Feature Button Assignments Voice Mail add station nnnn ABBR BUTTON ASSIGNM 1 OF WN SITE DATA Room Jack Cable Floor Building EVIATED DIALING Listl call appr call appr call appr audix rec release ENTS 4000 Ext voice mail Number STATION List2 o o _ limit call team Ext 5381231 cfwd enh Ext cfwd enh Ext 5502 aux work RC 1 Group Page 4 of X Headset Speaker Mounting Cord Length Set Color __ oas5 5 Rg Figure 5 Station Form Additional Feature Button Assignments FEAT URE BUTTON ANA OFWNR OC DM NMNNMND t BPwWNHrF CW change station nnnn ASSIGNMENTS STATION Page 5 of x Issue 2 December 2007 123 Sample Station Forms Figure 6 SIP Feature Options change station nnnn SIP Feature Options Type of 3PCC Enabled Page 5 of x STATION none Figure 7 Feature Related System Parameters Form change system parameters featu res page 11 of x FEATUR E RELATED SYSTEM PARAMETERS
71. anaging VoIP traffic in your LAN Avaya recommends that you establish a voice VLAN set L2QVLAN to that VLAN and provide voice traffic with priority over other traffic You can set VLAN tagging manually by DHCP or in the 46xxsettings txt file If VLAN tagging is enabled L2Q 0 or 1 the 9600 Series SIP IP Telephones set the VLAN ID to L2QVLAN and the VLAN priority for packets from the telephone to L2QAUD for audio packets and L2QSIG for signalling packets The default value 6 for these parameters is the recommended value for voice traffic in IEEE 802 1D Regardless of the tagging setting a 9600 Series SIP IP Telephone will always transmit packets from the telephone at absolute priority over packets from secondary Ethernet The priority settings are useful only if the downstream equipment is administered to give the voice VLAN priority 94 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 VLAN Considerations VLAN Detection The Avaya IP Telephones support automatic detection of the condition where the LZQVLAN setting is incorrect When VLAN tagging is enabled L2Q 0 or 1 initially the 9600 Series SIP IP Telephone transmits DHCP messages with IEEE 802 1Q tagging and the VLAN set to L2QVLAN The telephones will continue to do this for VLANTEST seconds e f the VLANTEST timer expires and L2Q 1 the telephone sets L2QVLAN 0 and transmits DHCP messages with the default VLAN 0 e f the VLANTEST timer expires and L
72. aracter string applies an algorithm and determines the string of digits to be sent to Avaya Communication Manager CM for dialing At this point the Phone application goes off hook and sends the digits to CM The Source Flag has two possible values e Yes the Called Party Number has been administered or otherwise identified as a valid outgoing phone number such as a Speed Dial button Redial number or Outgoing Call Log number or e No the Called Party Number comes from a source that is likely to require enhanced local dialing processing for example the Incoming Call Log application Note The Enhanced Local Dialing algorithm requires that telephone numbers be presented in a standard format The standard format depends on how you administer the parameters indicated in Table 11 The algorithm also assumes that international telephone numbers are identified as such in for example the Contacts application Precede international numbers with a plus sign anda space or some non digit character following the country code 110 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 9 Administering Applications and Options Customizing Telephone Applications and Options This chapter covers configuration options for activating deactivating options and applications The 9600 Series SIP IP Telephones offer the user numerous applications like Contacts Call Log Redial and so on Each of these applications allows t
73. at the Maximum Administered SIP Trunks are set correctly on the System Parameters Customer Options screen page 2 e Setting the OPS SIP station capacity on the System Parameters Customer Options screen page 1 e Setting the IP Node name for SES on the IP Node Names screen e Entering the IP Address and host name for the administered SES server on the IP Address Mapping screen e Setting the Authoritative Domain on the IP Network Region screen e Setting the intra and inter region IP IP Direct Audio to yes on the IP Network Region screen e Setting the Signaling Group on the Signaling Group screen page 1 SIP Trunk Administration SIP trunk administration tasks include e Setting the SIP Intercept Treatment and Trunk to Trunk Transfer on the System Parameters Features screen page 1 e Administering Trunk Groups on the Trunk Group screens pages 1 through 4 e Assigning public unknown numbering data on the Numbering Public Unknown Numbering screen e Assigning a SIP phone Set description on Configuration Set screen 38 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Communication Manager Administrative Requirements Call Routing Administration Call routing administration includes e Administering Feature Access Codes FACs on the Feature Access Code screen e Administering the ARS Digit Analysis Table on the ARS Digit Analysis Table screen e Administering the Route Pattern on the Route Pattern
74. ault SIP port for non secure message transfer only Values range from 1024 65535 List of SIP registrar server IP or DNS address es Server s used to address SIP registrations if operating in proxy mode In case of several entries the first address always first etc In some third party environments the SIP proxy and SIP registrar may be different servers In this case the SIP registrar will be set using SIPREGISTRAR If both functions are provided by the same server it is not necessary to set the SIPREGISTRAR i e this value will remain null Zero to 255 characters zero or more IP addresses in dotted decimal or DNS name format separated by commas without any intervening spaces Only set via SET command in settings file if you are operating in a non Avaya environment In an Avaya environment this value is not applicable because this is always identical to SIPPROXYSRVR SIP proxy registrar server IP or DNS address Zero or one IP Address Format is dotted decimal or DNS format separated by commas with no spaces 0 255 ASCII characters including commas SIP signaling transport protocol Values are 0 UDP 1 TCP 2 TLS over TCP 17 of 21 90 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range
75. ay Enhanced List ASG Analog Trunk Incoming Call ID A D Grp Sys List Dialing Start at 01 Answer Supervision by Call Classifier ARS ARS AAR Partitioning ARS AAR Dialing without FAC ASAI ASAI Async Transfer Transfer Mode Mode Link Core Capabilities Link Plus Capabilities ATM PNC page 3 of x Audible Message Waiting Authorization Codes CAS Branch CAS Main Change COR by FAC Computer Telephony Adjunct Links Cvg Of Calls Redirected Off net DCS Basic DCS Call Coverage DCS with Rerouting Plan Modification Async ATM Trunking ATM WAN Spare Processor ATMS Digital Loss DS1 DS1 MSP Echo Cancellation Attendant Vectoring Figure 25 System Parameters Customer Options Optional Features screen display system parameters customer options OPTIONAI Emergency Access to Attendant Enable dadmin Login Enhanced Conferencing Enhanced EC500 Enterprise Survivable Server Enterprise Wide Licensing ESS Administration Extended Cvg Fwd Admin ternal Device Alarm Admin Extended Cvg Fwd Admin ternal Device Alarm Admin Port Networks Max per MCC Flexible Billing Forced Entry of Account Codes Global Call Classification Hospitality Basic G3V3 Enhancements IP Trunks za x ie x Five Hospitality IP Attendant Consoles NOTE Li i Oa On Oa OL SS OL Oa SO OO Os SS Ss OL
76. both SIP and H 323 telephones support SNMP v2c and have custom Management Information Bases MIBs the MIBs are formatted somewhat differently RSVP amp VMON VMM 9600 Series SIP IP Telephones do not use RSVP Resource ReSerVation Protocol software to provide real time monitoring and historical data of audio quality for VoIP calls 9600 Series SIP IP Telephones do support Avaya Voice over IP VoIP Monitoring Manager VMON now called VMM 9600 Series IP Telephones use both RSVP and VMON QoS Unlike H 323 telephones 9600 Series SIP IP Telephones do not use Avaya Communication Manager to set Quality of Service QoS The SIP IP telephones use the parameters L2QAUD L2QSIG DSCPAUD and DSCPSIG described in Table 11 9600 Series SIP IP Telephones Customizeable System Parameters NAT 9600 Series SIP IP Telephones do not support Network Address Translation NAT 9600 Series IP H 323 Telephones do support NAT Features amp Functions Supported by H 323 and Not Supported by SIP SIP Software Release 2 0 SIP Software Release 1 0 Calltype Digit Conversion Link Layer Discovery Protocol LLDP RSVP GigE Gigabit Ethernet Calltype Digit Conversion IEEE 802 1X RSVP VMON Remote Ping amp Trace Route Web browser SBM24 Button Modules Push Top Line web page and or audio Autodial feature buttons Remote Ping amp Trace Route SBM24 Button Modules Push Top Line web page and or audio Change History Iss
77. ca ra Bae ae ee as 26 DHCP Server Administration 54 DHCP Server Setup 0 54 DHCP Server to Telephone initialization 2i DHCP Server Windows 2000 Setup 63 DHCP Server Windows NT 4 0 Setup a 59 DHCP Configuring for 9600 Series SIP IP Telephones aoaaa a 54 Dialing Enhanced Requirements 110 DIFESERV socs pora aaga moa a me a a a 40 DNS Addressing aooo 98 Document Organization ao oao a a 13 Documentation Related 13 11 Issue 2 December 2007 135 Index E Emergency Number Administration Enhanced Dialing Procedures Enhanced Local Dialing Enhanced Local Dialing Requirements Error Conditions 0 F Feature Support 2 4 Feature Related System Parameters Form Features amp Functions supported by H 323 Not Supported in SIP SW Release 1 0 Features Administering File download Choosing the Right Binary and Upgrade Script BGS a ts oe te hs es Fe ee oe ee eden eae WR ye Download File Content G General Download Process Generic Setup for DHCP Glossary of Terms aooaa a GROUP System Value H Hardware Requirements HTTP HTTPS Server 2 22 00 l IEC ISO Documents IEEE 802 1D and 802 1Q 29
78. ck CPN Unblock Directed Call Pickup EC500 EC500 Extend Call Extended Call Pickup MCT Activation Priority Call Send All Calls Transfer to Voicemail Whisper Page Leave the Ext field blank as the telephone does not support 3rd party call forwarding Leave the Ext field blank as the telephone does not support 3rd party call forwarding This SES feature will show up automatically without administration Regardless of CM Station screen administration these features will show on the Features menu automatically but not on a telephone button This SES feature will show up automatically without administration Regardless of CM Station screen administration these features will show on the Features menu automatically but not on a telephone button This SES feature will show up automatically without administration Regardless of CM Station screen administration these features will show on the Features menu automatically but not on a telephone button Leave the Ext field blank as the telephone does not support 3rd party send all calls This SES feature will show up automatically without administration Regardless of CM Station screen administration these features will show on the Features menu automatically but not on a telephone button Issue 2 December 2007 49 Communication Manager Administration For additional information about administering Avaya Communication Manager for 9600 Series SIP IP Telep
79. ct ED ge 1 of Standard 123456 the permission changes Figure 23 System Parameters Customer Options Optional Features screen display system parameters customer options IP PORT CAPACITI ES OPTIONAL FEATURE Maximum Administered H 323 Trunks Maximum Administered IP Trunks Maximum Concurrently Registered IP Stations Maximum Administered Remote Off Maximum Concurrently Registered Remote Offic iG anl Maximum Number of DS1 Boards with Maximum Number of NOTE Expanded Meet m Conference Ports Echo Cancellation Maximum TN2501 VAL Boards Maximum G250 G350 G700 VAL Sources Maximum TN2602 Boards with 80 VoIP Channels Maximum TN2602 Boards with 320 VoIP Channels runks Stations Maximum Concurrently Registered IP eCons Maximum Video Capable Stations Maximum Video Capable IP Softphones Maximum Administered SIP Trunks Maximum Administered Ad hoc Video Conferencing Ports 100 Or e O BENDO RA OO UrO OGO OG O 132 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 page 2 of x Ko G oN n Or Oo U O DO OOG N O OLG O You must logoff amp login to effect the permission changes X Figure 24 System Parameters Customer Options Optional Features screen display system parameters customer options OPTIONAL F EATUR ES Abbreviated Dialing Access Security Gatew
80. cumentation DHCP and File Servers on page 53 and especially DHCP Server Administration on page 54 Chapter 7 Telephone Software and Binary Files and Chapter 8 Administering Telephone Options Static Addressing Installation in the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide Link Layer Discovery Protocol LLDP on page 101 Link Layer Discovery Protocol LLDP on page 101 DHCP Server Administration on page 54 and Chapter 8 Administering Telephone Options DHCP and File Servers on page 53 and Chapter 8 Administering Telephone Options Static Addressing Installation in the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide DHCP Server Administration on page 54 and Network Time Protocol NTP Server on page 27 Chapter 8 Administering Telephone Options DHCP and File Servers on page 53 and Chapter 7 Telephone Software and Binary Files DHCP and File Servers on page 53 and Chapter 7 Telephone Software and Binary Files Link Layer Discovery Protocol LLDP on page 101 Secondary Ethernet Interface Enable Disable in the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide DHCP and File Servers on page 53 and especially DHCP Server Administration on page 54 Also Chapter 8 Administering Telephone Opti
81. destination MAC address Note When DOT1X 0 or 2 the Proxy Logoff function is not supported 802 1X Supplicant Operation 9600 SIP IP Telephones that support Supplicant operation also support Extensible Authentication Protocol EAP but only with the MD5 Challenge authentication method as specified in IETF RFC 3748 8 5 33a or with TLS If an EAP method in the configuration parameter DOT1XEAPS requires the authentication of a digital certificate the standard authentication requirements apply including matching the TLSSRVRID with that on the certificate If an EAP response requires an identity or a password the values of the DOT1XID and DOT1XPSWD parameters will be used unless a new identity and or password has been entered by the user via an 802 1X User Input interrupt screen in which case the new values entered by the user will be used instead The ID and password are not overwritten by telephone software downloads For all EAP methods if the Supplicant is unauthenticated an 802 1X Waiting interrupt screen is displayed when a response is transmitted unless an 802 1X User Input interrupt screen is already being displayed If an EAP Failure frame is received after transmitting a response that contains an identity or a password an 802 1X User Input interrupt screen is displayed unless an 802 1X User Input interrupt screen is already being displayed If an EAP Failure frame is received after Issue 2 December 2007 99 Administer
82. dex Numerical BO2 IX ee tock woke ee eo ke ae Ae wae 98 802 1X Pass Through and Proxy Logoff 99 802 1X Supplicant Operation 99 9600 Series IP Telephones Administration Alternatives and Options Iz General e ens acre g alee alae Mute Bob a 15 Initialization Process 21 9600 Series SIP IP Telephone Feature Support 42 9600 Series SIP IP Telephones Administering Options for T3 Network Audio Quality Display 30 Scripts and Application Files 68 A About This Guide 02 Administering Applications and Options 111 Administering Avaya Communication Manager 37 Administering Features a aooaa aaa 49 Administering Options and Settings on the Avaya MEMU Iaa a tet oe oe nee th ey eee eet E en 112 Administering Telephone Options 73 Administering the WML Browser z2 Administration Alternatives and Options for 9600 Series SIP IP Telephones iz Administration Overview and Requirements 15 Administration for Avaya Communication Manager 37 Administration for SES o oo 51 Administration for Telephones on server 42 Administrative Checklist 2 19 Administrative Options Local 107 Administrative Process The 19 Administrative Requirements for Communication Managers dya aoe o aii o a i a E ee 37 Alternatives Administration 17 ANSI
83. dst codec direct Total Video Dyn rgn rgn set WAN WAN BW limits Norm Prio Shr Intervening regions CAC IGAR 3 iL 1 y 256 Kbits y n 3 2 1 n NoLimit n n 3 3 1 NoLimit n 3 4 1 n NoLimit n n 3 5 1 n NoLimit n n 3 6 1 y NoLimit y n 3 7 1 y 10 Calls y n 3 8 3 9 3 10 3 11 3 12 3 13 3 14 3 15 The entries on the IP Address network map shown in Figure 8 might redirect endpoints into a particular network region That region could be different from what is administered on the previous forms Figure 15 Numbering Public Unknown Format Form change public unknown numbering 5 Page 1 of X NUMBERING PUBLIC UNKNOWN FORMAT Total Ext Extension Trk CPN CPN Len Code Grp s Prefix Len 12 1234567890123 123456789 123456789012345 12 5 4 777777 10 5 4 250 30379 10 5 4 253 30379 10 5 41 40 303222 11 5 41 45 5 5 41 87 30323 10 5 43 538 7 5 45 222 7 5 47 2222 9 5 EL 45 5 5 406 250 30379 10 5 406 253 30379 10 5 418 303538 11 5 419 222222222222222 15 5 770 970 8 128 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Figure 16 IP Options System Parameters Form display system parameters ip options IP OPTIONS SYSTEM PARAMETERS IP MEDIA PACKET PERFORMANC E THRESHOLDS Roundtrip Propagation Delay ms Packet Loss Ping Test Interval o Number of Pings Per Measurement Interval RTCP MONITOR SERVER Default Server IP Address
84. e One person might function as both the system administrator and the LAN administrator in some environments Issue 2 December 2007 19 Administration Overview and Requirements Table 2 Administrative Checklist Task Description For More Information See Network Requirements Assessment Administer Avaya Communication Manager Administer the Proxy Server DHCP server installation Administer DHCP application Administer Network Time Server HTTP HTTPS server installation Binary file s script file and settings file installation on HTTP HTTPS server Modify settings file as needed Administer telephones locally as applicable Determine that network hardware is in place and can handle telephone system requirements Verify that the call server is licensed and is administered for Voice over IP VoIP Verify the individual telephones are administered as desired Administer for SIP Enablement Services SES Install a DHCP application on at least one new or existing PC on the LAN Add IP telephone administration to DHCP application Set value s for Simple Network Time Protocol SNTP Install an HTTP HTTPS application on at least one new or existing PC on the LAN Download the files from the Avaya support site Edit the settings file as necessary for your environment using your own tools As a Group Individually Chapter 3 Network Requirements Chapter 4 Commu
85. e HTTP proxy server if defined by WMLPROXY WMLPROXY Proxy server address to be used by the WML browser application For detailed information about WML Browser configuration parameters see Table 11 9600 Series SIP IP Telephones Customizeable System Parameters Issue 2 December 2007 113 Administering Applications and Options 114 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Appendix A Glossary of Terms 802 1D 802 1Q 802 1X ARP CELP CLAN CNA DHCP DiffServ DNS EAP H 323 HTTP 802 1Q defines a layer 2 frame structure that supports VLAN identification and a QoS mechanism usually referred to as 802 1D Authentication method for a protocol requiring a network device to authenticate with a back end Authentication Server before gaining network access Applicable 9600 Series IP telephones support IEEE 802 1X for pass through and for Supplicant operation with the EAP MD5 authentication method SIP Software Release 2 0 and up supports 802 1X Address Resolution Protocol used for example to verify that the IP Address provided by the DHCP server is not in use by another IP telephone Code excited linear predictive Voice compression requiring only 16 kbps of bandwidth Control LAN a type of circuit pack Converged Network Analyzer an Avaya product to test and analyze network performance Dynamic Host Configuration Protocol an IETF protocol used to automate IP Address all
86. e of IPADD is not 0 0 0 0 or e the current L2QVLAN value was set by an IEEE 802 1 VLAN Name or e the current L2QVLAN value was set by a TIA LLDP MED Network Policy TLV POE CONS _ Proprietary PoE If the value of POE_CONS_ SUPPORT is 1 SUPPORT Conservation Level PPOE_CONS MODE is set to the level requested in Request TLV the TLV Visiting User Administration A visiting user is anyone who logs into a 9600 Series SIP IP Telephone that is not his or her primary phone at the user s home location This could mean that the visiting user can log into a telephone that is across the country from the home location or one in the office adjacent to the home office When registered as a visiting user e An inactivity timer is used to trigger inactivity and thereby un register a user The Visiting User Inactivity Timer value is communicated to the telephone via Personal Profile Manager PPM The Visiting User Inactivity Timer is a local timer VUTIMER in the telephone that has the same value as the EMU timer value that is set in Avaya Communication Manager CM The inactivity timer is relevant when users are served through a SIP Enablement Services SES that is not their home SES e Registration Events with Q value of 0 result in a logout When a new registration is sent from a visiting roaming or non roaming telephone the visiting telephone takes priority over the user s home or primary telephone Outbound calls can be made from
87. e 9600 Series IP Telephones do not support Regular Expression Matching and therefore do not use wildcards For more information see Administering Options for the 9600 Series SIP IP Telephones on page 73 In configurations where the upgrade script and binary files are in the default directory on the HTTP server do not use the HTTPDIR lt path gt Avaya recommends that you administer DHCP servers to deliver only the options specified in this document Administering additional unexpected options might have unexpected results including causing the IP telephone to ignore the DHCP server The SIP Proxy server name and HTTP server name must each be no more than 32 characters in length Examples of good DNS administration include Option 6 aaa aaa aaa aaa Option 15 dnsexample yourco com ZZZ ZZZ ZZZ ZZZ Option 42 aaa aaa aaa aaa Depending on the DHCP application you choose be aware that the application most likely does not immediately recycle expired DHCP leases An expired lease might remain reserved for the original client a day or more For example Windows NT DHCP reserves expired leases for about one day This reservation period protects a lease for a short time If the client and the DHCP server are in two different time zones the clocks of the computers are not in sync or the client is not on the network when the lease expires there is time to correct the situation The following example shows the implicati
88. e digits beginning with 7 and the access codes have various formats as indicated with the Call Type of fac 1 of 4 Issue 2 December 2007 45 Communication Manager Administration Table 8 CM SIP Configuration Requirements continued Task Form Command Field s Value s After defining the FACs change Used to support both OPS define the FNEs not off pbx telephone and Extension to Cellular provisioned by CM feature name feature buttons using the extensions command Set the appropriate change cos Varied y Yes or n No service permissions to support OPS features on the Class of Service form Enable applicable calling change cor Varied To use the Call Pickup features on the Class of feature the Can Use Directed Restriction form Call Pickup and Can Be Picked Up By Call Pickup fields must be set to y for the affected stations Note that Page 3 can be used to implement a form of centralized call screening for groups of stations and trunks Add a station for each add station Xxxxxx Extension Assign the same extension SIP phone to be supported using the Station form page 1 where XXXXXX represents the extension number Station Type Port Coverage Path COS and COR Name Message Lamp Ext as the CM call server extension administered in SIP Enablement Services See Chapter 5 SIP Enablement Services SES Administration for SES configuration information Use 9620 or 9630 Syste
89. e latter case is the default If the default is invoked after the DHCP lease expires the telephone sends an ARP Request for its own IP Address every five seconds The request continues either forever or until the telephone receives an ARP Reply After receiving an ARP Reply the telephone displays an error message sets its IP Address to 0 0 0 0 and attempts to contact the DHCP server again Option 52 Overload Option if desired If this option is received in a message the telephone interprets the sname and file fields in accordance with IETF RFC 2132 Section 9 3 listed in Appendix B Related Documentation Option 53 DHCP message type Value is 1 DHCPDISCOVER or 3 DHCPREQUEST Option 55 Parameter Request List Acceptable values are 1 subnet mask 3 router IP Address es 6 domain name server IP Address es 7 log server 15 domain name 26 Interface MTU 42 NTP servers SSON site specific option number Issue 2 December 2007 57 Server Administration Option 57 Maximum DHCP message size Option 58 DHCP lease renew time If not received or if this value is greater than that for Option 51 the default value of T1 renewal timer is used as per IETF RFC 2131 Section 4 5 listed in Related Documentation Option 59 DHCP lease rebind time If not received or if this value is greater than that for Option 51 the default value of T2 rebinding timer is used as per RFC 2131 Section 4 5 Th
90. e or customized settings for the telephones A Important This document covers administration for 9600 Series SIP IP Telephones only For administration for 9600 Series IP Telephones using the H 323 protocol see the Avaya one X Deskphone Edition for 9600 Series IP Telephones Administrator Guide Document Number 16 300698 available at www avaya com support This document does not cover administration for Avaya Distributed Office Full documentation for Avaya Distributed Office is available on the Avaya support Web site www avaya com support Avaya does not provide product support for many of the products mentioned in this document Take care to ensure that there is adequate technical support available for servers used with any 9600 Series IP and or SIP IP Telephone system If the servers are not functioning correctly the 9600 Series IP Telephones might not operate correctly Issue 2 December 2007 7 Introduction Major Differences Between 9600 Series SIP IP and 9600 Series H 323 IP Telephones Review this section if your administrative environment includes both SIP and H 323 signaling protocols for 9600 Series IP Telephones General IP Telephony Two major protocols handle Voice over IP VoIP signaling Session Initiation Protocol SIP and H 323 The two protocols provide connection control and call progress signaling but in very different ways These protocols can be used simultaneously over the same network but in gene
91. ed values in the script file For more information see Contents of the Settings File on page 70 Avaya recommends that you administer options on the 9600 Series SIP IP Telephones using script files Some DHCP applications have limits on the amount of user specified information The administration required can exceed those limits for the more full featured telephone models You might choose to completely disable the capability to enter or change option settings from the dialpad You can set the system value PROCPSWD as part of standard DHCP HTTP administration If PROCPSWD is non null and consists of 1 to 7 digits a user cannot invoke any local options without first entering the PROCPSWD value on the Craft Access Code Entry screen For more information on craft options see the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide Issue 2 December 2007 73 Administering Telephone Options A Important PROCPSWD is likely stored on the server in the clear and is sent to the telephone in the clear Therefore do not consider PROCPSWD as a high security technique to inhibit a sophisticated user from obtaining access to local procedures Administering PROCPSWD limits access to all local procedures including VIEW VIEW is a read only Craft option that allows review of the current telephone settings Table 11 9600 Series SIP IP Telephones Customizeable System Parameters Para
92. ed CNA registration TCP Source Port field of messages registration messages sent by the telephone s CNA Agent PORTAUD or the port number Any Received RTP and SRTP UDP reserved for CNA RTP tests packets PORTAUD 1 if PORTAUD Any Received RTCP and UDP is even or PORTAUD 1 if SRTCP packets PORTAUD is odd or the port number reserved for CNA RTP tests plus or minus one as for PORTAUD above If signaling is initiated by the Any Received signaling protocol UDP TCP telephone the number used packets in the Source Port field of the signaling packets sent by the telephone If signaling is initiated by the server System Specific 2 of 2 Table 6 Transmitted Packets Source SIP IP Telephone Destination Port Source Port Use UDP or TCP 53 Any unused Transmitted DNS messages UDP port number 67 68 Transmitted DHCP UDP messages 80 unless explicitly specified Any unused Packets transmitted by the TCP otherwise i e in a URL port number telephone s HTTP client 123 Any unused Transmitted SNTP UDP port number messages The number used in the 161 Transmitted SNMP UDP Source Port field of the SNMP messages query packet received by the telephone 443 unless explicitly specified Any unused TLS SSL packets TCP otherwise i e in a URL port number transmitted by the telephone s HTTP client 1 of 3 32 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Other Network Considerations Table 6 Transmitted Packets Source
93. ed before forwarding For example tagging is not added or removed and the VLAN ID and tagged frames priority are not changed The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary Ethernet interface or to the telephone without regard to specific VLAN IDs or the existence of tags All tagged frames received on the secondary Ethernet interface are changed before forwarding to make the VLAN ID equal to the PHY2VLAN value and the priority value equal to the PHY2PRIO value Untagged frames received on the secondary Ethernet interface are not changed before forwarding Tagged frames with a VLAN ID of zero priority tagged frames will either be forwarded without being changed preferred or changed before they are forwarded such that the VLAN ID of the forwarded frame is equal to the PHY2VLAN value and the priority value is equal to the PHY2PRIO value The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary Ethernet interface or to the telephone without regard to specific VLAN IDs or the existence of tags Frames received on the secondary Ethernet interface will not be changed before forwarding In other words tagging is not added or removed and the VLAN ID and priority of tagged frames is not changed Tagged frames received on the Ethernet line interface will only be forwarded to t
94. ed to synchronize computer clocks in the internet Secure Real time Transport Control Protocol Secure Real time Transport Protocol Transmission Control Protocol Internet Protocol a network layer protocol used on LANs and internets Trivial File Transfer Protocol used to provide downloading of upgrade scripts and application files to certain IP telephones SIP IP telephones use HTTP or HTTPS instead of TFTP Transport Layer Security an enhancement of Secure Sockets Layer SSL TLS is compatible with SSL 3 0 and allows for privacy and data integrity between two communicating applications Type Length Value elements transmitted and received as part of Link Layer Discovery Protocol LLDP User Datagram Protocol a connectionless transport layer protocol Registration with Avaya Communication Manager by an IP telephone with no extension Allows limited outgoing calling Uniform Resource Identifier and Uniform Resource Locator Names for the strings used to reference resources on the Internet for example HTTP URI is the newer term Virtual LAN Voice over IP a class of technology for sending audio data and signaling over LANs Wireless Markup Language used by the 9600 Series IP Telephone Web Browser to communicate with WML servers 3 of 3 Issue 2 December 2007 117 Glossary of Terms 118 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Appendix B Related Documentation IETF Docu
95. efault Value Description and Value Range DSTSTOP DTMF_PAYLOAD_TYPE ENABLE_AVAYA__ ENVIRONMENT ENABLE_CALL_LOG ENABLE_CONTACTS ENABLE_EARLY_MEDIA ENABLE_G711A TSunNov2L 120 Used to identify stop date for automatic change to Daylight Saving Time Default string length with a format of either odddmmmht or Dmmmhit where o one character representing an ordinal adjective of 1 first 2 Second 3 third 4 fourth or L last ddd 3 characters containing the English abbreviation for the day of the week mmm 3 characters containing the English abbreviation for the month h one numeric digit representing the time to make the adjustment exactly on the hour at hAM 0h00 in military format where valid values of h are 0 through 9 t one character representing the time zone relative to the adjustment where L is local time and U is universal time D one or two ASCII digits representing the date of the month from 1 or 01 to 31 or the character L which means the last day of the month RTP dynamic payload used for RFC 2833 signaling Range is 96 to 127 Determines whether the phone operates in a mode to comply with 3rd party standard SIP proxy provision of SIPPING 19 feature or the Avaya environment mode provision of SIP AST features and use of PPM for download and backup restore Valid values are 0 Non Avaya environment 1 Avaya environment Enable or disable complete Call L
96. either the visiting telephone or the primary telephone The home SES lowers the q value of previous registrations to zero and promotes the new registration to ensure that inbound calls will be routed to the most recent telephone registered Issue 2 December 2007 105 Administering Telephone Options e The telephone will un register if it is a visiting user telephone But that telephone will become registered inactive if it is the primary telephone Set the VU_MODE configuration parameter value in the settings file to determine the visiting user login routine VU_MODE determines whether the phone will support Visiting User capabilities as follows e f the VU_MODE value is zero Off the telephone is considered a non VU phone This is the default value and the value associated with the user s home phone The inactivity timer is not applied when VU_MODE is 0 e f the VU_MODE value is 1 Optional the telephone presents the user with the Login Screen with a Primary Phone yes no toggle field for the user to designate whether the telephone is that user s primary phone If the user selects yes then the phone operates as a non visiting user telephone and the inactivity timer is not applied If the user selects no then the telephone operates in the visiting user mode where an inactivity timer will log the user off after a predetermined time e If the value is 2 Forced the telephone is always in the visiting user mode and the inactivity t
97. election Issue 2 December 2007 39 Communication Manager Administration RSVP and RTCP SRTCP Avaya SIP IP Telephones support the RTP SRTP Control Protocol RTCP SRTCP The 9600 Series SIP IP Telephones do not support RSVP Resource ReSerVation Protocol QoS The 9600 Series SIP IP Telephones support both IEEE 802 1D Q and DiffServ Other network based QoS initiatives such as UDP port selection do not require support by the telephones However the initiatives contribute to improved QoS for the entire network IEEE 802 1D and 802 1Q The 9600 Series IP Telephones can simultaneously support receipt of packets using or not using 802 1Q parameters To support IEEE 802 1D Q you can administer 9600 Series SIP IP Telephones by the value of the following configuration parameters e L2Q e L2QVLAN e L2QAUD and e L2QSIG NAT 9600 Series SIP IP Telephones do not support Network Address Translation NAT interworking DIFFSERV Type of Service bits 0 5 also called the Differentiated Services Code Point are set to the binary equivalent of the decimal number represented by the value of the following configuration parameters e DSCPAUD for transmitted audio RTP RTCP SRTP and SRTCP packets e DSCPSIG for transmitted system specific signaling packets e Zero for all other transmitted packets e g DHCP DNS HTTP SNMP etc Received DSCP information will be ignored 40 9600 Series SIP IP Telephones Admin
98. elephone number as a link on the Human Resources page A 9620 user selects that link The 9620 deduces the telephone number is part of the corporate network because the length of the telephone number is the same as the corporate dialing plan The telephone dials the number without further processing Example A user notes a Web site contains an international telephone number that needs to be called and initiates the call The telephone determines the number to be called is from another country code The telephone then prepends the rest of the telephone number with PHNOL to get an outside line and PHNIC to get an international trunk The telephone then dials normally with the CM call server routing the call appropriately Issue 2 December 2007 109 Administering Telephone Options Enhanced Local Dialing Requirements The enhanced local dialing feature is invoked when all the following conditions are met e An application on a 9600 Series IP Telephone obtains or otherwise identifies a character string as containing a telephone number the user wants to dial and e The Phone application determines a call appearance is available for an outgoing call and e The originating application passes the character string to the Phone application and e The originating application specifies a Source Flag set to No and e The current value of ENHDIALSTAT is 1 partially enabled or 2 fully enabled The Phone application takes the incoming ch
99. ent to in 3rd party non Avaya environments only Defines if a custom skin is currently selected non empty string or built in default skin is used empty string or not set If a custom skin is selected non empty string this value points to the corresponding skin resource definition i e contains a label as defined in SKINS configuration parameter Can also be set by the end user via Avaya Menu Screen amp Sounds option Formatting string defining how to display the date in the top line and the call log Controls daylight saving setting Values are O daylight saving time is deactivated no offset to local time 1 daylight saving time is activated offset to local time as configured in DSTOFFSET 2 the device switches automatically to daylight saving time and back according to the contents of DSTSTART and DSTSTOP DHCP Standard lease violation flag Indicates whether to keep the IP Address if there is no response to lease renewal If set to 1 No the telephone strictly follows the DHCP standard with respect to giving up IP Addresses when the DHCP lease expires If set to 0 Yes the telephone continues using the IP Address until it detects reset or a conflict see DHCP Generic Setup Dial plan in non PPM format Used to identify the end of dialing information to accelerate dialing Valid value is 0 to 1023 characters that define the dial plan Text string containing the IP Address of zero or more DNS serve
100. file and binary file s from the Avaya Web site http www avaya com support to the HTTP server For more information see Contents of the Settings File on page 70 Note Many LINUX servers distinguish between upper and lower case names Ensure that you specify the settings file name accurately as well as the names and values of the data within the file If you choose to enhance the security of your HTTP environment by using Transport Layer Security TLS you also need to e Install the TLS server application e Administer the system parameter TLSSRVR to the address es of the Avaya HTTP server 66 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 7 Telephone Software and Binary Files General Download Process The 9600 Series SIP IP Telephones download script files binary files and settings files from either an HTTP or HTTPS server The HTTPS server applies only if the server supports Transport Layer Security TLS encryption Note The script files binary files and settings files discussed in this chapter are identical for HTTP and HTTPS servers The generic term file server refers to both HTTP server and HTTPS server The file downloading process is the same for both servers except that when you use an HTTPS server a TLS server is contacted first The telephone queries the file server which transmits a script file to the telephone The script file tells the telephone which binary
101. g spaces Limits the hands free audio operation mode Valid values are O no speakerphone allowed 1 one way speakerphone operation allowed monitor 2 two way speakerphone operation allowed Controls the use of SIP and SIPS subscriptions Valid values are 0 2 Flag indicating whether the telephone supports GigE Gigabit Ethernet Valid values are 0 Telephone does not support GigE 1 Telephone supports GigE System Default Language definition String representing a file name shall be identical to one of the file names received via LANGUAGES parameter or null Time interval number of seconds after which TCP keep alive packets are re transmitted The interval is started by the system TCP IP stack when TCP keep alive is enabled with specified time intervals Values are 5 60 seconds Indicates whether TCP IP keep alive should be enabled at the system Values are 0 TCP keep alive disabled 1 TCP keep alive enabled 18 of 21 Issue 2 December 2007 91 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range TCP_KEEP_ALIVE_TIME 60 TIMEFORMAT 0 TLSDIR Null TLSPORT 443 TLSSRVRID 1 TRUSTCERTS Null USE_EXCHANGE_ 0 CALENDAR USE_QUAD_ZEROS_ 0 FOR_HOLD VLANSEP 1 VLANTEST 60 This time interval is the time 9600 Series SIP IP Telephones will wait before sending out a TC
102. gh the settings file or on a per phone basis using a local Craft procedure Support for Non Avaya Third Party Environments Several parameters most notably ENABLE_AVAYA_ENVIRONMENT have been added to cover operation for either e an Avaya environment which provisions SIP AST features and uses Personal Profile Manager PPM for download and backup restore or e anon Avaya mode which complies with 3rd party standard SIP proxy with provision for SIPPING 19 feature 10 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 What s New in SIP Software Release 2 0 WML Applications Browser 9600 Series SIP IP Telephones now provide access to WML applications via a WML Browser as described in Chapter 9 Administering Applications and Options New Revised and Deleted Configuration Parameters The following configuration parameters have been added for this release and are linked to the table that describes them in detail e CALL_TRANSFER_MODE e CALLFWDADDR e CALLFWDDELAY e CALLFWDSTAT e CNAPORT e CNASRVR e CONFIG_SERVER_SECURE_MODE e COVERAGEADDR e DIALPLAN e DOTIX e DOTIXEAPS e DOT1XSTAT e ENABLE_AVAYA_ ENVIRONMENT e INTER_DIGIT_TIMEOUT replaces INTER_DIGIT_DIALING_TIMEOUT_DURATION e LAST_LOGIN_STATUS system set only e LLDP_ ENABLED e MWISRVR e NO DIGITS TIMEOUT e PHNEMERGNUM e PHNNUMOFSA e POE CONS SUPPORT e PRESENCE SERVER e PROVIDE_EDITED_ DIALING e PROVIDE_EXCHANGE_CALENDAR e
103. guage file or review pertinent information go to htip support avaya com unicode Note Specifying a language other than English in the configuration file has no impact on Avaya Communication Manager settings values or text strings Enhanced Local Dialing The 9600 Series SIP IP Telephones have a variety of telephony related applications that might obtain a telephone number during operation For example the Call Log saves a number of an incoming caller but does not consider that the user has to then prepend the saved number with a digit to dial an outside line and possibly a digit to dial long distance 9600 Series SIP IP Telephones can evaluate a raw telephone number based on administered parameters The telephone can automatically prepend the correct digits saving the user time and effort This is the Enhanced Local Dialing feature The key to the success of this feature is accurate administration of several important values summarized below 108 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Enhanced Local Dialing The system values relevant to the Enhanced Dialing Feature are ENHDIALSTAT Enhanced dialing status If set to 1 the enhanced local dialing feature is partially enabled meaning dialing rules do not apply to dialing from the Contacts list If set to 2 the enhanced local dialing feature is fully enabled and does apply to dialing from the Contacts list If set to O enhanced local di
104. hat you created from the Unused Options list Click the Add button Select option 003 from the Unused Options list Click the Add button Click the OK button Select the Global parameter under DHCP Options Select the 242 option that you created from the Unused Options list Click the Add button Click the OK button Issue 2 December 2007 61 Server Administration Activating the Leases Use the following procedure to activate the leases e Click Activate under the Scope menu The light bulb icon for the scope lights Verifying Your Configuration This section describes how to verify that the 96XXOPTIONSs are correctly configured for the Windows NT 4 0 DHCP server Verify the Default Option 242 96XXOPTION 1 Select Start gt Programs gt Admin Tools gt DHCP Manager 2 Expand Local Machine in the DHCP servers window by double clicking until the sign changes to a sign In the DHCP servers frame click the scope for the IP telephone Select Defaults from the DHCP_Options menu In the Option Name pull down list select 242 96XXOPTION Verify that the Value String box contains the correct string from DHCP Server Administration O oa A O If not update the string and click the OK button twice Verify the Scope Option 242 96XXOPTION 1 Select Scope under DHCP OPTIONS 2 In the Active Options scroll list click 242 96XXOPTION 3 Click the Value button 4 Verify that the Value Str
105. he secondary Ethernet interface if the VLAN ID equals PHY2VLAN Tagged frames received on the Ethernet line interface will only be forwarded to the telephone if the VLAN ID equals the VLAN ID used by the telephone Untagged frames will continue to be forwarded or not forwarded as determined by the Ethernet switch forwarding logic Tagged frames with a VLAN ID of zero priority tagged frames will either be forwarded to the secondary Ethernet interface or the telephone as determined by the forwarding logic of the Ethernet switch preferred or dropped Issue 2 December 2007 97 Administering Telephone Options DNS Addressing The 9600 Series SIP IP Telephones support DNS addresses and dotted decimal addresses The telephone attempts to resolve a non ASCll encoded dotted decimal IP Address by checking the contents of DHCP Option 6 See DHCP Generic Setup on page 56 for information At least one address in Option 6 must be a valid non zero dotted decimal address otherwise DNS fails The text string for the DOMAIN system parameter Option 15 Table 11 is appended to the address es in Option 6 before the telephone attempts DNS address resolution If Option 6 contains a list of DNS addresses those addresses are queried in the order given if no response is received from previous addresses on the list As an alternative to administering DNS by DHCP you can specify the DNS server and or Domain name in the HTTP script file But fir
106. he settings Abort Transfer Transfer Upon Hang up and Toggle Swap on page 7 of the system parameters features screen Issue 2 December 2007 41 Communication Manager Administration Conferencing Call Considerations Unlike 9600 H 323 IP Telephones the 9600 Series SIP IP Telephones conference operation is controlled locally by the phone and is not affected by the settings Abort Conference Upon Hang up No Dial Tone Conferencing Select Line Conferencing and Toggle Swap on page 7 of the system parameters features screen Telephone Administration Table 7 summarizes the calling features available on 9600 Series SIP IP Telephones Some features are supported locally at the telephone while others are only available with Avaya SIP Enablement Services and Communication Manager with OPS The features shown in Table 7 can be invoked at the phone either directly or by selecting a CM provisioned feature button Communication Manager automatically handles many other standard calling features via OPS such as call coverage trunk selection using Automatic Alternate Routing AAR or Automatic Route Selection ARS Class Of Service Class Of Restriction COS COR and voice messaging Details on feature operation and administration can be found in the Avaya Extension to Cellular and OPS Installation and Administration Guide Document Number 210 100 500 The Avaya SIP solution configures all SIP telephones in Communication Manager as OPS Table 7 96
107. he user to add delete or in some cases edit entries As the administrator you might not want the user to have that level of functionality This chapter also contains information related to administering the Avaya A Menu to include the WML browser and other browser setup information In 4600 and 9600 Series H 323 IP Telephones the parameters APPSTAT meaning Application permission status and OPSTAT meaning Options permission status control application access and functionality However 9600 Series SIP IP Telephones have a more granular way of assigning functionality with a specific parameter for each permission as follows e ENABLE CALL _LOG Allows end user access to the list of unanswered and answered calls If disabled the Call Log application is not displayed to the user and calls are not logged e ENABLE REDIAL Allows the end user to redial one to three previously called numbers If disabled redialing is not available to the end user e ENABLE_REDIAL LIST Allows the end user to select a number to redial from a list If disabled only the previously dialed number can be redialed e ENABLE CONTACTS Allows end user access to a list of numbers and to make calls by selecting a Contact Name Number If disabled the Contacts application is not displayed to the user and a Contact list cannot be set up or maintained e ENABLE _MODIFY_CONTACTS If the Contacts application is enabled ENABLE_CONTACTS 1 this option allows or preven
108. hernet interface e will not forward frames received with the 802 1AB LLDP group multicast address as the destination MAC address between the Ethernet line interface and the secondary Ethernet interface A 9600 Series IP Telephone initiates LLDP after receiving an LLDPDU message from an appropriate system Once initiated the telephones send an LLDPDU every 30 seconds with the following contents Table 13 LLDPDU Transmitted by 9600 Series SIP IP Telephones Category TLV Name Type TLV Info String Value Basic Mandatory Chassis ID IPADD of telephone IANA Address Family Numbers enumeration value for IPv4 or subtype 5 Network address Basic Mandatory Port ID MAC address of the telephone Basic Mandatory Time To Live 120 seconds 1 of 3 Issue 2 December 2007 101 Administering Telephone Options Table 13 LLDPDU Transmitted by 9600 Series SIP IP Telephones continued Category TLV Name Type TLV Info String Value Basic Optional System Name The Host Name sent to the DHCP server in DHCP option 12 Basic Optional System Capabilities Bit 2 Bridge will be set in the System Capabilities if the telephone has an internal Ethernet switch If Bit 2 is set in Enabled Capabilities then the secondary port is enabled Bit 5 Telephone will be set in the System Capabilities If Bit 5 is set in the Enabled Capabilities than the telephone is registered Basic Optional Management Address Mgmt IPv4 IP Address of telephone
109. hones see the following Avaya documents available on the Avaya Support Web site e Administrator Guide for Avaya Communication Manager Document 03 300509 e Feature Description and Implementation for Avaya Communication Manager Document 555 245 770 50 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 5 SIP Enablement Services SES Administration Introduction SIP Enablement Services SES software resides on the SIP Proxy server and provides most of the features and functionality to SIP telephones This chapter describes using the SES Web browser to configure SES for use with 9600 Series SIP IP Telephones Avaya provides a Web browser to simplify SES administration Using the Web Browser to Configure SES Follow this configuration procedure 1 Set the browser URL to http IP address admin where IP address is the IP Address of the Avaya SIP Enablement Services Edge or Edge Home Server Log in as the administrator admin and when prompted enter the password The main administration screen displays after login Note This example administers station 34071 as a SIP endpoint using a 9630 telephone Click on Launch Administration Web Interface The SIP Enablement Services Web interface screen displays Click Add under the Users heading on the left side menu The Add User screen displays 5 Complete all required fields indicated by asterisks 6 Enter a handle in
110. ies SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration Adding DHCP Options Use the following procedure to add DHCP options to the scope you created in the previous procedure 1 On the DHCP window right click the Scope Options folder under the scope you created in the last procedure A drop down menu displays 2 In the left pane of the DHCP window right click the DHCP Server name then click Set Predefined Options 3 Under Predefined Options and Values click Add 4 Inthe Option Type Name field enter any appropriate name for example Avaya IP Telephones 5 Change the Data Type to String 6 In the Code field enter 242 then click the OK button twice The Predefined Options and Values dialog box closes leaving the DHCP dialog box enabled 7 Expand the newly created scope to reveal its Scope Options 8 Click Scope Options and select Action gt Configure Options from the menu 9 In the General tab page under the Available Options check the Option 242 checkbox 10 In the Data Entry box enter the DHCP IP telephone option string as described in DHCP Generic Setup on page 56 Note You can enter the text string directly on the right side of the Data Entry box under the ASCII label 11 From the list in Available Options check option 003 Router 12 Enter the gateway router IP Address from the IP Address field of Table 3 Required Network Information Before Installati
111. ignored or e delete unused or unsupported 9600 IP Series Telephone parameters to shorten the DHCP message length Only the following parameters can be set in the DHCP site specific option for 96xx telephones although most of them can be set in a 46xxsettings txt file as well 54 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration Table 9 Parameters Set by DHCP Parameter Description HTTPDIR Specifies the path name to prepend to all file names used in HTTP GET operations during startup 0 to 127 ASCII characters no spaces The command is SET HTTPDIR myhttpdir The path relative to the root of the HTTP file server where 96xx telephone files are stored If an Avaya file server is used to download configuration files over TLS but a different server is used to download software files via HTTP set the path of the Avaya server in the DHCP site specific option and set HTTPDIR again in the 46xxsettings txt file with the appropriate path for the second server HTTPDIR is the path for all HTTP operations except for BRURI HTTPPORT Destination port for HTTP requests 0 65535 default is 80 HTTPSRVR IP Address es or DNS name s of HTTP file server s used for file download settings file language files code during startup The files are digitally signed so TLS is not required for security ICMPDU Controls the extent to which ICMP Destination Unreachable messages are sent in response to
112. imer is always applied Emergency Number Administration Set the PHNEMERGNUM configuration parameter in the settings file to assign an emergency telephone number This telephone number will be automatically dialed whenever the Emerg softkey is selected on the Login screen or the Phone screen or when the user chooses the Yes softkey on an Emergency pop up screen The local proxy routes emergency calls from a user at a visited phone so that the local emergency number is called When PHNEMERGNUM is administered using the Emerg softkey overrides the SPEAKERSTAT parameter setting or a user selected referred audio path This means that the even if the Speakerphone is disabled it is the default transducer when the user presses the Emerg softkey 106 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Local Administrative Craft Options Using the Telephone Dialpad Local Administrative Craft Options Using the Telephone Dialpad The Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide details how to use Craft local procedures at the telephone for administration The local procedures you might use most often as an administrator are ADDR Static address programming CLEAR Remove all administered values user specified data option settings etc and return a telephone to its initial out of the box default values DEBUG Enable or disable debug mode for the button module se
113. ing Telephone Options transmitting a response that did not contain an identity or a password an 802 1X Failure interrupt screen is displayed When a telephone is installed for the first time and 802 1x is in effect the dynamic address process prompts the installer to enter the Supplicant identity and password The IP telephone does not accept null value passwords See Dynamic Addressing Process in the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide The telephone stores 802 1X credentials when successful authentication is achieved Post installation authentication attempts occur using the stored 802 1 X credentials without prompting the user for ID and password entry An IP telephone can support several different 802 1X authentication scenarios depending on the capabilities of the Ethernet data switch to which it is connected Some switches may authenticate only a single device per switch port This is known as single supplicant or port based operation These switches typically send multicast 802 1X packets to authenticating devices These switches support the following three scenarios e Standalone telephone Telephone Only Authenticates When the telephone is configured for Supplicant Mode DOT1X 2 the telephone can support authentication from the switch e Telephone with attached PC Telephone Only Authenticates When the telephone is configured for Supplicant Mode DOT 1
114. ing box contains the correct string from DHCP Generic Setup on page 56 If not update the string and click the OK button Verify the Global Option 242 96XXOPTION 1 Select Global under DHCP OPTIONS 2 In the Active Options scroll list click 242 96XXOPTION 3 Click the Value button 4 Verify that the Value String box contains the correct value from DHCP Generic Setup on page 56 If not update the string and click the OK button 62 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration Windows 2000 DHCP Server Verifying the Installation of the DHCP Server Use the following procedure to verify whether the DHCP server is installed 1 2 3 Select Start gt Program gt Administrative Tools gt Computer Management Under Services and Applications in the Computer Management tree find DHCP If DHCP is not installed install the DHCP server Otherwise proceed directly to Creating and Configuring a DHCP Scope for instructions on server configuration Creating and Configuring a DHCP Scope Use the following procedure to create and configure a DHCP scope 1 2 Select Start gt Programs gt Administrative Tools gt DHCP In the console tree click the DHCP serverto which you want to add the DHCP scope for the IP telephones This is usually the name of your DHCP server machine Select Action gt New Scope from the menu Windows displays the New Scope Wizard t
115. ion parameter value is 1 Network Information is listed if and only if the PROVIDE_NETWORKINFO_SCREEN configuration parameter value is 1 Logout is listed if and only if the PROVIDE_LOGOUT configuration parameter value is 1 Administering the WML Browser SIP software Release 2 0 provides a WML Browser which if administered follows the Options and Settings listing on the Avaya A Menu Note WML applications are accessed from the Browser Set the configuration parameter WMLHOME in the settings file to link the Browser Home page to the Avaya A Menu and to include the Browser option on the Avaya A Menu The Browser application is listed if and only if it is properly administered as specified in Avaya one X 112 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Avaya A Menu Administration Deskphone Edition for 9600 IP Telephones Application Programmer Interface API Guide Document Number 16 600888 In addition to WMLHOME other browser related configuration parameters which can be set using the 46xxsettings txt file as applicable to your environment are WMLEXCEPT Exception domain for the WML browser proxy server WMLIDLETIME Number of minutes of inactivity until the Web browser will display the idle URL specified in WMLIDLEURI WMLIDLEURI URL of web page to be displayed after idle timer WMLIDLETIME expires WMLPORT TCP port number the WML browser application should use to access th
116. ist Access to MCT y Fully Restricted Service n Group II Category For MFC 7 Hear VDN of Origin Annc n Send ANI for MFE n_ Add Remove Agent Skills y MF ANI Prefix lt Automatic Charge Display n Hear System Music on Hold y PASTE Display PBX Data on telephone n Can Be Picked Up By Directed Call Pickup n Can Use Directed Call Pickup n Group Controlled Restriction inactive Figure 19 Class of Restriction screen page 2 change cor nn Page 2 of x CLASS OF RESTRICTION MF Incoming Call Trace n Brazil Collect Call Blocking n Block Transfer Display n y n Block Enhanced Conference Transfer Displays Remote Logout of Agent Station Lock COR 10 Outgoing Trunk Disconnect Timer minutes Line Load Control Maximum Precedence Level Preemptable MLPP Service Domain Station Button Display of UUI IE Data Service Observing by Recording Device ERASE 24xx USER DATA UPON Dissociate or unmerge at this phone none EMU login or logoff at this phone none Mask CPN NAME for Internal Calls 130 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Figure 20 Class of Restriction screen page 3 hange cor nn CLASS OF RI SAC CF Override by SAC CF Override Protection for ESTRICTION Team Btn n Team Btn n Page 3 of x NOTE Use pages 4 to 13 to assign up to 995 CORs Figure 21 Class of Restriction screen page 4 change cor nn CLASS OF RES
117. istrator Guide SIP Release 2 0 Voice Mail Integration Voice Mail Integration 9600 Series SIP IP Telephones use the settings file to configure the Messages button by setting the system parameter MSGNUM to any dialable string MSGNUM examples are e astandard telephone number the telephone should dial to access your voice mail system such as AUDIX or Octel e a Feature Access Code FAC that allows users to transfer an active call directly to voice mail FACs are supported only for QSIG integrated voice mail systems like AUDIX or Octel QSIG is an enhanced signaling system that allows the voice mail system and Avaya Communication Manager Automated Call Processing ACP to exchange information When the user presses the Messages button on the telephone that number or FAC is automatically dialed giving the user one touch access to voice mail The settings file specifies the telephone number to be dialed automatically when the user presses this button The command is SET MSGNUM 1234 where 1234 is the Voice Mail extension CM hunt group or VDN For more information see Table 11 Auto Hold 9600 Series SIP IP Telephones always provide auto hold regardless of whether or not the Auto Hold parameter is administered on the IP Network System Parameters form Call Transfer Considerations Unlike 9600 H 323 IP Telephones the 9600 Series SIP IP Telephones transfer operation is controlled locally by the telephone and is not affected by t
118. ith only three 3 call appearances set the field to n for proper SIP conference and transfer operation In this mode all call appearances are available for making or receiving calls Enter the name of the voice messaging system administered for this system This field with a default of s for system governs whether an unanswered forwarded call is given CM coverage treatment If CM is configured to always send Caller ID you can individually block certain stations by setting this field to n This field also needs to be set to n if you want to use the Calling Number nblock FNE Fill in the number of call appearances call appr buttons to be supported for this telephone Use the following guidelines to determine the correct number To support certain transfer and conference scenarios the minimum number of call appr buttons should be 3 3 of 4 Issue 2 December 2007 47 Communication Manager Administration Table 8 CM SIP Configuration Requirements continued Task Form Command Field s Value s Stations With Off PBX change Station Use to map the Telephone Integration off pbx telephone Extension Communication Manager form page 1 station mapping extension to the same SIP XXXXXxX where Application Enablement Services call XXXXXxX represents server extension The the extension number Dial Prefix Application is OPS Enter Stations With Off PBX Telephone Integration form page 2 of the
119. l list can contain up to 255 total ASCII characters You must separate IP Addresses with commas with no intervening spaces At least one address in Option 6 must be a valid non zero dotted decimal address Option 12 Host Name Value is AVohhhhhh where o has one of the following values based on the OID first three octets of the telephone s MAC address A if the OID is 00 04 0D B if the OID is 00 1B 4F SIP software Release 2 0 E if the OID is 00 09 6E L if the OID is 00 60 1D T if the OID is 00 07 3B SIP software Release R2 0 and X if the OID is anything else and where hhhhhh are ASCII characters for the hexadecimal representation of the last three octets of the telephone s MAC address Option 15 DNS Domain Name This string contains the domain name to be used when DNS names in system parameters are resolved into IP Addresses This domain name is appended to the DNS name before the 9600 IP Telephone attempts to resolve the DNS address Option 15 is necessary if you want to use a DNS name for the HTTP server Otherwise you can specify a DOMAIN as part of customizing HTTP as indicated in DNS Addressing on page 98 56 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration Option 42 SNTP Server This option specifies a list of IP Addresses indicating NTP servers available to the telephone List servers in the order of preference The minimu
120. lephone to display All downloadable language files contain all the information needed for the telephone to present the language as part of the user interface Use the configuration file 46xxsettings txt and these parameters to customize the settings for up to four languages e LANGUAGES the list of languages to be downloaded from which the end user can select a desired display language Each language is listed in the following format Mls_Spark_German xml Mls_Spark_English xml Mls_Spark_CastilianSpanish xml and so on e SYSTEM_LANGUAGE a string indicating the filename of the default system language The string indicates which of the available languages to use for display purposes If this parameter is not set or if no other language has been set by the user or if a user language choice cannot be satisfied the built in English strings are used e LANGOSTAT Allows the user to select the built in English language when other languages are downloaded If LANGOSTAT is 0 and at least one language is downloaded the user cannot select the built in English language If LANGOSTAT is 1 the default the user can select the built in English language text strings For more information see 9600 Series SIP IP Telephones Customizeable System Parameters To view multiple language strings see the MLS local procedure in the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide To download a lan
121. lity for a remote reboot of the 9600 Series SIP IP Telephones As a result the telephone automatically starts reboot procedures If new software is available on the server the telephone downloads it as part of the reboot process The Avaya one X Deskphone Edition for 9600 IP Telephones Installation and Maintenance Guide covers upgrades to a previously installed telephone and related information Issue 2 December 2007 67 Telephone Software and Binary Files 9600 Series SIP IP Telephone Scripts and Binary Files Choosing the Right Binary File and Upgrade Script File The software releases containing the files needed to operate the 9600 Series IP Telephones are bundled together You download this self extracting executable file to your file server from the Avaya support Web site at http www avaya com support The file is available in both zipped and unzipped format You must select one of two software bundles to download the latest software depending on whether your telephone environment is primarily SIP centric or H 323 centric Each bundle contains e An upgrade script file 96xxupgrade txt which allows you to upgrade to new software releases and new functionality without having to replace SIP IP telephones The upgrade script tells the telephone whether a software upgrade is needed All Avaya IP Telephones attempt to read this file whenever they reset The upgrade script file is also used to point to the settings file An
122. m length is 4 and the length must be a multiple of 4 Option 51 DHCP lease time If this option is not received the DHCPOFFER is not be accepted Avaya recommends a lease time of six weeks or greater If this option has a value of FFFFFFFF hex the IP Address lease is assumed to be infinite as per RFC 2131 Section 3 3 so that renewal and rebinding procedures are not necessary even if Options 58 and 59 are received Expired leases cause Avaya IP Telephones to reboot Avaya recommends providing enough leases so an IP Address for an IP telephone does not change if it is briefly taken offline Note Regarding Option 51 The DHCP standard states that when a DHCP lease expires the device should immediately cease using its assigned IP Address If the network has problems and the only DHCP server is centralized the server is not accessible to the given telephone In this case the telephone is not usable until the server can be reached Avaya recommends that once assigned an IP Address the telephone continues using that address after the DHCP lease expires until a conflict with another device is detected As Table 11 9600 Series SIP IP Telephones Customizeable System Parameters indicates the system parameter DHCPSTD allows an administrator to specify that the telephone will either a Comply with the DHCP standard by setting DHCPSTD to 1 or b Continue to use its IP Address after the DHCP lease expires by setting DHCPSTD to 0 Th
123. m populated For voice messaging or other hunt group if available Same values as administered in the previous COS amp COR section s The person associated with the telephone This name should match what is entered for name in the Avaya SES proxy configuration Enter the extension of the station you want to track with the message waiting lamp Usually the same extension initially entered on the Station form 2014 46 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 CM SIP IP Telephone Configuration Requirements Table 8 CM SIP Configuration Requirements continued Task Form Command Field s Value s Continue adding station add station xxxxxx Bridged Call Set to y if the extension for information for the SIP where XXXxxx Alerting this SIP telephone will have a phone using the Station form page 2 Continue adding station button assignments for the SIP telephone using the Station form page 4 represents the extension number Restrict Last Appearance AUDIX Name Coverage After Forwarding Per Station CPN Send Calling Number BUTTON ASSIGNMENTS 1 call appr 2 call appr etc bridged appearance defined on another non SIP telephone Note that no other attributes of the bridged appearance feature apply to SIP telephones e g off hook indication bridge on etc By default the last call appearance is reserved for outgoing calls from a phone On stations w
124. mas with no intervening spaces Depending on the specific DHCP application only 127 characters might be supported When specifying IP Addresses for the file server use either dotted decimal format XXX XXX XXX XXX or DNS names If you use DNS the system value DOMAIN is appended to the IP Addresses you specify If DOMAIN is null the DNS names must be fully qualified in accordance with IETF RFCs 1034 and 1035 For more information about DNS see DHCP Generic Setup on page 56 and DNS Addressing on page 98 Table 3 Required Network Information Before Installation Per DHCP Server 1 Gateway router IP Address es 2 HTTP server IP Address es 3 Subnetwork mask 4 HTTP server file path HTTPDIR 5 Telephone IP Address range From To 6 DNS server address es If applicable 7 HTTPS server address es If applicable Issue 2 December 2007 27 Network Requirements The default file server file path is the root directory used for all transfers by the server All files are uploaded to or downloaded from this default directory In configurations where the upgrade script and binary files are in the default directory do not use item 4 in Table 3 As the LAN or System Administrator you are also responsible for e Administering the DHCP server as described in Chapter 6 Server Administration e Editing the configuration file on the applicable HTTP or HTTPS file server as covered in 9600 Series SIP IP Telephone Scrip
125. ments IETF documents provide standards relevant to IP Telephony and are available for free from the IETF Web site http www ietf org rfc html ITU Documents Access the ITU Web site for more information about ITU guidelines and documents available for a fee from the ITU Web site http www itu int ISO IEC ANSI IEEE Documents Access the ISO IEC standards Web site for more information about IP Telephony standards guidelines and published documents http www iec ch Issue 2 December 2007 119 Related Documentation 120 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Appendix C Sample Station Forms Use the sample screens that follow as guidelines for telephone setup Figure 1 Station Form Basic Telephone Information add station next Page 1 of X STATION Extension Lock Messages n BCG 0 Type Security Code TN 1 Port Coverage Path 1 COR 1 Name Coverage Path 2 cos 1 Hunt to Station STATION OPTIONS XOIP Endpoint type auto Time of Day Lock Table Loss Group 2 Personalized Ringing Pattern 3 Data Module n essage Lamp Ext 1014 Speakerphone 2 way Mute button enabled y Display Language English Model Expansion Module Survivable GK Node Name Media Complex Ext Survivable COR IP Softphone y Survivable Trunk Dest Remote Office Phone y IP Video Softphone IP Video Customizable Labels Issue 2 December 2007 121 Sample Station For
126. meter Name Default Description and Value Range Value AGCHAND 1 Automatic Gain Control status for handset Values are O disabled 1 enabled AGCHEAD 1 Automatic Gain Control status for headset Values are O disabled 1 enabled AGCSPKR 1 Automatic Gain Control status for speaker Values are O disabled 1 enabled AUDASYS 3 Globally controls audible alerting Values range from 0 through 3 Value 0 or 2 audible alerting off Value 1 or 3 audible alerting on AUDIOENV 0 Audio environment selection index Values range from 0 through 191 AUDIOSTHD 0 Headset sidetone setting Values are 0 Nominal 1 3dB below nominal 2 90B below nominal 15dB below nominal 4 30dB below nominal essentially no sidetone 5 10dB above nominal AUDIOSTHS 0 Handset sidetone setting Values are 0 Nominal 1 3dB below nominal 2 9dB below nominal 15dB below nominal 4 30dB below nominal essentially no sidetone 5 10dB above nominal AUTH 0 Authentication flag for settings file download Values are O secure setting file download is not required 1 secure setting file download is required BAKLIGHTOFF 120 Number of minutes without display activity to wait before turning off the backlight Values range from zero never turn off through 999 minutes 16 65 hours T of 27 74 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series
127. mizeable System Parameters 2 of 2 Telephone Initialization Process These steps offer a high level description of the information exchanged when the telephone initializes and registers This description assumes that all equipment is properly administered ahead of time This description can help you understand how the 9600 Series SIP IP Telephones relate to the routers and servers in your network Step 1 Telephone to Network The telephone is appropriately installed and powered After a short initialization process the telephone identifies the LAN speed and sends a message out into the network identifying itself and requesting further information A router on the network receives and relays this message to the appropriate DHCP server Step 2 Telephone to LLDP Enabled Network An LLDP enabled network provides information to the telephone as described in Link Layer Discovery Protocol LLDP on page 101 Among other data passed to the telephone is the IP Address of the HTTP or HTTPS server Issue 2 December 2007 21 Administration Overview and Requirements Step 3 Telephone to DHCP Server The DHCP server provides information to the telephone as described in DHCP and File Servers on page 53 Among other data passed to the telephone is the IP Address of the HTTP or HTTPS server Step 4 Telephone and File Server The 9600 Series IP Telephones can download script files binary files certificates
128. ms Figure 2 Station Form Feature Options Q hange station nnnn Page 2 of X STATION FEATURE OPTIONS LWC Reception spe Auto Select Any Idle Appearance n LWC Activation Coverage Msg Retrieval y LWC Log External Calls Auto Answer none CDR Privacy Data Restriction n y n n Redirect Notification y Call Waiting Indication Per Button Ring Control n Attd Call Waiting Indication Bridged Call Alerting n Idle Appearance Preference n Switchhook Flash n Bridged Idle Line Preference y Ignore Rotary Digits n Restrict Last Appearance y Active Station Ringing single Conf Trans On Primary Appearance n EMU Login Allowed H 320 Conversion n Per Station CPN Send Calling Number _ Service Link Mode as needed Busy Auto Callback without Flash y Multimedia Mode basic MWI Served User Type _ Display Client Redirection n Automatic Moves AUDIX Name Select Last Used Appearance n A Coverage After Forwarding _ Recall Rotary Digit n Multimedia Early Answer n Remote Softphone Emergency Calls as on local Direct IP IP Audio Connections n Emergency Location Ext 75001 Always use n IP Audio Hairpinning n Precedence Call Waiting y Figure 3 Station Form Call Appearance Info amp Enhanced Call Forwarding add station next Page 3 of x STATION Conf Trans on Primary Appearance y Bridged Appearance Origination Restriction y Call Appearance Display Format loc par
129. multiple devices registered on user s behalf and that this can cause service disruption Values 0 warning disabled 1 warning enabled Enable or disable complete Presence functionality If disabled Presence icons do not show in Contacts or Call History Lists Presence is not displayed to the user incoming Presence updates are ignored and menu items of User Interface to set Presence options are not displayed if available Values are O disabled off 1 enabled on Enable or disable complete Redial functionality If disabled pressing the redial button has no effect and the redial softkeys and menu items are not displayed Values are O disabled 1 enabled 6 of 21 Issue 2 December 2007 79 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Description and Value Range Value ENABLE_REDIAL_LIST 1 Enables or disables the capability to redial out of a list of recently dialed numbers instead of performing last number redial Values are O disabled last number redial only is offered to the user 1 enabled user can select either last number redial or redial from a list ENHDIALSTAT 1 Enhanced Dialing Status Valid range is 0 to 2 If set to O the feature is turned off If set to 1 it is partially enabled dialing rules do not apply for dialing from Contacts If set to 2 the Enhanced Local Dialing feature is fully
130. munity string for all SNMP messages the telephone sends e Restricting dialpad access to Craft Local Procedures to experienced installers and technicians and requiring password entry to access Craft procedures e Restricting the end user s ability to use a telephone Options application to view network data 34 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Other Network Considerations Registration and Authentication A 9600 Series SIP IP Telephone requires an off PBX station OPS extension on Avaya Communication Manager and a login and password on the SES Server to register and authenticate it For more information see the current version of your call server administration manual Issue 2 December 2007 35 Network Requirements 36 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 4 Communication Manager Administration Call Server Requirements Avaya Communication Manager CM extends advanced telephony features to SIP telephones via Outboard Proxy SIP OPS support This feature set offers enhanced calling features in advance of SIP protocol definitions and telephone implementations Before you perform administration tasks ensure that the proper hardware is in place and your call server software is compatible with the 9600 Series SIP IP Telephones Avaya recommends the latest CM software and the latest SIP IP telephone firmware Switch Compatibility As of SIP Release S1 0
131. n y TCP SIGNALING LINK ESTABLISHMENT FOR AVAYA H 323 ENDPOINTS Near End Establishes TCP Signaling Socket y Near End TCP Port Min 61440 Near End TCP Port Max 61444 Figure 11 Stations with Off PBX Telephone Integration Form add off pbx telephone station mapping Page 1 of 2 STATIONS WITH OFF PBX TELEPHONE INTEGRATION Station Application Dial CG Phone Number Trunk Config Extension Prefix Selection Set 43001 EC500 oe eee 1 9736831204 ars 1 43001 OPS _ 4 12345 ars 5 43009 EMU pos tS 67890 aar 2 43011 CSP 998 6095343211 ars 3 126 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Figure 12 IP Codec Set Form change ip codec set n Codec Set Audio Codec G 711MU Silence Suppression y YHA OBWNE Media Encryption 1 aes 2 aea 3 srtp aescml128 hmac80 IP Codec Set Frames Packet Per Pkt Size ms 3 30 Figure 13 IP Codec Set screen page 2 change ip codec set n Page 1 of x Page 2 of x IP Codec Set Allow Direct IP Multimedia Maximum Bandwidth Per Call for Direct IP Multimedia FAX Modem TDD TTY Clear channel Mode relay OLE us Redundancy y 256 Kbits Issue 2 December 2007 127 Sample Station Forms Figure 14 Inter Network Region Connection Management screen change ip network region n Page 3 of x Inter Network Region Connection Management src
132. n quotation marks for proper interpretation The upgrade script file Avaya provides includes a line that tell the telephone to GET 46xxsettings txt This lines causes the telephone to use HTTP to attempt to download the file specified in the GET command If the file is obtained its contents are interpreted as an additional script file That is how your settings are changed from the default settings If the file cannot be obtained the telephone continues processing the upgrade script file If the configuration file is successfully obtained but does not include any setting changes the telephone stops using HTTP This happens when you initially download the script file template from the Avaya support Web site before you make any changes When the configuration file contains no setting changes the telephone does not go back to the upgrade script file Avaya recommends that you do not alter the upgrade script file If Avaya changes the upgrade script file in the future any changes you have made will be lost Avaya recommends that you use the 46xxseitings file to customize your settings instead However you can change the settings file name if desired as long as you also edit the corresponding GET command in the upgrade script file For more information on customizing your settings file see Contents of the Settings File Contents of the Settings File After checking the software the 9600 Series IP Telephone looks for a 46xxsettings file
133. name in the TLV does not contain the substring voice in lower case upper case or mixed case ASCII characters anywhere in the VLAN Name L2Q set to 2 off If T the Tagged Flag is set to 0 set to 1 on if T is set to 1 L2QVLAN set to the VLAN ID in the TLV L2QAUD and L2QSIG set to the Layer 2 Priority value in the TLV DSCPAUD and DSCPSIG set to the DSCP value in the TLV A check is made as to whether a reset is necessary to obtain a new IP address due to a change in the values of the parameters L2Q or L2QVLAN This TLV is ignored if e the value of USE DHCP is 0 and the value of IPADD is not 0 0 0 0 or e the Application Type is not 1 Voice or e the Unknown Policy Flag U is set to 1 SIPPROXYSRVR will be set to the IP Address es in this TLV value TLSSRVR and HTTPSRVR will be set to the IP Address es in this TLV value 104 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Visiting User Administration Table 14 Impact of TLVs on System Parameter Values continued System TLV Parameter Name Name Impact L2Q Proprietary 802 1 If TLV 1 L2Q set to 1 On If TLV 2 L2Q set Q Framing to 2 Off If TLV 3 L2Q set to 0 Auto A check is made as to whether a reset is necessary to obtain a new IP address due to a change in the values of the parameters L2Q or L2QVLAN This TLV is ignored if e the value of USE_DHCP is 0 and the valu
134. nary Files 68 Choosing the Right Binary File and Upgrade Script File 68 Upgrade Script File 2 2 220045022826 048 424028246546 08 69 SS Sais a Oa OE Oe OSS PEGA a ES 69 Contents of the Settings File 2 2 2 eee eee ee ee 70 The GROUP System Value s oc ace sesend ede eee ne ee be wee 72 Chapter 8 Administering Telephone Options 73 Administering Options for the 9600 Series SIP IP Telephones 73 VLAN ConsideralionS o a 6 64 6 PORE RHEE CES DRO ES 94 VLAN TAGGING 244 ce 4A READE HERDS OR HER OOS 94 VLAN Detection 6 6 26 5 bao se ee OE Be ele Sd Be S 95 VLAN Default Value and Priority Tagging 2 2222 eee eae 95 VLAN Separation s sss cb ec ee eee eae ede eee ee eee es 96 DNS Addressihg 362254 eG 64265 e es 4 SSE SS eS SHS Ss SEK 98 IEEE 6021A 144860 ee PH b SD ee SCOR ELE REG HOR REDO CORES ORS 98 802 1X Pass Through and Proxy Logoff 00 0022 ee ae 99 802 1X Supplicant Operation osoa a 99 Link Layer Discovery Protocol LLDP 2 00 2 ee eee eae 101 Visiting User Administration 2 2 ee 105 Emergency Number Administration 2 1 ee ee 106 Local Administrative Craft Options Using the Telephone Dialpad 107 Language Selection bic cde eek eee eden otiet Soe teehee dese 107 Issue 2 December 2007 5 Contents Enhanced Local Dialing gt lt s ss ire ee sidwadneun dene ee iio se eee 108 Enhanced Local Dialing Requirements
135. nd Maintenance Guide Issue 2 December 2007 25 Network Requirements Server Requirements Four server types can be configured for the 9600 Series IP Telephones DHCP server HTTP or HTTPS server SIP Proxy or Registration server Network Time Protocol server for SNTP Note 9600 Series SIP IP Telephones need SIP Enablement Services SES to work properly The SIP Proxy and Registration servers reside on the SES server Avaya Communication Manager CM is considered a feature server behind SES that provides Outboard Proxy SIP OPS features SIP software Release 2 0 supports both SES 4 X and 5 X but when the corresponding server is running SES 4 X the telephones assume only those features compatible with SES 4 X While the servers listed provide different functions that relate to the 9600 Series IP Telephones they are not necessarily different boxes For example DHCP provides network information whereas HTTP provides configuration and application file management yet both functions can co exist on one hardware unit Any standards based server is recommended For parameters related to Avaya Communication Manager information see Chapter 4 Communication Manager Administration For parameters related to DHCP and file servers see Chapter 6 Server Administration A Important The telephones obtain important information from the script files on the server s and depend on the binary file for software upgrades If the
136. ne always properly directs unicast packets from the Authenticator to the telephone or its attached PC as dictated by the MAC address in the packet 98 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 IEEE 802 1X 802 1X Pass Through and Proxy Logoff 9600 Series SIP IP Telephones with a secondary Ethernet interface support pass through of 802 1X packets to and from an attached PC This enables an attached PC running 802 1X supplicant software to be authenticated by an Ethernet data switch The SIP IP Telephones support two pass through modes e pass through and e pass through with proxy logoff The DOT1X parameter setting controls the pass through mode In Proxy Logoff mode DOT1X 1 when the secondary Ethernet interface loses link integrity the telephone sends an 802 1X EAPOL Logoff message on the Ethernet line interface to the data switch on behalf of the attached PC The message alerts the switch that the device is no longer present Proxy logoff occurs only after at least one EAPOL frame with the Port Access Entity PAE group multicast address as the destination MAC address was received on the secondary Ethernet interface The destination MAC address of the proxy EAPOL Logoff frame is the PAE group multicast address The source MAC address of the proxy EAPOL Logoff frame is the same as the source MAC address of the last frame received on the secondary Ethernet interface that had the PAE group multicast address as the
137. nication Manager Administration Chapter 4 Communication Manager Administration Installing and Administering SIP Enablement Services 03 600768 available on the Avaya support Web site http Wwww avaya com support Vendor provided instructions DHCP Server Administration in Chapter 6 Server Administration Option 42 under DHCP Generic Setup Vendor provided instructions http www avaya com support Chapter 7 Telephone Software and Binary Files Chapter 7 Telephone Software and Binary Files The GROUP System Value on page 72 and the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide The applicable Craft Local Procedures in the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide 1 of 2 20 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Table 2 Administrative Checklist continued Task Description Telephone Initialization Process For More Information See Installation of telephones in the network Allow user to modify Set the following parameters in the Options if applicable settings file ENABLE CALL LOG ENABLE CONTACTS ENABLE _MODIFY_CONTACTS ENABLE PRESENCE PROVIDE_OPTIONS SCREEN E a NINE t PROVIDE_LOGOUT Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide 9600 Series SIP IP Telephones Custo
138. o guide you through rest of the setup Click the Next button The Scope Name dialog box displays In the Name field enter a name for the scope such as CM IP Telephones where CM would represent Avaya Communication Manager then enter a brief comment in the Description field When you finish Steps 1 5 click the Next button The IP Address Range dialog box displays Define the range of IP Addresses used by the IP telephones listed in Table 3 Required Network Information Before Installation Per DHCP Server The Start IP Address is the first IP Address available to the IP telephones The End IP Address is the last IP Address available to the IP telephones Note Avaya recommends not mixing 9600 Series IP Telephones and PCs in the same scope Define the subnet mask in one of two ways e The number of bits of an IP Address to use for the network subnet IDs e The subnet mask IP Address Enter only one of these values When you finish click the Next button The Add Exclusions dialog box displays Issue 2 December 2007 63 Server Administration 9 10 15 16 Exclude any IP Addresses in the range specified in the previous step that you do not want assigned to an IP telephone a In the Start Address field under Exclusion Range enter the first IP Address in the range you want to exclude b In the End Address field under Exclusion Range enter the last IP Address in the range you want to
139. ocation and management Differentiated Services an IP based QoS mechanism Domain Name System an IETF standard for ASCII strings to represent IP Addresses The Domain Name System DNS is a distributed Internet directory service DNS is used mostly to translate between domain names and IP Addresses Avaya 9600 Series IP Telephones can use DNS to resolve names into IP Addresses In DHCP TFTP and HTTP files DNS names can be used wherever IP Addresses were available as long as a valid DNS server is identified first Extensible Authentication Protocol or EAP a universal authentication framework frequently used in wireless networks and Point to Point connections defined by RFC 3748 EAP provides some common functions and a negotiation of the desired authentication methods two of which are EAP MD5 and EAP TLS When EAP is invoked by an 802 1X enabled NAS Network Access Server device such as an 802 11 a b g Wireless Access Point modern EAP methods provide a secure authentication mechanism and negotiate a secure PMK Pair wise Master Key between the client and the NAS A TCP IP based protocol for VoIP signaling An alternative to SIP for VoIP signaling One of the two protocols 9600 Series IP Telephones support Hypertext Transfer Protocol used to request and transmit pages on the World Wide Web 1 of 3 Issue 2 December 2007 115 Glossary of Terms HTTPS IETF LAN LLDP MAC Media Channel Encryption NAPT NAT OPS
140. og application If disabled no calls are logged screens related to Call Log are not displayed to user and menu items of User Interface to set Call Log options are not displayed Values are O disabled 1 enabled Enable or disable complete Contact application If disabled no contacts are downloaded during initialization from PPM screens related to Contacts application are not displayed to user and menu items of the User Interface to set Contacts options are hidden Values are O disabled 1 enabled Flag that indicates if SIP early is enabled If enabled and 18x progress message includes early SDP Spark uses that information to open a VoIP channel to the far end before the call is answered Values are 0 disabled 1 enabled Enable or disable G711A codec capability of the phone If the parameter is set to 1 the phone includes G711A capability in an outbound INVITE request and accepts G711A when received in an incoming INVITE request Values are O disabled 1 enabled 5 of 21 78 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range ENABLE_G711U ENABLE_G722 ENABLE_G726 ENABLE_G729 ENABLE_MODIFY_ CONTACTS ENABLE MULTIPLE _ CONTACTS_WARNING ENABLE_PRESENCE ENABLE_REDIAL 1 Enable or dis
141. on Per DHCP Server 13 Click the Add button 14 Click the OK button Activating the New Scope Use the following procedure to activate the new scope 1 In the DHCP console tree click the IP Telephone Scope you just created 2 From the Action menu select Activate The small red down arrow over the scope icon disappears indicating that the scope was activated Issue 2 December 2007 65 Server Administration HTTP Generic Setup You can store the same binary file script file and settings file on an HTTP server as you can on a TFTP server TFTP is not supported for 9600 Series SIP IP Telephones With proper administration the telephone seeks out and uses that material Some functionality might be lost by a reset if the HTTP server is unavailable For more information see DHCP and File Servers on page 53 A Important The files defined by HTTP server configuration must be accessible from all IP telephones invoking those files Ensure that the file names match the names in the upgrade script including case since UNIX systems are case sensitive Note Use any HTTP application you want Commonly used HTTP applications include Apache and Microsoft IIS A Important To set up an HTTP server e Install the HTTP server application e Administer the system parameter HTTPSRVR to the address of the HTTP server Include this parameter in DHCP Option 242 or the appropriate SSON Option e Download the upgrade script
142. on of having a reservation period Assume two IP Addresses therefore two possible DHCP leases Assume three IP telephones two of which are using the two available IP Addresses When the lease for the first two telephones expires the third telephone cannot get a lease until the reservation period expires Even if the other two telephones are removed from the network the third telephone remains without a lease until the reservation period expires 58 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration In Table 10 the 9600 Series IP Telephone sets the system values to the DHCPACK message field values shown Table 10 DHCPACK Setting of System Values System Value Set to DHCP lease time Option 51 if received DHCP lease renew time Option 58 if received DHCP lease rebind time Option 59 if received DOMAIN Option 15 if received DNSSRVR Option 6 if received which might be a list of IP Addresses HTTPSRVR The siaddr field if that field is non zero IPADD The yiaddr field LOGSRVR Option 7 if received MTU_SIZE Option 26 NETMASK Option 1 if received ROUTER Option 3 if received which might be a list of IP Addresses SNTPSRVR Option 42 Windows NT 4 0 DHCP Server Verifying the Installation of the DHCP Server Use the following procedure to verify whether the DHCP server is installed 1 Select Start gt Settings gt Control Panel 2 Double click the
143. on so that the value retrieved from DHCP server has a lower precedence than the value retrieved from the settings file and the value retrieved from the settings file is higher than the value retrieved from PPM then the following determination occurs e f the most recent value the telephone has is from DHCP and new server address information is retrieved from the settings file the telephone will use the new value from the settings file e f later on the telephone receives a new server address value from PPM it will not use this value because PPM s precedence as a data source for the server address is lower than the current value which came from the settings file e f the server to which a specific telephone points is changed manually using the Craft ADDR procedure that value now takes precedence over the previous value 18 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 The Administrative Process Note The only exception to this sequence is in the case of VLAN IDs In the case of VLAN IDs LLDP settings of VLAN IDs are the absolute authority Then the usual sequence applies For the L2EQVLAN and L2Q system values LLDP settings of VLAN IDs are the absolute authority only if the LLDP task receives the VLAN IDs before DHCP and the DHCP client of the telephone is activated at all If the LLDP task receives the VLAN IDs after DHCP negotiation several criteria must be successful before the telephone accepts VLAN IDs
144. ones covered in this document The settings file also includes 9600 Series H 323 IP Telephones 4600 Series IP Telephones and 1600 Series IP Telephones as covered in their respective administrator guides The settings file can include any of five types of statements one per line e Comments which are statements with a character in the first column e Tags which are comments that have exactly one space character after the initial followed by a text string with no spaces e Goto commands of the form GOTO tag Goto commands cause the telephone to continue interpreting the configuration file at the next line after a tag statement If no such statement exists the rest of the configuration file is ignored Issue 2 December 2007 69 Telephone Software and Binary Files e Conditionals of the form IF name SEQ string GOTO tag Conditionals cause the Goto command to be processed if the value of name is a case insensitive equivalent to string If no such name exists the entire conditional is ignored The only system values that can be used in a conditional statement are BOOTNAME GROUP and SIG e SET commands of the form SET parameter_name value Invalid values cause the specified value to be ignored for the associated parameter_name so the default or previously administered value is retained All values must be text strings even if the value itself is numeric a dotted decimal IP Address and so on Note Enclose all data i
145. ons DHCP and File Servers on page 53 and especially HTTP Generic Setup on page 66 Also Chapter 8 Administering Telephone Options Issue 2 December 2007 17 Administration Overview and Requirements General information about administering DHCP servers is covered in DHCP and File Servers on page 53 and more specifically DHCP Server Administration on page 54 General information about administering HTTP servers is covered in DHCP and File Servers and more specifically HTTP Generic Setup Once you are familiar with that material you can administer telephone options as described in Chapter 8 Administering Telephone Options Parameter Data Precedence As shown in Table 1 Administration Alternatives and Options for 9600 Series SIP IP Telephones you can administer a given parameter in a number of ways If a given parameter is administered through different mechanisms the last server to provide the parameter has precedence The precedence from lowest to highest is 1 LLDP 2 DHCP 3 Settings file 4 Personal Profile Manager PPM Note Exception In the case of the parameter SIPDOMAIN the settings file has a higher precedence than PPM 5 Manual administration unless the system parameter USE_DHCP is set to 1 Get IP Address automatically by DHCP or backup file data obtained through PPM For example if the SIP outbound proxy server address is defined to have the precedence informati
146. pt file also points to the settings file You download the upgrade script file sometimes called the script file from http Awww avaya com support This file allows the telephone to use default settings for customer definable options All files must reside in the same directory An alternate upgrade script is also included designed for environments that will support both the H323 and SIP modes of operation For such environments the file needs to be edited in those sections having headings of H 323 EDIT INSTRUCTIONS Specific instructions are provided in the Readme file that accompanies each software bundle Once these changes are made the alternate file should be renamed to 96xxupgrade txt and placed in the HTTP download directory The HTTP download directory holds the telephone backup and application binaries the telephone will download Renaming the alternate file causes any 96xxupgrade txt files residing in that directory to be overwritten Note Avaya recommends that the settings file have the extension txt The Avaya IP Telephones can operate without this file You can also change these settings with DHCP or in some cases from the dialpad of the telephone Settings File The settings file contains the option settings you need to customize the Avaya IP Telephones for your enterprise Note Use one settings file for all your Avaya IP Telephones The settings file includes the 9600 Series SIP IP Teleph
147. r as covered in Chapter 4 Communication Manager Administration Administration on SIP Enablement Services SES as covered in Chapter 5 SIP Enablement Services SES Administration IP Address management for the telephone as covered in Chapter 6 Server Administration for dynamic addressing For static addressing see the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide Tagging Control and VLAN administration for the telephone if appropriate as covered in Chapter 8 Administering Telephone Options Quality of Service QoS administration for the telephone if appropriate QoS is covered in QoS on page 29 and QoS on page 40 Protocol administration for example Simple Network Management Control SNMP and Link Layer Discovery Protocol LLDP Interface administration for the telephone as appropriate Administer the telephone to LAN interface using the PHY1 parameter described in Chapter 3 Network Requirements Administer the telephone to PC interface using the PHY2 parameter described in Interface Control in the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide Application specific telephone administration if appropriate as described in Chapter 8 Administering Telephone Options An example of application specific data is Web specific information required for the optional Web browser application
148. ral no endpoint supports both protocols at the same time Neither protocol is necessarily superior but each offers some unique advantages SIP telephones for example do not require centralized call servers and can route telephone calls when a URL identifies the destination H 323 telephones leverage the call server s presence into the potential availability of hundreds of telephone related features that a standalone SIP telephone cannot provide Signaling 96xx Series IP Telephones ship from the factory with H 323 signaling To use the SIP protocol applicable H 323 96xx Series IP Telephones must be appropriately converted and configured See the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide for detailed conversion configuration information Avaya Communication Manager Release 9600 Series SIP IP Telephones are supported only by Communication Manager Release 4 0 and greater SIP telephones use Avaya OPS Outbound SIP Proxy features on the trunk side of Avaya Communication Manager whereas the H 323 IP telephones are supported on the line side of the Communication Manager When a 9600 Series SIP IP Telephone is running under Communication Manager Release 5 0 an additional feature Extend Call is available Required Servers SIP 9600 Series IP Telephones use two additional servers that H 323 telephones do not SIP Proxy server provided by SIP Enablement Services SE
149. ration 51 SNMP a e i ata i a 0000002 a aa 28 Software a saoao sco sacma rearua de n 67 Software Checklist a a a oa a a 53 Software Telephone aoao aoa 67 SRTP ae goo i a a alae S ee a 15 32 34 89 Station Form Additional Feature Button Assignments 123 Basic Telephone Information 121 Feature Options oaoa 122 Site Data Abbreviated Dialing amp Button A SS JNMEN S s s ee a E 122 Station Form Basic Telephone Information 121 Station Form Call Appearance Info amp Enhanced Call Forwarding oaoa 122 Station Form Feature Options 122 Station Form Site Data Feature Button Assignments Voice Mail 123 StationForms 2 121 Station Forms Samples 121 Station Number Portability 30 Stations With Off PBX Telephone Integration Form 126 Switch Compatibility a a a a a 37 System Parameter Values Impact of TLVs on 104 System Parameters Customizeable 74 T Tagging and VLAN administering 17 TCP UDP Port Utilization oa aaa at Telephone Administration 17 42 Telephone and File Server initialization 22 Telephone and SES Server initialization 22 Telephone Initialization Process a Pal Telephone Options Administering is Telephone Software and Application Files 67 Telephone to Network initialization 21
150. ration to the maximum Enter a sensible name for the Name field such as CM IP Telephones where CM would represent Avaya Communication Manager Click OK A dialog box prompts you Activate the new scope now Click No Note Activate the scope only after setting all options 60 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 DHCP Server Administration Editing Custom Options Use the following procedure to edit custom options 1 ol Highlight the newly created scope 2 Select DHCP Options gt Defaults in the menu 3 4 In the Add Option Type dialog box enter an appropriate custom option name for example Click the New button 9600OPTION Change the Data Type Byte value to String 6 Enter 242 in the Identifier field 7 Click the OK button 8 9 10 11 12 The DHCP Options menu displays Select the Option Name for 242 and set the value string Click the OK button For the Option Name field select 003 Router from the drop down list Click Edit Array Enter the Gateway IP Address recorded in Table 3 Required Network Information Before Installation Per DHCP Server for the New IP Address field 13 Select Add and then OK Adding the DHCP Option Use the following procedure to add the DHCP option l 1 O O ON DO oO fF W PL Highlight the scope you just created Select Scope under DHCP Options Select the 242 option t
151. rial port GROUP Set the group identifier on a per phone basis INT Locally enable or disable the secondary Ethernet hub RESET Reset the telephone to default values including the registration extension and password any values administered through local procedures and values previously downloaded using DHCP or a settings file RESTART Restart the telephone in response to an error condition including the option to reset system values SIG Change the default signaling value to from SIP or change SIG to from H 323 Chapter 2 of the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide also describes how to determine which SIG value is appropriate for your environment SIP Configure SIP call settings VIEW Review the system parameters for the telephone to verify current values and file versions Language Selection 9600 Series IP Telephones are factory set to display information in the English language In addition to English SIP software bundle downloads include the following language files Canadian French Parisian French Latin American Spanish German Brazilian Portuguese Italian Dutch Castilian Spanish Russian Simplified Chinese Japanese Korean Issue 2 December 2007 107 Administering Telephone Options Administrators can specify from one to four languages per telephone to replace English End users can then select which of those languages they want their te
152. rotocol LLDP Link Layer Discovery Protocol LLDP is an open standards layer 2 protocol IP Telephones use to advertise their identity and capabilities and to receive administration from an LLDP server LAN equipment can use LLDP to manage power administer VLANs and provide some administration The transmission and reception of LLDP is specified in IEEE 802 1AB 2005 The 9600 Series IP Telephones use Type Length Value TLV elements specified in IEEE 802 1AB 2005 TIA TR 41 Committee Media Endpoint Discovery LLDP MED ANSI TIA 1057 and Proprietary elements LLDP Data Units LLDPDUs are sent to the LLDP Multicast MAC address 01 80 c2 00 00 0e 9600 Series IP Telephones running SIP Release 2 0 software support IEEE 802 1AB if the value of the configuration parameter LLDP_ENABLED is 1 On or 2 Auto If the value of LLDP_ENABLED is 0 off the transmission and reception of Link Layer Discovery Protocol LLDP is not supported When the value of LLDP_ENABLED is 2 the transmission of LLDP frames will not begin until or unless an LLDP frame is received and the first LLDP frame will be transmitted within 2 seconds after the first LLDP frame is received Once transmission begins an LLDPDU will be transmitted every 30 seconds Note There could be a delay of up to 30 seconds in telephone initialization if the file server address is delivered by LLDP and not by DHCP These telephones e do not support LLDP on the secondary Et
153. rs in dotted decimal format separated by commas with no intervening spaces 0 255 ASCII characters including commas Text string containing the domain name to be used when DNS names in system values are resolved into IP Addresses Valid values are 0 255 ASCII characters Defines the telephone s operational mode for IEEE 802 1X Valid values are 0 Unicast Supplicant operation only with PAE multicast pass through but without proxy Logoff 1 Unicast Supplicant operation only with PAE multicast pass through and proxy Logoff 2 Unicast or multicast Supplicant operation without PAE multicast pass through or proxy Logoff 3 of 27 76 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range DOT1XEAPS MD5 DOT1XSTAT 1 DSCPAUD 46 DSCPSIG 34 DSTOFFSET 1 DSTSTART 2Sun Mar2L Specifies the EAP authentication method s to be used with IEEE 802 1X Comma separated list of key words defining EAP methods In SIP Release 2 0 this value is restricted to a single EAP method Valid values are either MD5 or TLS IEEE 802 1X status Enables disables IEEE 802 1X function and if enabled additionally defines reaction on received multicast or unicast EAPOL messages Valid values are 0 Supplicant operation
154. rty environments only Values are O user cannot select a transfer type transfer type not shown 1 user can select a transfer type transfer type shown Quick login status indicator Specifies whether a password must always be entered manually when the telephone is in a registered and inactive state another telephone is used to take over a primary extension e g SIP visiting User Valid values are 0 manual password eniry is mandatory 1 quick login is enabled a quick login is possible by pressing the Continue softkey on the login screen to accept the current password value Number of seconds for next re registration to SIP server Range in second 10 1 000 000 000 Address es of default router s gateway s in the IP network Range is 7 127 characters defining one or more IP Addresses in dotted decimal format separated by commas without any intervening spaces 15 of 27 88 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Default Value Parameter Name Description and Value Range RTCPCONT 1 RTCPMON Null RTCPMONPORT 5005 RTP_PORT_LOW 5004 RTP_PORT_RANGE 40 SCREENSAVERON 240 SEND_DTMF_ TYPE 2 SIG 0 SIG_PORT_LOW 1024 SIG_PORT_RANGE 64511 SIP_MODE 0 Enables disables the RTCP in parallel to RTP audio streams Values are
155. s Network Assessment Perform a network assessment to ensure that the network will have the capacity for the expected data and voice traffic and that it can support for all applications e SIP e DHCP e HTTP HTTPS and e Jitter buffers Also QoS support is required to run VoIP on your configuration For more information see Appendix B Related Documentation and the QoS parameters L2QAUD L2QSIG DSCPAUD and DSCPSIG in Table 11 9600 Series SIP IP Telephones Customizeable System Parameters Hardware Requirements To operate properly you need e Category 5e cables designed to the IEEE 802 3af 2003 standard for LAN powering e TN2602 IP Media Processor circuit pack Sites with a TN2302 IP Media Processor circuit pack are strongly encouraged to install a TN2602 circuit pack to benefit from the increased capacity e TN799C or D Control LAN C LAN circuit pack A Important IP telephone firmware Release 1 0 or greater requires TN799C V3 or greater C LAN circuit pack s For more information see the Communication Manager Software and Firmware Compatibility Matrix on the Avaya support Web site http Awww avaya com support To ensure that the appropriate circuit pack s are administered on your Communication Manager call server see Chapter 4 Communication Manager Administration For more information about hardware requirements in general see the Avaya one X Deskphone Edition for 9600 SIP IP Telephones Installation a
156. s SIP IP Telephone Administration 2 2 40 Other Network Considerations 28 P Parameter Data Precedence Parameters in Real Time 0 30 Port Utilization Selection 00 04 ee eee TCP UDP 0 0 0 ee ee 31 Q GOS ee sop ee BOE bo ea a ads Be 29 Administrative Parameters IEEE 802 1D and 802 1Q 136 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Index R Registration and Authentication 35 Related Documentation 119 121 Reliability and Performance 29 Requirements sos c a e e a e e 15 Call Server aooaa oF Hardware o a sos os soma coma ee domaa 25 NEIWOPK sacr i a a a aa ae eee at G 25 SOVET y doere athe ao a ata E e e a ahh A 26 RSVP and RTCP saaa aaa 40 RTCP and RSVP oaaae 40 S Sample Station Forms a aoao 121 Scripts and Binary Files for 9600 Series SIP IP Telephones soso cos a a 68 SOCUMILY e a ae es oh a Ge eae h aa A ck Be 34 Server Administration 53 Server Administration DHCP 54 Server Requirements 26 SES Administration 0 ot SES SEVEn fis ide sae ath de e esate UE Sat eee 22 SES Configuring 0 5 51 Setting Up the WML Browser 112 Settings Fille o acc ea asuda sua nud 69 Settings File Contents aoa oa aoaaa 70 SIP Enablement Services SES Administ
157. s SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range L2QVLAN LANGOSTAT LANGUAGES LLDP_ENABLED LOCAL_LOG_LEVEL n a Null 802 1Q VLAN Identifier 0 to 4094 Null is not a valid value and the value cannot contain spaces This parameter is preserved in RAM which survives reset and stored to flash as L2EQVLAN_INIT only upon successful registration This value is initialized from L2EQVLAN_INIT after power up This value will not be initialized from L2QVLAN_INIT after reset but can be modified using the ADDR craft procedure This flag defines whether or not the built in English is offered to the user as selectable item in the language selection UI menu At least one other language file must be downloaded before not offering built in English Values are O not offered 1 selectable List of links to language files to be downloaded Substrings are delimited by commas Maximum length is 1023 characters Each substring shall follow one of the these naming rules A substring is identical to a file name without any prefix specifying the path or server The files are downloaded from the same source as the setting file s A substring can provide a prefix to the file name which specifies the relative path for next higher directory level from the directory the settings file s has been downloaded to the directory the language file shall
158. screen e Adding the Route Pattern to the Numbering Public Unknown Numbering screen e Administering the Proxy Selection Route Pattern on the Locations screen Allowing the system to identify the location of a caller who dials a 911 emergency call from a SIP endpoint on the IP Network Map screen The Administrator Guide for Avaya Communication Manager Document Number 03 300509 provides detailed instructions for administering an IP telephone system on Avaya Communication Manager See Chapter 3 Managing Telephones which describes the process of adding new telephones Also you can locate pertinent screen illustrations and field descriptions in Chapter 19 Screen References of that guide You can find this document on the Avaya support Web site IP Interface and Addresses Follow these general guidelines e Define the IP interfaces for each C LAN and Media processor circuit pack on the switch that uses the IP Interfaces screen For more information see Administration for Network Connectivity for Avaya Communication Manager Document 555 233 504 e On the Customer Options form verify that the IP Stations field is set to y Yes If it is not contact your Avaya sales representative UDP Port Selection The 9600 Series SIP IP Telephones use an even numbered port selected from the interval 4000 to 10000 The telephones cannot be administered from the Avaya Communication Manager Network Region form to support UDP port s
159. se servers are unavailable when the telephones reset the telephones will not operate properly Some features might not be available To restore them you need to reset the telephone s when the file server is available DHCP Server Avaya recommends that a DHCP server and application be installed and that static addressing be avoided Install the DHCP server and application as described in DHCP and File Servers on page 53 26 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Required Network Information HTTP HTTPS Server Administer the HTTP or HTTPS file server and application as described in HTTP Generic Setup on page 66 Network Time Protocol NTP Server SIP IP telephones require NTP server support to set the time and date used in system log time stamps and other time date functions The NTP server is typically needed by one or more servers within the enterprise Administration of the NTP server is beyond the scope of this document Required Network Information Before you administer DHCP and HTTP HTTPS as applicable complete the information in Table 3 If you have more than one router HTTP TLS server and subnetwork mask in your configuration complete Table 3 for each DHCP server The 9600 Series SIP IP Telephones support specifying a list of IP Addresses for a gateway router and the HTTP HTTPS server Each list can contain up to 255 total ASCII characters with IP Addresses separated by com
160. server for network analysis in case of several entries first address always first etc Format is 0 to 255 characters zero or more IP addresses in dotted decimal or DNS name format separated by commas without any intervening spaces Currently set to a maximum of 5 servers Indicates whether end user can personalize button labels Valid values are 0 User cannot change button labels 1 User has ability to change button labels Address of Avaya configuration server currently this parameter when used Is set to the PPM server address Format is dotted decimal or DNS format separated by commas with no spaces 0 255 ASCII characters including commas optionally followed by colon and port number Indicates whether or not secure communication via HTTPS is required to access the configuration server 0 Use HTTP 1 Use HTTPS 2 Use HTTPS if the SIP transport type is TLS otherwise use HTTP 2 of 21 Issue 2 December 2007 75 Administering Telephone Options Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range COUNTRY COVERAGEADDR CURRENT_SKIN DATEFORMAT DAYLIGHT_SAVING_ SETTING_MODE DHCPSTD DIALPLAN DNSSRVR DOMAIN DOT1X USA Null Null YM d Yy 2 Null 0 0 0 0 Null Country of operation for specific dial tone generation The URI to which call coverage is s
161. ssary to set MWISRVR The SIP proxy server is then used for MWI indications Zero to 255 characters zero or more IP addresses in dotted decimal or DNS name format separated by commas without any intervening spaces if operating ina non Avaya environment this value is set via a SET command in the settings file otherwise the address of SIP Proxy server is used Certificate Authority Identifier String identifying whether the endpoints can work with another certificate authority Common name CN for SUBJECT in SCEP certificate request Values are SERIALNO the phone s serial number is included as CN parameter in the SUBJECT of a certificate request MACADDR the phone s MAC address is included as CN parameter in the SUBJECT in the certificate request Common part of SUBJECT in SCEP certificate request String which defines the part of SUBJECT in a certificate request including Organizational Unit Organization Location State Country of 0 to 255 characters starting with and separating items with Private Key length in range of 1024 to 2048 Threshold to renew certificate given as percentage of device certificate s Validity Object Range is 1 to 99 URL of SCEP server String representing zero or one URI starting with http 0 to 255 characters 11 of 21 84 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP I
162. st SET the DNSSRVR and DOMAIN values so you can use those names later in the script Note Administer Options 6 and 15 appropriately with DNS servers and Domain names respectively IEEE 802 1X Certain 9600 Series SIP IP Telephones support the IEEE 802 1X standard for pass through and Supplicant operation but only if the value of the configuration parameter DOT1XSTAT is 1 the default meaning supplicant operation is enabled and the telephone responds only to received unicast EAPOL messages or 2 supplicant operation enabled and telephone responds to received unicast and multicast EAPOL messages If DOT1XSTAT has any other value supplicant operation will not be supported The system parameter DOT1X determines how the telephones handle 802 1X multicast packets and proxy logoff as follows e When DOT1X 0 the default the telephone forwards 802 1X multicast packets from the Authenticator to the PC attached to the telephone and forwards multicast packets from the attached PC to the Authenticator multicast pass through Proxy Logoff is not supported e When DOT 1X 1 the telephone supports the same multicast pass through as when DOT1X 0 Proxy Logoff is supported e When DOT 1X 2 the telephone forwards multicast packets from the Authenticator only to the telephone ignoring multicast packets from the attached PC no multicast pass through Proxy Logoff is not supported Regardless of the DOT1X setting the telepho
163. the Primary Handle field The Primary Handle must be all numeric Set the Host field to the DNS host name of the Avaya SIP Enablement Services Home or Home Edge server to which the telephone will register Check the Add Media Server Extension checkbox and click Add The confirmation screen displays Issue 2 December 2007 51 SIP Enablement Services SES Administration 12 13 14 Click Continue The Add Media Server Extension page displays In the Extension field enter the same extension you entered on page 1 of the Communication Manager Station form This step links the extension recorded in Avaya Communication Manager to the extension recorded in SES See Feature Description and Implementation for Avaya Communication Manager Document Number 555 245 205 for information about Station form entries if necessary Click Add Since the user is being added to Avaya SES Home the Communication Manager CM call server corresponding to the SIP trunk between the CM server and SES Home is selected The confirmation page displays Click Continue Repeat Steps 4 11 for each SIP telephone When you finish configuring all applicable telephones click Update on the left side menu This link appears on the current page whenever updates are outstanding and can be selected at any time to save the administration performed to that point 52 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 6 Ser
164. the telephone attempts to establish communication with its home Personal Profile Manager PPM SIP Enablement Services SES server based on the User Name and Password Assume a situation where the company has multiple locations in London and New York all sharing a corporate IP network Users want to take their telephone functionality from their offices in London to their New York office When users start up their telephones in the new location and enter their credentials the local SES PPM server usually routes them to the local call server With proper administration of the local SES PPM server the telephone knows to try its home SES PPM server the one in London The user can then be automatically registered with the London SES PPM server 30 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Other Network Considerations TCP UDP Port Utilization The 9600 Series SIP IP Telephones use a variety of protocols particularly TCP Transmission Control Protocol UDP User Datagram Protocol and TLS Transport Layer Security to communicate with other equipment in the network Part of this communication identifies which TCP or UDP ports each piece of equipment uses to support each protocol and each task within the protocol For additional TCP UDP port utilization information as it applies to Avaya Communication Manager see UDP Port Selection on page 39 Depending on your network you might need to know what ports or ranges are
165. to from the secondary Ethernet interface receive IEEE 802 1Q tagging treatment The tagging treatment enables frames to be forwarded based on their tags ina manner separate from telephone frames If tags are not changed no tag based forwarding is employed Values are 1 On Enabled 2 Off Disabled This parameter is used with several related parameters For more information see VLAN Separation on page 96 Number of seconds to wait fora DHCPOFFER when using a non zero VLAN ID 1 3 ASCII digits from 0 to 999 19 of 27 92 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Administering Options for the 9600 Series SIP IP Telephones Table 11 9600 Series SIP IP Telephones Customizeable System Parameters continued Parameter Name Default Value Description and Value Range VU_MODE 0 VU_TIMER 36000 WAIT_FOR_ 32 REGISTRATION_TIMER WMLEXCEPT Null WMLHOME Null WMLIDLETIME 10 WMLIDLEURI Null WMLPORT 8080 Visiting User mode Determines if and how the telephone supports Visiting User capabilities 0 Off the telephone operates normally and Visiting User has no essential impact for normal operation 1 Optional the telephone prompts the user at registration time if they are Visiting or Not 2 Forced the telephone only allows Visiting User registrations To Be Determined Time in seconds the SIP application will wait for a register response message If
166. tocols e H 225 for registration admission status RAS and call signaling e H 245 for control signaling e Real Time Transfer Protocol RTP and Secure Real Time Transfer Protocol SRTP e Real Time Control Protocol RTCP and Secure Real Time Control Protocol SRTCP SIP was developed by the IETF Like H 323 SIP provides for real time audio video and data communications transmission over a packet network SIP uses various messages or methods to provide e Registration REGISTER e Call signaling INVITE BYE e Control signaling SUBSCRIBE NOTIFY The 9600 Series SIP IP Telephones support Media Encryption SRTP and use built in Avaya SIP Certificates for trust management Trust management involves downloading certificates for additional trusted Certificate Authorities CA and the policy management of those CAs Identity management is handled by Simple Certificate Enrollment Protocol SCEP with phone certificates and private keys The 9600 Series IP Telephones are loaded with either H 323 or SIP software as part of initial script file administration and initialization during installation Post installation software upgrades automatically download using the proper signaling protocol Issue 2 December 2007 15 Administration Overview and Requirements The parameters under which the 9600 Series SIP IP Telephones need to operate are summarized as follows Telephone Administration on the Communication Manager CM call serve
167. ts and Binary Files Other Network Considerations SNMP The 9600 Series SIP IP Telephones are fully compatible with SNMPv2c and with Structure of Management Information Version 2 SMIv2 The telephones respond correctly to queries from entities that comply with earlier versions of SNMP such as SNMPv1 Fully compatible means that the telephones respond to queries directed either at the MIB II or the read only Custom MIB Read only means that the values therein cannot be changed externally by means of network management tools You can restrict which IP Addresses the telephone accepts SNMP queries from You can also customize your community string with system values SNMPADD and SNMPSTRING respectively For more information see Chapter 6 Server Administration and Table 11 9600 Series SIP IP Telephones Customizeable System Parameters Note SNMP is disabled by default Administrators must initiate SNMP by setting the SNMPADD and SNMPSTRING system values appropriately For more information about SNMP and MIBs see the IETF Web site listed in Appendix B Related Documentation The Avaya Custom MIB for the 9600 Series SIP IP Telephones is available for download in txt format on the Avaya support Web site at http www avaya com support Registration and Authentication A 9600 Series SIP IP Telephone requires an outboard proxy SIP OPS extension on Avaya Communication Manager and a login and password on the S
168. ts the end user from changing or updating the Contact list e PROVIDE_OPTIONS SCREEN If disabled the Options amp Settings menu is not displayed on the Avaya menu The user cannot change any of the features and options associated with the Options amp Settings menu e PROVIDE_NETWORKINFO_SCREEN If disabled the Network Information menu is not displayed on the Avaya menu e PROVIDE LOGOUT If disabled Logout is not displayed to the user as an option on the Avaya menu These parameters have On 1 enabled Off O disabled settings and are described in detail in Table 11 9600 Series SIP IP Telephones Customizeable System Parameters Issue 2 December 2007 111 Administering Applications and Options Note To facilitate administration of application related parameters the 9600 Series both SIP and H 323 and 4600 Series IP Telephones use the same 46xxsettings txt file Avaya A Menu Administration The A Avaya Menu is a list of sub applications the user can select to invoke the corresponding functionality The Avaya Menu contains these entries in this order e Options amp Settings e Browser only if WMLHOME administered in settings file e Network Information e Log Out e About Avaya one X Each individual sub application is listed left justified on an individual Application Line Administering Standard Avaya Menu Entries Options amp Settings is listed if and only if the PROVIDE_OPTIONS_ SCREEN configurat
169. ttings Typical and Maximum Power values OUI 00 40 0D hex Subtype 1 Current conservation level POE_CONS_ MODE Call Server IP Address Subtype 3 Phone IP Address Phone Address Mask Gateway IP Address Subtype 4 CNA Server IP Address in use value from CNASRVR Subtype 5 File Server IP Address Subtype 6 802 1Q Framing 1 if tagging or 2 if not Subtype 7 Not applicable 3 of 3 On receipt of a LLDPDU message the Avaya IP Telephones will act on these TLV elements Issue 2 December 2007 103 Administering Telephone Options Table 14 Impact of TLVs on System Parameter Values System Parameter Name TLV Name Impact PHY2VLAN L2QVLAN and L2Q L2Q L2QVLANID L2QAUD L2QSIG DSCPAUD DSCPSIG SIPPROXYSRVR TLSSRVR and HTTPSRVR IEEE 802 1 Port VLAN ID IEEE 802 1 VLAN Name TIA LLDP MED Network Policy TLV Proprietary Call Server TLV Proprietary File Server TLV System value changed to the Port VLAN identifier in the TLV The system value is changed to the TLV VLAN Identifier L2Q is set to 1 ON A check is made as to whether a reset is necessary to obtain a new IP address due to a change in the values of the parameters L2Q or L2QVLAN VLAN Name TLV is ignored if e the value of USE_DHCP is 0 and the value of IPADD is not 0 0 0 0 or e the current value of LZAQVLAN was set by a TIA LLDP MED Network Policy TLV or e the VLAN
170. ttings file with associates in different groups at the same location IF GROUP SEQ 1 goto GROUP1 IF GROUP SEQ 2 goto GROUP2 specify settings unique to Group 0 goto END GROUP1 specify settings unique to Group 1 goto END GROUP2 specify settings unique to Group 2 END specify settings common to all Groups 72 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Chapter 8 Administering Telephone Options Administering Options for the 9600 Series SIP IP Telephones This chapter explains how to change parameters by means of the DHCP or HTTP servers In all cases you are setting a system parameter in the telephone to a desired value Table 11 lists e the parameter names e their default values e the valid ranges for those values and e adescription of each one Table 11 is a comprehensive list of all the parameters you can configure However you do not have to set every parameter In most cases you will include only those parameters in the settings file that are specific to your own environment and let the telephones use the default values for the remaining ones At a minimum be sure to set these important SIP related parameters SIPPROXYSRVR SIPDOMAIN SNTPSRVR SIPSIGNAL ENABLE_PRESENCE GMTOFFSET DSTOFFSET DSTSTART and DSTSTOP For DHCP the DHCP Option sets these parameters to the desired values as discussed in DHCP and File Servers on page 53 For HTTP the parameters in Table 11 are set to desir
171. uality of Service used to refer to several mechanisms intended to improve audio quality over packet based networks Resource ReSerVation Protocol used by hosts to request resource reservations throughout a network RTP Control Protocol monitors quality of the RTP services and can provide real time information to users of an RTP service Real time Transport Protocol Provides end to end services for real time data such as voice over IP Simple Certificate Enrollment Protocol used to obtain a digital certificate Session Description Protocol A well defined format for conveying sufficient information to discover and participate in a multi session 2of3 116 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 SES Signaling Channel Encryption SIP SNTP SRTCP SRTP TCP IP TFTP TLS TLV UDP Unnamed Registration URI amp URL VLAN VoIP WML SIP Enablement Services the Avaya solution for SIP telephony with Avaya Communication Manager Encryption of the signaling protocol exchanged between the IP telephone and the call server Signaling channel encryption provides additional security to the security provided by channel encryption Session Initiation Protocol an open standard defined initially by IETF RFC 3261 SIP is an alternative to H 323 for VoIP signaling both of which 9600 Series IP Telephones support Simple Network Time Protocol An adaptation of the Network Time Protocol us
172. ue 1 This document was issued for the first time in May 2007 to support the first release of 9600 Series SIP IP Telephones Issue 2 This is the current version of the document revised and issued in December 2007 to support SIP IP Software Release 2 0 This release provides the 9600 SIP IP Telephones with similar functionality to their H323 9600 IP Telephone counterparts despite their signaling protocol differences Release 2 0 introduces several new functions new configuration parameters and adds telephone models 9630G and 9640G What s New in SIP Software Release 2 0 describes this release in more detail Issue 2 December 2007 9 Introduction What s New in SIP Software Release 2 0 New material in this issue to support SIP Release 2 0 software includes New GigE Models Support SIP This release extends SIP capability to two additional telephones the 9630G and 9640G Both models provide built in Gigabyte Ethernet GigE support but are otherwise identical to their 9630 and 9640 SIP IP telephone counterparts Language Support 9600 Series SIP IP Telephones now support 13 languages See Language Selection on page 107 for more information Emergency Button Administrators can now program an Emergency number using the new PHNEMERGNUM parameter Users can dial the Emergency Number whether or not they are logged into the telephone from which they are calling for assistance For more information see Emergency Number
173. ut must be put back on the L2QVLAN VLAN ID you must Reset the telephone See the Reset procedure in the Avaya one X Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide VLAN Separation VLAN separation is available to control priority tagging from the device on the secondary Ethernet typically PC data The following system parameters control VLAN separation e VLANSEP enables 1 or disables 0 VLAN separation e PHY2VLAN provides the VLAN ID for tagged frames received on the secondary Ethernet interface e PHY2PRIO the layer 2 priority value to be used for tagged frames received on the secondary Ethernet interface Table 12 provides several VLAN separation guidelines 96 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Table 12 VLAN Separation Rules If VLAN Considerations Then VLANSEP is 0 OR the telephone is not tagging frames OR the telephone is tagging frames with a VLAN ID equal to PHY2VLAN VLANSEP is 1 On Enabled VLANSEP is 1 On Enabled VLANSEP is 1 On Enabled AND the telephone is not tagging frames OR if the telephone is tagging frames with a VLAN ID equal to PHY2VLAN OR if the PHY2VLAN value is zero AND the telephone is tagging frames with a VLAN ID not equal to PHY2VLAN AND the PHY2VLAN value is not zero Frames received on the secondary Ethernet interface will not be chang
174. ver Administration Software Checklist Ensure that you own licenses to use the DHCP HTTP and HTTPS server software Note You can install the DHCP and HTTP server software on the same machine A Important The firmware in the 9600 Series SIP IP Telephones reserves IP Addresses of the form 192 168 2 x for internal communications The telephone s improperly use addresses you specify if they are of that form DHCP and File Servers Dynamic Host Configuration Protocol DHCP minimizes maintenance for a 9600 Series SIP IP Telephone network by removing the need to individually assign and maintain IP Addresses and other parameters for each telephone on the network The DHCP server provides the following information to the 9600 Series SIP IP Telephones e IP Address of the 9600 Series SIP IP Telephone s e IP Address of the HTTP or HTTPS server e IP Address of the NTP Network Time Protocol server using Option 42 e The subnet mask e IP Address of the router e DNS Server IP Address Administer the LAN so each SIP IP telephone can access a DHCP server that contains the IP Addresses and subnet mask Issue 2 December 2007 53 Server Administration A Important An IP telephone cannot function without an IP Address The failure of a DHCP server at boot time leaves all the affected telephones unusable A user can manually assign an IP Address to an IP telephone When the DHCP server finally returns the telephone never looks for a DHCP ser
175. ver unless the static IP data is unassigned manually In addition manual entry of IP data is an error prone process Avaya recommends that e A minimum of two DHCP servers be available for reliability e A DHCP server be available when the IP telephone reboots e A DHCP server be available at remote sites if WAN failures isolate IP telephones from the central site DHCP server s The file server provides the 9600 Series SIP IP Telephone with a script file and if appropriate new or updated binary software See Step 4 Telephone and File Server on page 22 under Telephone Initialization Process In addition you can edit an associated settings file to customize telephone parameters for your specific environment For more information see Chapter 8 Administering Telephone Options DHCP Server Administration This document concentrates on the simplest case of the single LAN segment Information provided here can be used for more complex LAN configurations A Important Before you start understand your current network configuration An improper installation will cause network failures or reduce the reliability and performance of your network Configuring DHCP for 9600 Series SIP IP Telephones To administer DHCP option 242 make a copy of an existing option 176 for your 46xx IP Telephones You can then either e leave any parameters the 9600 Series SIP IP Telephones do not support for setting via DHCP in option 242 to be
176. ware programs or portions thereof included in the Product may contain software distributed under third party agreements Third Party Components which may contain terms that expand or limit rights to use certain portions of the Product Third Party Terms Information identifying Third Party Components and the Third Party Terms that apply to them is available on Avaya s Web site at http support avaya com ThirdPartyLicense Interference Using a cell mobile or GSM telephone or a two way radio in close proximity to an Avaya IP Telephone might cause interference Security See http support avaya com security to locate and or report known vulnerabilities in Avaya products See http support avaya com to locate the latest software patches and upgrades For information about secure configuration of equipment and mitigation of toll fraud threats see the Avaya Toll Fraud and Security Handbook at http support avaya com Contents Chapter 1 Introduction 000 ee eee ee es 7 About THIS Guide 6 44 44D A ODA AAAS OO 7 Major Differences Between 9600 Series SIP IP and 9600 Series H 323 IP Telephones 8 Features amp Functions Supported by H 323 and Not Supported by SIP 9 Change nisi cise ei eee se ee ede de see bb oT AR OK ESS 9 What s New in SIP Software Release 2 0 0 000 2 ee eee eee 10 Document Organizatlon 2 66 8 se eee cee ieee eee vere wees 13 Other Documentation aoao 13
177. y Page 4 of F EATURES Mode Code for Centralized Voice Mail Multimedia Appl Multimedia Call Handling Multimedia Call Handling Enhanced IP Stations y ISDN Feature Plus y ISDN SIP Network Call Redirection ISDN BRI Trunks ISDN PRI Local Survivable Processor Malicious Call Trace Multifrequency Signaling Server Interface MASI Basic Multimedia IP SIP Trunking You must logoff amp login to effect the permission changes Issue 2 December 2007 K KARR ER Le M R M NS 133 Sample Station Forms Figure 26 System Parameters Customer Options Optional Features screen display system parameters customer options page 5 of x OPTIONAL FEATURES Multinational Locations Station and Trunk MSP n Multiple Level Precedence and Preemption Station as Virtual Extension n Multiple Locations System Management Data Transfer n Personal Station Access PSA y Posted Messages n Tenant Partitioning n PNC Duplication n Terminal Trans Init TTI y Port Network Support y Time of Day Routing y Processor and System MSP n Uniform Dialing Plan y Private Networking y Usage Allocation Enhancements y Processor Ethernet y TN2501 VAL Maximum Capacity y Remote Office n Wideband Switching y Restrict Call Forward Off Net y Wireless n Secondary Data Module y 134 9600 Series SIP IP Telephones Administrator Guide SIP Release 2 0 Index In

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