###################################################################################### ## ## AVAYA IP TELEPHONE CONFIGURATION FILE TEMPLATE ## *** 11 Dec 2018 *** ## ## This file is intended to be used as a template for configuring Avaya IP telephones. ## Parameters supported by software releases up through the following are included: ## ## J100 SIP R4.0.0.0 (J129, J139, J169, J179) ## Avaya Vantage Devices SIP R2.0.0.0 (K155/K165/K175) ## J169/J179 R6.7 H.323 ## Avaya Vantage Basic Application SIP R1.1.0.1 ## Avaya Equinox 3.4 (running on Avaya Vantage Devices) ## 96x1 SIP R7.1.1.0 ## 96x1 H.323 R6.6.5 ## B189 H.323 R6.6.5 ## 96x0 H.323 R3.2.4 ## 96x0 SIP R2.6.14.5 ## H1xx SIP R1.0.2 ## 16xx H.323 R1.3.3 ## ## Note: At the end of the file there is HISTORY TABLE to track changes in this file. ## ###################################################################################### ## ## Any line that does not begin with "SET ", "IF ", "GOTO ", "# " or "GET " is treated as a comment. ## To activate a setting, remove the "## " from the beginning of the line for that parameter so ## that the line begins with "SET ", and change the value to one appropriate for your environment. ## ## To include spaces in a value, the entire value must be enclosed in double quotes, as in: ## SET MYCERTCN "Avaya telephone with MAC address $MACADDR" ## Double quotes (" ASCII 34) shall only be used. Left double quotation mark (“ ASCI 8220) and right double quotation mark (” ASCII 8221) ## shall NOT be used. ## ###################################################################################### ## ## List of MODEL4 values for models which support MODEL4 as testable parameter in the ## configuration file (for example: IF $MODEL4 SEQ 9621 GOTO SETTINGS9621). ## 1603 ## 1608 ## 1616 ## 9610 ## 9620 ## 9630 ## 9640 ## 9650 ## 9670 ## 9608 ## 9611 ## 9621 ## 9641 ## B189 ## J129 ## J139 - Supported by J100 SIP R3.0.0.0 and later ## J169 - Supported by J100 SIP R2.0.0.0 and later, Supported by H.323 R6.7 and later. ## J179 - Supported by J100 SIP R2.0.0.0 and later, Supported by H.323 R6.7 and later. ## H175 ## K155 which represents "Avaya Vantage Entry with camera and hard key pad"; supported in R2.0.0.0 and later. ## K165 which represents "Avaya Vantage without camera" ## K175 which represents "Avaya Vantage with camera" ## ## Note: Avaya Vantage Basic Application (as well to any other Android Avaya Breeze Client SDK based application) running ## on Avaya Vantage Devices retrieves this configuration file after testable parameters ## (such as $GROUP, $MODEL4, $MODEL4, $IPADD, $MACADDR and $SUBNET) were analyzed ## by Avaya Vantage Devices. Therefore, any configuration assigned to specific GROUP, etc will be provided to the ## Avaya Vantage Devices belong to this GROUP, etc and to the Android Avaya Breeze client SDK based application running on them. ## ## Note: Avaya Vantage Basic Application (as well to any other Android Avaya Breeze Client SDK based application) can use ## any configuration parameter defined in this file or even use their own NEW parameters configured ## in such file. It is the application responsibility to extract these parameters from the configuration ## file that Avaya Vantage Devices generates for the application. Only Android Avaya Breeze Client SDK Based application ## can access the configuration file generated by the Avaya Vantage Device according to ## ACTIVE_CSDK_BASED_PHONE_APP configuration parameter. The configuration file generated by the Avaya Vantage devices for ## the Android Avaya Breeze Client SDK based application includes analyzed version of this file, then configuration received ## from Avaya Aura Device Services(if enabled) and then specific configurations parameters received from other sources such ## as DHCP/LLDP/PPM/UI. ## ## Note for Avaya Vantage device and Avaya Vantage Basic Application (as well to any other Android Avaya Breeze Client SDK based application): ## Any parameter configured using this file can also be configured in Avaya Aura Device Services. AADS has higher precedence compare to this file ## download from HTTP/S file server. ## ## Avaya Vantage Open application retrieves all its configuration from the Avaya MPS server. All Avaya Vantage configuration parameters are ## applicable when Avaya Vantage Open application is used unless explicitly stated otherwise below. ## Avaya Vantage open is NOT based on Avaya Breeze Client SDK. ###################################################################################### GET $MACADDR.txt ## SET FQDN_IP_MAP "eastpri01.allstate-lab.avayacloud.com=135.169.56.71,eastsec01.allstate-lab.avayacloud.com=135.169.65.71" ## SET FQDN_IP_MAP "alstl9slamon1.d09-c01.c034.avayacloud.com=135.169.56.127,attpri01.allstate-lab.avayacloud.com=135.169.56.71,attsec01.allstate-lab.avayacloud.com=135.169.71.235" SET FQDN_IP_MAP "alstl1slamon1.d01-c01.c034.avayacloud.com=135.169.65.127,vzpri01.allstate-lab.avayacloud.com=135.169.56.122,vzsec01.allstate-lab.avayacloud.com=135.169.71.241" ## SET FQDN_IP_MAP "ec2-15-207-130-214.ap-south-1.compute.amazonaws.com=15.207.130.214,attpri01.allstate-lab.avayacloud.com=135.169.56.71,attsec01.allstate-lab.avayacloud.com=135.169.65.71" SET SIPDOMAIN allstate-lab.avayacloud.com SET SIP_CONTROLLER_LIST "vzpri01.allstate-lab.avayacloud.com:5061;transport=tls,vzsec01.allstate-lab.avayacloud.com:5061;transport=tls" SET SIPENABLED 1 SET CONFIG_SERVER_SECURE_MODE 2 SET SNMPSTRING avayaphonesnmp SET SNTPSRVR 216.239.35.12,216.239.35.0,216.239.35.4 SET SNTP_SYNC_INTERVAL 5 SET INTER_DIGIT_TIMEOUT 3 SET KEEP_CURRENT_CA 0 SET APPSTAT 1 SET PHNOL "" SET ENABLE_AVAYA_ENVIRONMENT 1 SET WAIT_FOR_REGISTRATION_TIMER 40 ## SET RECOVERYREGISTERWAIT 10 SET REGISTERWAIT 60 SET DISCOVER_AVAYA_ENVIRONMENT 1 SET ENABLE_IPOFFICE 0 SET ASTCONFIRMATION 600 ## SET PROCPSWD 27238 SET PROCPSWD 13579 SET ENABLE_3PCC_ENVIRONMENT 0 SET TRUSTCERTS AllstateLabPubCert.txt SET MSGNUM +15551106000 ## SET FBONCASCREEN 1 SET CC_INFO_TIMER 8 SET BUTTON_MAPPINGS Forward=na,Speaker=cc-release,Hookswitch=na,Headset=na SET WAIT_FOR_CALL_OPERATION_RESPONSE 0 SET HEADSET_PROFILE_DEFAULT 7 ## Below setting is to set the Blue background display SET BACKGROUND_IMAGE_DISPLAY 4 ## Below setting removes from phone menu the option to choose background color SET BACKGROUND_IMAGE_SELECTABLE 1 ## Disabling Enhanced Local Dialing for System Numbers SET ELD_SYSNUM 0 ## Fix for Favorite buttons issue SET SMGR_AUTO_FAVORITE 1 ## SLA Mon Agent is active in Remote Worker mode SET SLMSTAT 2 SET SLMCTRL 2 SET SLMCAP 2 SET SLMPERF 1 SET SLMSRVR 135.169.56.127 ## SET SLMSRVR 135.169.65.127 ## SET SLMSRVR 15.207.130.214 SET SDPCAPNEG 1 SET ENFORCE_SIPS_URI 1 SET MEDIAENCRYPTION 1 SET ENCRYPT_SRTCP 0 SET RTCPCONT 1 SET RTCP_XR 1 SET DTMF_PAYLOAD_TYPE 100 SET ENABLE_EARLY_MEDIA 1 SET DNSSRVR 8.8.8.8,8.8.4.4,64.6.64.6 SET SIMULTANEOUS_REGISTRATIONS 2 SET SIPREGPROXYPOLICY simultaneous SET ENABLE_PPM_SOURCED_SIPPROXYSRVR 1 SET ALLOW_DND_SAC_LINK_CHANGE 0 SET DND_SAC_LINK 1 SET AGENT_ENABLED 1 SET CONF_TRANS_ON_PRIMARY_APPR 1 SET LOCAL_LOG_LEVEL 7 SET SYSLOG_ENABLED 1 SET SYSLOG_LEVEL 7 SET LOG_CATEGORY REG,ALSIP,SIPMESSAGE SET ENABLE_WEBSERVER 1 SET WEBSERVER_ON_HTTP 1 SET FORCE_WEB_ADMIN_PASSWORD Avaya@123 SET CODEC_PRIORITY G711U ## Eastern 51 ## Central 52 ## Moutain 53 ## Pacfic 54 ## Alaska 55 ## Hawaii 56 No DST ## Arizonia 57 No DST ## Belfast 60 ## Engineering 99 # Default TZ is GMT ## SET GMTOFFSET 0:00 IF $GROUP SEQ 51 goto GROUP51 IF $GROUP SEQ 52 goto GROUP52 IF $GROUP SEQ 53 goto GROUP53 IF $GROUP SEQ 54 goto GROUP54 IF $GROUP SEQ 55 goto GROUP55 IF $GROUP SEQ 56 goto GROUP56 IF $GROUP SEQ 57 goto GROUP57 IF $GROUP SEQ 60 goto GROUP60 IF $GROUP SEQ 99 goto GROUP99 # GROUP51 SET GMTOFFSET -5:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP52 SET GMTOFFSET -6:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP53 SET GMTOFFSET -7:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP54 SET GMTOFFSET -8:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP55 SET GMTOFFSET -9:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP56 SET GMTOFFSET -10:00 SET DSTOFFSET 0 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP57 SET GMTOFFSET -7:00 SET DSTOFFSET 0 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END # GROUP60 SET GMTOFFSET 0:00 SET DSTOFFSET 1 SET DSTSTART LSunMar2L SET DSTSTOP LSunOct2L GOTO END # GROUP99 SET GMTOFFSET -6:00 SET DSTOFFSET 1 SET DSTSTART 2SunMar2L SET DSTSTOP 1SunNov2L GOTO END ###################################################################################### ## ## COMMON SETTINGS ## ## Settings in this section will be processed by all telephones, ## but not all parameters are supported by all telephones or all software releases. ## Settings for parameters that are not supported will be ignored. ## For more information, see the Administrator's Guide available at support.avaya.com ## ############### LAYER 2 VLAN AND QOS SETTINGS ############## ## ## L2Q specifies whether layer 2 frames generated by the telephone will have IEEE 802.1Q tags. ## Value Operation ## 0 Auto - frames will be tagged if the value of L2QVLAN is non-zero (default). ## 1 On - frames will always be tagged. ## 2 Off - frames will never be tagged. ## Note: This parameter may also be set via DHCP or LLDP. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later. Note: Value 1 has the same behavior as value 0. ## J169/J179 SIP R1.5.0 - if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## J129 SIP R1.0.0.0 (or R1.1.0.0) ## H1xx SIP R1.0 and later. Note: Value 1 has the same behavior as value 0. ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later; R7.1.0.0 and later, if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET L2Q 0 ## ## L2QVLAN specifies the voice VLAN ID to be used by IP telephones. ## Valid values are 0 through 4094; the default value is 0. ## Note: This parameter may also be set via DHCP or LLDP. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET L2QVLAN 5 ## ## L2QAUD specifies the layer 2 priority value for audio frames generated by the telephone. ## Valid values are 0 through 7; the default value is 6. ## Note: This parameter may also be set via LLDP and H.323 signaling, ## which would overwrite any value set in this file. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET L2QAUD 7 ## ## L2QVID specifies the layer 2 priority value for video frames generated by the telephone. ## Valid values are 0 through 7; the default value is 5. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET L2QVID 7 ## ## L2QSIG specifies the layer 2 priority value for signaling frames generated by the telephone. ## Valid values are 0 through 7; the default value is 6. ## Note: This parameter may also be set via LLDP or H.323 signaling, ## which would overwrite any value set in this file. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET L2QSIG 7 ## ## VLANSEP specifies whether VLAN separation will be enabled by the built-in Ethernet switch ## while the telephone is tagging frames with a non-zero VLAN ID. When VLAN separation is enabled, ## only frames with a VLAN ID that is the same as the VLAN ID being used by the telephone ## (as well as priority-tagged and untagged frames) will be forwarded to the telephone. ## Also, if the value of PHY2VLAN (see below) is non-zero, only frames with a VLAN ID that is ## the same as the value of PHY2VLAN (as well as priority-tagged and untagged frames) will be ## forwarded to the secondary (PHY2) Ethernet interface, and tagged frames received on the ## secondary Ethernet interface will have their VLAN ID changed to the value of PHY2VLAN and ## their priority value changed to the value of PHY2PRIO (see below). ## Value Operation ## 0 Disabled. ## 1 Enabled if L2Q, L2QVLAN and PHY2VLAN are set appropriately (default). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169, J179 only), J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch; see comments for H1xx SIP R1.0 and later. All K155 devices have embedded Ethernet switch. ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later; VLAN separation supported on H1xx have the following exceptions: ## 1. Priority-tagged and untagged frames from the network port will be forwarded to the PC port only when VLANSEP==1, ## H1xx sends tagged packets (L2Q==0 or 1 and VLANTEST==0 or timer < VLANTEST) and L2QVLAN<>0, else to both phone and PC ports. ## 2. No enforcement of PHY2VLAN and PHY2PRIO on tagged VLAN packets received from PC port. If VLANSEP==1, ## H1xx sends tagged packets (L2Q==0 or 1 and VLANTEST==0 or timer < VLANTEST) and 0<>PHY2VLAN<>L2QVLAN<>0 then: ## a. Untagged packets from PC port will be tagged with PHY2VLAN and priority==0. ## b. Tagged packets will be forwarded as tagged packets only if their VLAN equal to PHY2VLAN. ## Otherwise the packets from PC will be sent unmodified. ## Only in case of VLANSEP==1,H1xx sends tagged packets (L2Q==0 or 1 and VLANTEST==0 or timer < VLANTEST) and 0<>PHY2VLAN<>L2QVLAN<>0, ## there will be full separation between PC and phone traffic. In all other cases, PC traffic can reach the phone. ## 3. When VLANSEP ==0, H1xx sends untagged packets even if L2Q==0 or 1 and VLANTEST==0 or timer < VLANTEST. ## 16xx H.323 R1.0 and later ## SET VLANSEP 0 ## ## VLANSEPMODE specifies whether full VLAN separation will be enabled by the built-in Ethernet switch ## while the telephone is tagging frames with a non-zero VLAN ID. This VLAN separation is enabled when: ## VLANSEP=1, L2QVLAN<> PHY2VLAN (and both has value different than 0), L2Q is auto (0) or (1) tagging. ## In this new VLAN separation scheme: ## - Untagged packets from PC port will be forwarded to network port only as untagged packets. ## - Tagged packets from PC port will be forwarded to network port only as tagged packets only in case ## their VLAN is equal to PHY2VLAN. ## In this mode, tagged and untagged packets from PC port will never reach phone’s port. ## - Untagged packets from the network will be sent to the PC port only. ## - Tagged packets from the network port will be sent to the PC port if their VLAN is equal to PHY2VLAN ## and to the phone if their VLAN is equal to L2QVLAN. ## - 802.1x/LLDP and Spanning tree packets are supported as in previous releases in this new mode. ## When VLANSEPMODE is 0, then the VLAN separation is based on previous releases where untagged packets ## from PC port can reach the phone. ## Please note that PHY2PRIO is NOT supported when VLANSEPMODE is 1. ## Value Operation ## 0 Disabled ## 1 Enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, Default is 0. ## J169/J179 SIP R1.5.0 , J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later, Default is 0. ## 96x1 SIP R7.1.0.0 and later, Default is 0. ## 96x1 H.323 R6.6 and later, Default is 0. ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later (J129 only); Default is 1. VLANSEP is not supported by J129. The conditions for VLAN separation mode are ## as described above (except no support for VLANSEP). If one the conditions is not fulfilled then J129 ## will get any tagged/untagged unknown/broadcast/multicast/known DA equal to CPU MAC address packets from the network or PC port. ## SET VLANSEPMODE 1 ## ## PHY2VLAN specifies the VLAN ID to be used by frames forwarded to and from the secondary ## (PHY2) Ethernet interface when VLAN separation (see VLANSEP above) is enabled. ## Valid values are 0 through 4094; the default value is 0. ## Note: This parameter may also be set via LLDP. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch, All K155 devices have embedded Ethernet switch. ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET PHY2VLAN 1 ## ## PHY2PRIO specifies the layer 2 priority value to be used for frames received on the secondary ## (PHY2) Ethernet interface when VLAN separation (see VLANSEP above) is enabled. ## Valid values are 0 through 7; the default value is 0. ## The parameter is not supported when VLANSEPMODE is 1. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET PHY2PRIO 2 ## ## PHY2TAGS specifies whether or not tags will be removed ## from frames forwarded to the secondary (PC) Ethernet interface. ## Value Operation ## 0 Tags will be removed (default) ## 1 Tags will not be removed ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch, All K155 devices have embedded Ethernet switch. ## H1xx SIP R1.0 and later ## 96x1 SIP R6.3 and later ## 96x1 H.323 R6.6 and later ## SET PHY2TAGS 1 ## #################### LAYER 3 QOS SETTINGS ################## ## ## DSCPAUD specifies the layer 3 Differentiated Services (DiffServ) Code Point ## for audio frames generated by the telephone. ## Valid values are 0 through 63; the default value is 46. ## Note: This parameter may also be set via LLDP or H.323 signaling, ## which would overwrite any value set in this file. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Basic Application SIP R1.1.0.1 and later; used in IP office environment only (for Aura environment ## DSCPAUD is taken from PPM and configured using SMGR) ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET DSCPAUD 43 ## ## DSCPVID specifies the layer 3 Differentiated Services (DiffServ) Code Point ## for video frames generated by the telephone. ## Valid values are 0 through 63; the default value is 34. ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.1.0.1 and later; used in IP office environment only (for Aura environment ## DSCPVID is taken from PPM and configured using SMGR) ## H1xx SIP R1.0 and later ## SET DSCPVID 43 ## ## DSCPSIG specifies the layer 3 Differentiated Services (DiffServ) Code Point ## for signaling frames generated by the telephone. ## Valid values are 0 through 63; the default value is 34. ## Note: This parameter may also be set via LLDP or H.323 signaling, ## which would overwrite any value set in this file. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Basic Application SIP R1.1.0.1 and later; used in IP office environment only (for Aura environment ## DSCPSIG is taken from PPM and configured using SMGR) ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET DSCPSIG 41 ## ###################### CALL QUALITY INDICATION SETTINGS ####################### ## ## WBCSTAT and QLEVEL_MIN configuration parameters related to the LOCAL network quality (MAY not be end to end indication). ## ## WBCSTAT specifies whether a wideband codec indication will be displayed when a wideband codec is being used ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.4 and later ## H1xx SIP R1.0 and later ## SET WBCSTAT 0 ## ## QLEVEL_MIN specifies the minimum quality level for which a low local network quality indication will not be displayed ## Value Operation ## 1 Never display icon (default) ## 2 Packet loss is > 5% or round trip network delay is > 720ms or jitter compensation delay is > 160ms ## 3 Packet loss is > 4% or round trip network delay is > 640ms or jitter compensation delay is > 140ms ## 4 Packet loss is > 3% or round trip network delay is > 560ms or jitter compensation delay is > 120ms ## 5 Packet loss is > 2% or round trip network delay is > 480ms or jitter compensation delay is > 100ms ## 6 Packet loss is > 1% or round trip network delay is > 400ms or jitter compensation delay is > 80ms ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.4 and later ## H1xx SIP R1.0 and later ## SET QLEVEL_MIN 4 ## ###################### DHCP SETTINGS ####################### ## ## DHCPSTD specifies whether DHCP will comply with the IETF RFC 2131 standard and ## immediately stop using an IP address if the lease expires, or whether it will ## enter an extended rebinding state in which it continues to use the address and ## to periodically send a rebinding request, as well as to periodically send an ## ARP request to check for address conflicts, until a response is received from ## a DHCP server or until a conflict is detected. ## Value Operation ## 0 Continue using the address in an extended rebinding state (default). ## 1 Immediately stop using the address. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET DHCPSTD 1 ## ## DHCPSTDV6 specifies whether DHCPv6 will comply with the IETF RFC 3155 standard and immediately stop using ## an IPv6 address if the address valid lifetime expires, or whether it will enter an extended rebinding state ## in which it continues to use the address and to periodically send a rebinding request, as well as to periodically send ## a NS (Neighbor Solicitation) request to check for address conflicts, until a REPLY response is received from ## a DHCPv6 server (either a new address, or zero lifetimes, or error status codes) or until a DAD conflict is detected. ## If the address is duplicated, DHCPv6 client transitions into STOPPED state and the phone reboots. ## Value Operation ## 0 Enter proprietary extended rebinding state (continue to use IPv6 address , if DHCPv6 lease expires) (default) ## 1 Comply with DHCPv6 standard (immediately release IPv6 address, if DHCPv6 lease expires) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET DHCPSTDV6 1 ## ## VLANTEST specifies the number of seconds that DHCP will be attempted with a ## non-zero VLAN ID before switching to a VLAN ID of zero (if the value of L2Q is 1) ## or to untagged frames (if the value of L2Q is 0). ## Valid values are 0 through 999; the default value is 60. ## A value of zero means that DHCP will try with a non-zero VLAN ID forever. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (only J169/J179), J139 SIP R3.0.0.0 and later - if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## Avaya Vantage Devices SIP R1.0.0.0 and later. Note: L2Q==1 has the same behavior as L2Q==0. ## J129 SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later. Note: L2Q==1 has the same behavior as L2Q==0. ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later. R7.1.0.0 and later, if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET VLANTEST 90 ## ## REUSETIME specifies the number of seconds that DHCP will be attempted with a VLAN ID of ## zero (if the value of L2Q is 1) or with untagged frames (if the value of L2Q is 0 or 2) ## before reusing the IP address (and associated address information) that it had the last ## time it successfully registered with a call server, if such an address is available. ## While reusing an address, DHCP will enter the extended rebinding state described above ## for DHCPSTD. ## Valid values are 0 and 20 through 999; the default value is 60. ## A value of zero means that DHCP will try forever (i.e., no reuse). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J100 SIP R4.0.0.0 and later: REUSETIME specifies the number of seconds that DHCPV4 or DHCPv6 discovery will be attempted before either: ## reusing the previously cached value of IPv4 address (and associated address information) that the Phone had the last time successfully ## registered with a call server, if such an address is available, or continue discovery DHCPv6 server: IPv6 does not support reuse, ## so there is no corresponding parameter to IPv4's REUSE_IPADD. ## While reusing an address, DHCPV4 will enter the extended rebinding state described for DHCPSTD. ## A value of zero means that DHCP or DHCPv6 will be tried forever (i.e., no reuse). ## H1xx SIP R1.0 and later (REUSE mechanism is supported on Ethernet interface only (not Wi-Fi)) ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R2.5 and later ## SET REUSETIME 90 ## ####################### DNS SETTINGS ####################### ## ## DNSSRVR specifies a list of DNS server addresses. ## Addresses can be in dotted-decimal (IPv4) or colon-hex (IPv6, if supported) ## format, separated by commas without any intervening spaces. ## A value set in this file will replace any value set for DNSSRVR via DHCP. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET DNSSRVR 198.152.15.15 ## ## DOMAIN specifies a character string that will be appended to parameter values ## that are specified as DNS names, before the name is resolved. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET DOMAIN mycompany.com ## ###################### LOGIN SETTINGS ###################### ## ## QKLOGINSTAT specifies whether a password must always be entered manually at the login screen. ## Value Operation ## 0 Manual password entry is mandatory. ## 1 A "quick login" is allowed by pressing the # or Continue key (Default). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R2.0 and later ## SET QKLOGINSTAT 0 ## ## CLEAR_EXTPSWD_ON_LOGOUT specifies whether extension and password are deleted as part of logout. ## Value Operation ## 0 Extension and password are not deleted in case of logout (Default) ## 1 Extension and password are deleted in case of logout ## Note: "quick login" (QKLOGINSTAT ==1) will not be supported when CLEAR_EXTPSWD_ON_LOGOUT==1. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6.3 and later ## B189 H.323 R6.6.3 and later ## SET CLEAR_EXTPSWD_ON_LOGOUT 1 ## ## SHOW_LAST_EXTENSION specifies whether extension is presented after logout. ## Value Operation ## 0 Extension is not presented after logout (Default) ## 1 Extension is presented after logout ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later ## SET SHOW_LAST_EXTENSION 1 ## ## USER_AUTH_FILE_SERVER_URL specifies the user authenticated file server URLs which is used for authentication of user enterprise credentials login, ## using Avaya Aura Device Services (AADS). In addition, the AADS server is used to retrieve configuration (46xxsettings.txt file format), ## picture of the logged-in user, etc. USER_AUTH_FILE_SERVER_URL support comma separated list of URLs without any intervening spaces. ## When USER_AUTH_FILE_SERVER_URL is configured and the login screen is presented, the user is expected to enter the user enterprise credentials ## in the login screen (username,password). ## When USER_AUTH_FILE_SERVER_URL is not configured (the default value is "") and the login screen is presented, the user is expected to enter the SIP credentials ## in the login screen (extension,password). ## The login screen is presented if ACTIVE_CSDK_BASED_PHONE_APP<>"" and the package name defined is installed. ## The default port for https:// is 443. AADS supports only port 443. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## SET USER_AUTH_FILE_SERVER_URL https://aads.service.com:8443 ## Note: Fresh installations of AADS 7.1.2+ will default to port 443. If an older AADS is upgraded to 7.1.2+, it will retain the old 8443 port. ## ## ALLOW_LOGOUT_WHEN_LOCKED specifies whether lock screen will allow logout of existing user. ## Value Operation ## 0 No option to do logout when the device is locked. ## 1 The lock screen provides an option to do logout of the existing user by user/administrator (default) ## 2 The logout option when device is locked is provided through settings application only which is ## accessible for administrator. The logout option is available for administrator only also when ## the device is not locked, but logged-in. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## SET ALLOW_LOGOUT_WHEN_LOCKED 0 ## ################### AVAYA AURA DEVICE SERVICES (AADS) CONTACTS SERVICES ################ ## ## Note: AADS support on Avaya Vantage devices is with unified login only (ACSSSO is enforced internally to 1). ## ## ACSENABLED specifies whether to use contacts from Avaya Aura Device Services (AADS) or not. ## Value Operation ## 0 Contacts from Avaya Aura Device Services (AADS) are NOT used (PPM contacts are used) (default) ## 1 Contacts from Avaya Aura Device Services (AADS) are used (PPM contacts are not used) ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ACSENABLED 1 ## ## ACSSRVR specifies IP address or FQDN of Avaya Aura Device Services (AADS) Contacts Services. Default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ACSSRVR 135.2.2.2 ## ## ACSPORT specifies the port number of Avaya Aura Device Services (AADS) Contacts Services. ## The default value is 443. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ACSPORT 444 ## Note: Fresh installations of AADS 7.1.2+ will default to port 443. If an older AADS is upgraded to 7.1.2+, it will retain the old 8443 port. ## ## ACSSECURE specifies whether to use HTTPS/TLS or HTTP/TCP. ## Value Operation ## 0 Use HTTP/TCP ## 1 Use HTTPS/TLS (default). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ACSSECURE 0 ## ## CONTACT_MATCHING_SEARCH_LOCATION specifies whether to resolve the contact in local contact cache or search the AADS or both. ## Value Operation ## 1 All (default). ## 2 Local ## 3 AADS ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONTACT_MATCHING_SEARCH_LOCATION 2 ## ################### AVAYA MULTIMEDIA MESSAGING ################ ## ## Note: Avaya Multimedia Messaging support on Avaya Vantage devices is with unified login only ESMSSO is enforced internally to 1). ## ## ESMENABLED specifies whether Avaya Multimedia Messaging Service is enabled or not. ## Value Operation ## 0 Avaya Multimedia Messaging Service is disabled (default) ## 1 Avaya Multimedia Messaging Service is enabled ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMENABLED 1 ## ## ESMHIDEONDISCONNECT specifies whether to hide Avaya Multimedia Messaging conversations and message details in the Messages screen and ## Messaging area of the Top Of Mind screen when not connected to Avaya Multimedia Messaging. ## 0: Presents Avaya Multimedia Messaging conversations and message details in the Messages screen and ## Messaging area of the Top Of Mind screen when not connected to Avaya Multimedia Messaging. This is the default. ## 1 Hide Avaya Multimedia Messaging conversations and message details in the Messages screen and ## Messaging area of the Top Of Mind screen when not connected to Avaya Multimedia Messaging. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMHIDEONDISCONNECT 1 ## ## ESMSRVR specifies IP address or FQDN of Avaya Multimedia Messaging server. Default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMSRVR 135.2.2.2 ## ## ESMPORT specifies the port number of Avaya Multimedia Messaging server. Default value is "". ## The default value is 8443. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMPORT 444 ## ## ESMSECURE specifies whether to use TLS or TCP. ## Value Operation ## 0 Use TCP ## 1 Use TLS (default). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMSECURE 0 ## ## ESMREFRESH specifies Messaging refresh interval in minutes. ## Value Operation ## 0 Continuous mode (default value). ## 10,30,60,1000 interval in minutes ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ESMSECURE 10 ## ## ADDRESS_VALIDATION specifies whether messaging address validation is enabled or not. ## Value Operation ## 0 Messaging address validation is disabled (default) ## 1 Messaging address validation is enabled ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET ADDRESS_VALIDATION 1 ## ################### EXCHANGE WEB SERVICES (EWS) ################ ## ## Note: EWS support on Avaya Vantage devices is with unified login only (EWSSSO is enforced internally to 1). ## ## EWSENABLED specifies whether EXCHANGE WEB SERVICES (EWS) is enabled or not. ## Value Operation ## 0 EXCHANGE WEB SERVICES (EWS) is disabled (default) ## 1 EXCHANGE WEB SERVICES (EWS) is enabled ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET EWSENABLED 1 ## ## EWSSERVERADDRESS specifies the Server Address that can be used to connect to EWS directly. Default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET EWSSERVERADDRESS 135.2.2.2 ## ## EWSDOMAIN specifies the Exchange Server domain. Default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET EWSDOMAIN "avaya.com" ## ################### UNIFIED PORTAL ################ ## ## Note: Unified Portal support on Avaya Vantage devices is with unified login only (UNIFIED_PORTAL_SSO is enforced internally to 1). ## ## UNIFIEDPORTALENABLED specifies whether user's Equinox meeting account is enabled or not. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## Avaya Equinox 3.2 and later ## SET UNIFIEDPORTALENABLED 1 ## ################### AVAYA EQUINOX MEETINGS ONLINE (AEMO) ################ ## ## ENABLE_EQUINOX_MEETING_ACCOUNT_DISCOVERY specifies when running on Avaya Vantage whether "Check new services" SK button appears or not. ## There is no auto discovery whether there is Avaya Equinox Meetings Online account or not. ## Note: As Avaya Equinox on Avaya Vantage does not support Avaya Equinox Meeting Online account, this parameter shall be configured to 0. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## Avaya Equinox 3.3.1 and later ## SET ENABLE_EQUINOX_MEETING_ACCOUNT_DISCOVERY 0 ## ################### AVAYA EQUINOX CLOUD ACCOUNTS ################ ## ## ENABLE_AVAYA_CLOUD_ACCOUNTS specifies when running on Avaya Vantage whether Avaya Spaces integration is enabled or not ## Note: As Avaya Equinox on Avaya Vantage does not support Avaya Equinox Cloud Accounts, this parameter shall be configured to 0. ## Value Operation ## 0 Avaya Spaces integration is disabled ## 1 Avaya Spaces integration is enabled (default) ## This parameter is supported by: ## Avaya Equinox 3.4 and later ## SET ENABLE_AVAYA_CLOUD_ACCOUNTS 0 ## ################### SERVER SETTINGS (H.323) ################ ## ## MCIPADD specifies a list of H.323 call server IP addresses. ## Addresses can be in dotted-decimal (IPv4), colon-hex (IPv6, if supported), or ## DNS name format, separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## A value set in this file will replace any value set for MCIPADD via DHCP. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET MCIPADD 135.9.49.202,135.9.10.12,135.9.134.50,135.11.27.15,135.11.28.66 ## ## VUMCIPADD specifies a list of H.323 call server IP addresses for the Visiting User feature. ## Addresses can be in dotted-decimal (IPv4) or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## 96x0 H.323 R3.1.5 and later ## SET VUMCIPADD callsv1.myco.com,callsv2.myco.com,135.42.28.66 ## ## STATIC specifies whether a file server or call server IP address that has been ## manually programmed into the telephone will be used instead of values received ## for TLSSRVR, HTTPSRVR or MCIPADD via DHCP or this settings file. ## Value Operation ## 0 File server and call server IP addresses received via DHCP or ## this file are used instead of manually programmed values (default). ## 1 A manually programmed file server IP address will be used. ## 2 A manually programmed call server IP address will be used. ## 3 A manually programmed file server or call server IP address will be used. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET STATIC 0 ## ## UNNAMEDSTAT specifies whether unnamed registration will be initiated by the telephone ## if a value is not entered at the Extension registration prompt within one minute. ## Unnamed registration provides the telephone with a restricted class of service ## (such as emergency calls) if administered on the call server. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET UNNAMEDSTAT 0 ## ## REREGISTER specifies the delay interval in minutes before and between reregistration attempts. ## Valid values are 1 through 120; the default value is 20. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET REREGISTER 25 ## ## UDT Specifies the Unsuccessful Discovery Timer (UDT) in minutes. ## The Unsuccessful Discovery Timer is the time that the phone perform discovery ## with list of gatekeepers configured and after which the phone will reboot if there is no ## successful discovery with a gatekeeper from the list. ## Valid values are 10 through 960; the default value is 10. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## B189 H.323 R6.6 and later ## SET UDT 960 ## ## H323SIGPROTOCOL specifies which security profiles are enabled with H.323 signaling. ## The phone publishes (in the GRQ message) the list of security profiles configured in H323SIGPROTOCOL. ## The phone ignores responses from call server with security profiles that are not configured in H323SIGPROTOCOL. ## Value Operation ## 0 TLS, Annex-H and Challenge authentication are allowed (default). ## 1 TLS and Annex-H are allowed. ## 2 TLS only is allowed ## Note: The security profile in ip-network-region SAT screen can be configured as "H323TLS" for TLS, "strong" for both TLS and Annex-H, ## "pin-eke" for Annex-H and "challenge" for Challenge authentication. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6.2 and later releases ## SET H323SIGPROTOCOL 1 ## ## GRATARP specifies whether an existing ARP cache entry will be updated with a MAC address ## received in a gratuitous (unsolicited) ARP message. ## Value Operation ## 0 Gratuitous ARP messages will be ignored (default). ## 1 Gratuitous ARP messages will be processed to update an existing ARP cache entry. ## Note: In an H.323 Processor Ethernet Duplication (PE Dup) environment, ## if the PE Dup server and the telephone are in the same subnet, this should be set to 1. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later releases ## B189 H.323 R1.0 and later ## 96x0 H.323 R3.1 and later releases ## SET GRATARP 0 ## ######### GUEST LOGIN (AND VISITING USER) SETTINGS ######### ## ## GUESTLOGINSTAT specifies whether the Guest Login feature is available to users. ## Value Operation ## 0 Guest Login feature is not available to users (default) ## 1 Guest Login feature is available to users ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## 96x1 H.323 R6.0 and later releases ## 96x0 H.323 R2.0 and later releases ## SET GUESTLOGINSTAT 0 ## ## GUESTDURATION specifies the duration (in hours) before a Guest Login or a ## Visiting User login will be automatically logged off if the telephone is idle. ## Valid values are integers from 1 to 12, with a default value of 2. ## Note: Visiting user feature in this context related to H.323 endpoints using VUMCIPADD. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## 96x1 H.323 R6.0 and later releases ## 96x0 H.323 R2.0 and later releases ## SET GUESTDURATION 2 ## ## GUESTWARNING specifies the number of minutes before time specified by GUESTDURATION that ## a warning of the automatic logoff is initially presented to the Guest or Visiting User. ## Valid values are integers from 1 to 15, with a default value of 5. ## Note: Visiting user feature in this context related to H.323 endpoints using VUMCIPADD. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## 96x1 H.323 R6.0 and later releases ## 96x0 H.323 R2.0 and later releases ## SET GUESTWARNING 5 ## ################### APPLICATIONS SETTINGS (SIP) ################ ## ## ACTIVE_CSDK_BASED_PHONE_APP specifies the Android package name (as defined in the application APK manifest file) of active phone application. ## Up to one package name shall be defined. By default the value is "". When ACTIVE_CSDK_BASED_PHONE_APP is defined and the package ## name defined is installed, then Vantage login screen appears before reaching Android home screen. ## This parameter shall only be used when the active phone application is an Avaya Breeze client SDK application ## else it shall remain with default value (""). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## Note: For Avaya Vantage Basic Application ## SET ACTIVE_CSDK_BASED_PHONE_APP "com.avaya.android.vantage.basic" ## Note: For Avaya Equinox Application ## SET ACTIVE_CSDK_BASED_PHONE_APP "com.avaya.android.flare" ## ## PUSH_APPLICATION specifies a list of third party applications (APKs) for installation on Avaya Vantage devices. ## Support a list of URLs. The URL may be specified relative path format ("../" for next higher directory level in relative path format; ## origin is the directory specified by FILE_SERVER_URL or HTTPDIR and TLSDIR depending on download via http or https). ## URL can be also absolute path – in this case it shall begin with http:// or https://. ## Avaya Vantage Basic and Equinox (3.1 and up) applications (APKs) can be pushed as well using this feature. ## The default value is "". ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET PUSH_APPLICATION "com.avaya.android.vantage.basic.apk,flare-android.apk,clock.apk,outlook.apk" ## ## PIN_APP specifies the Android package name (as defined in the application APK manifest file) of the application to be pinned after boot up. ## The default value is "". Non-Avaya CSDK based application that wish to support the PIN_APP feature, then the application manifest file shall ## support the following property: android:lockTaskMode="if_whitelisted". This will ensure that the application will be pinned after reboot even if lock screen is enabled. ## On the other hand, Avaya CSDK based applications (such as Avaya Vantage Basic application) support special handling of pin after initial login ## to prevent pinning of the application without having the login screen. ## Avaya Vantage Basic application supports pinning using PIN_APP. Pinning feature prevents the user from moving to other applications from the pinned application. ## Only administrator can pin or unpin the application from the Avaya Vantage Basic Application User preferences menu using ADMIN_PASSWORD if configured, else PROCPSWD if configured. ## R1.0.0.2 and later - PIN_APP supports list of applications which can be pinned when using an Avaya Vantage Android launcher application for kiosk mode. Up to 6 Android applications can be displayed when ## Avaya Vantage Android launcher application for kiosk mode is used. The Avaya Vantage Android launcher application is part of the ZIP distribution file. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## SET PIN_APP "com.avaya.android.vantage.basic" --> This an example for pinning Avaya Vantage Basic application only. No need for Avaya Vantage Android launcher application. ## SET PIN_APP "com.avaya.android.vantage.basic,com.android.calculator2,com.avaya.endpoint.avayakiosk,com.avaya.endpoint.login,com.avaya.endpoint.upgrade" ## The above example is for case where Avaya Android launcher is used (com.avaya.endpoint.avayakiosk). The applications that are pinned are: Avaya Vantage Basic application and the Android Calculator. ## com.avaya.endpoint.login and com.avaya.endpoint.upgrade,package are required for login and upgrade operations when there is pinning with Avaya Vantage Android launcher. ## ## APPS_CONTROL_FILE specifies third party application XML control file URL (black list and white list of third party applications that can be installed by end user ## from Google Play Store when USER_INSTALL_APPS_GOOGLE_PLAY_STORE is set to 1). ## The default value is "". Up to one URL is supported. The URL may be specified relative path format ("../" for next higher directory level in relative path format; ## origin is the directory specified by FILE_SERVER_URL or HTTPDIR and TLSDIR depending on download via http or https). ## URL can be also absolute path – in this case it shall begin with http:// or https://. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET APPS_CONTROL_FILE https://149.49.77.1/appcontrol.xml ## SET APPS_CONTROL_FILE ../appcontrol.xml ## ## USER_INSTALL_APPS_GOOGLE_PLAY_STORE specifies whether third party applications can be installed by end users/administrators from Google Market Store. ## Value Operation ## 0 Google Play Store is disabled ## 1 Google Play Store is enabled (default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## Note: The parameter is also used by "Avaya Vantage Basic Application" to present "Rate us" menu only when ## USER_INSTALL_APPS_GOOGLE_PLAY_STORE is set to 1. ## SET USER_INSTALL_APPS_GOOGLE_PLAY_STORE 0 ## ## USER_INSTALL_APPS_UNKNOWN_SOURCES specifies whether third party applications can be installed from unknown sources (non-Google Market Store). ## Value Operation ## 0 Installation of third party applications is disabled, user cannot change the status in the settings application (default) ## 1 Installation of third party applications is disabled by default, user can change the status in the settings application. ## 2 Installation of third party applications is enabled by default, user can change the status in the settings application. ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.0.0 and later ## SET USER_INSTALL_APPS_UNKNOWN_SOURCES 0 ## ## ID_CERT_APPLICATION_LIST specifies which applications can access the identity certificate stored on the Avaya Vantage device. ## The default value is "all". ## Value Operation ## "all" all applications can access the identity certificate. User’s shall be able to grant access for a specific application. ## "" NO application can access the identity certificate. There will be no prompt to the users to grant access. This is the securest mode. ## "list of package names" List of all applications that shall be able to access the identity certificate installed. Each approved application will NOT require users approval for such access. ## Non approved application shall not be able to access the identity certificate (users will NOT be able to approve access to the certificate). ## Note: In ALL cases, the active phone application according to ACTIVE_CSDK_BASED_PHONE_APP shall be granted automatically. ## Note: This parameter control access to the identity certificate generated using SCEP or downloaded as PKCS12 file. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET ID_CERT_APPLICATION_LIST "com.games.clock" ## SET ID_CERT_APPLICATION_LIST "" ## ## CERT_INSTALL_APPLICATION_LIST specifies the list of applications that can install trusted certificates and identity certificates on the device. ## The default value is "all". ## Value Operation ## "all" all Android applications can install trusted and client/identity certificates (in addition to Avaya trusted certificates/PKCS12 file download ## and identity certificate generation using SCEP). This is default Android behavior. User will be prompted for the certificate installation and approve it. ## "" In this mode, NO application can install trusted and identity/client certificates. Trusted and client/identity certificates can only be downloaded by ## Avaya trusted certificates/PKCS12 file download and identity certificate generation using SCEP. Users shall not be prompted for certificate installation approval. ## This is the securest mode. ## "list of package names" List of package names of all applications that shall be able to install trusted and client/identity certificates ## (in addition to Avaya trusted certificates/PKCS12 file download and identity certificate generation using SCEP). ## Users shall be prompted for certificate installation approval for the applications listed only. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET CERT_INSTALL_APPLICATION_LIST "com.games.clock" ## SET CERT_INSTALL_APPLICATION_LIST "" ## ################### SECURE ELEMENTS LINUX (SELINUX ) ################ ## ## SELINUX_MODE specifies whether Android Secure Elements Linux (SELinux) is in permissive or enforcing mode. ## Value Operation ## 0 Permissive mode ## 1 Enforcing mode (Default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET SELINUX_MODE 0 ## Note: Please note that changing SELINUX_MODE triggers resets on the Avaya Vantage devices. There is a confirmation message to the end user that reset is about to happen and users can do the reset immediately or later. ## ################# SERVER SETTINGS (SIP) ################ ## Note: Third party SIP call controllers (3PCC) support is only provided by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later. ## ## SIPDOMAIN specifies the domain name to be used during SIP registration. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## Avaya Vantage Basic Application SIP R1.0.0.0 and later; The configuration file from the Avaya Vantage Device ## include the highest precedence value from the following sources (High to low): UI, AADS, this file and PPM. ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET SIPDOMAIN example.com ## ## SIPPORT specifies the port the telephone will open to receive SIP signaling messages. ## Valid values are 1024 through 65535; the default value is 5060. ## This parameter is supported by: ## 96x1 SIP R6.0 and later; Supported up to R6.4.0 (excluded), from R6.4.0 and up to R7.1.0.0 (excluded) SIPPORT is only applied if CONNECTION_REUSE was set to 0 and ## from 7.1.0.0 and later SIPPORT is obsolete. ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## Note: Older SIP software releases also use the value of this parameter as the ## destination port for transmitted SIP messages. However, for newer releases ## that support SIP_CONTROLLER_LIST (see below), the value of that parameter ## is used to specify the destination port for transmitted SIP messages. ## SET SIPPORT 5060 ## ## SIP_CONTROLLER_LIST specifies a list of IPv4 SIP controller designators, ## separated by commas without any intervening spaces. ## The list is used on IPv4-only and dual mode phones (if SIP_CONTROLLER_LIST_2 is not provided). ## Each controller designator has the following format: ## host[:port][;transport=xxx] ## host is an IP address in dotted-decimal (DNS name format is not supported unless stated otherwise below). ## [:port] is an optional port number. ## [;transport=xxx] is an optional transport type where xxx can be tls, tcp or udp. ## If a port number is not specified a default value of 5060 for TCP and UDP or 5061 for TLS is used. ## If a transport type is not specified, a default value of tls is used. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0); J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later; DNS name format is supported for 3PCC environment only. ## For 3PCC environment, only one SIP controller is supported. ## J100 SIP R4.0.0.0 and later; used on dual mode phones if SIP_CONTROLLER_LIST_2 is not provided. ## When 3PCC_SERVER_MODE = 1 (a BroadSoft server), SIP_CONTROLLER_LIST should contain one sip controller entry and host should be an FQDN (DNS name format). ## The FQDN would resolve to primary and alternate servers to support redundant configuration. ## When 3PCC_SERVER_MODE = 0 (a generic SIP server), SIP_CONTROLLER_LIST may contain one or two sip controller entries (to support redundant configuration). ## “host” of sip controller entry could be an FQDN(DNS name format) or an IP address. If FQDN is provided, it will resolve to one primary server. ## IPv6 is not supported for 3PCC environment. In 3PCC environment, there is no support for resolving an FQDN to an IPv6 address in the SIP_CONTROLLER_LIST or SIP_CONTROLLER_LIST_2. ## IPv6 is supported for Aura environment and in Aura there is no support for FQDN yet (only IP addresses can be configured). ## J169/J179 SIP R1.5.0 ## Avaya Equinox 3.1.2 and later; DNS name format is supported. ## Avaya Vantage Devices SIP R1.0.0.0 and later; DNS name format is supported; UDP is not supported; not applicable when Avaya Vantage Open application is used. ## Avaya Vantage Basic Application SIP R1.0.0.0 and later; DNS name format is supported; UDP is not supported. The ## configuration file from the Avaya Vantage Device combines the configuration of this parameter from all sources (in the following order): ## UI, LLDP, DHCP, this file, PPM and AADS. ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.4.1 and later ## H1xx SIP R1.0 and later; udp is not supported. ## SET SIP_CONTROLLER_LIST proxy1:5555;transport=tls,proxy2:5556;transport=tls ## ## SIP_CONTROLLER_LIST_2 ## Valid Values ##    String The comma separated list of SIP proxy/registrar servers ##    0 to 255 characters: zero or more IP addresses in dotted decimal or colon-hex format, ## separated by commas without any intervening spaces. ## Default: "" (null) ## Description ##     This parameter replaces SIP_CONTROLLER_LIST for dual mode phones. It is used on IPv6-only phones to provide the list of SIPv6 servers. ## SIPv4 servers are ignored in IPv6-only mode. It is used to select the registration address. ## The list has the following format: host[:port][;transport=xxx] ## where: ## - host: is an IP addresses in dotted-decimal format or hex format ## - port: is the optional port number. If a port number is not specified the default ## value (5060 for TCP, 5061 for TLS) will be used ## - transport: is the optional transport type (where xxx is tls or tcp) ## If a transport type is not specified the default value TLS will be used ## A dual mode controller has addresses of both families within curly brackets. ## A settings file example is: ## SIP_CONTROLLER_LIST_2 "{[2007:7::5054:ff:fe35:c6e]:5060;transport=tcp, 47.11.15.142:5060;transport=tcp}, ## {[2007:7::5054:ff:fe80:d4b0]:5060;transport=tcp, 47.11.15.174:5060;transport=tcp}" ## Dual mode phones use SIGNALING_ADDR_MODE to select SM IP addresses from SIP_CONTROLLER_LIST_2. ## If SIGNALING_ADDR_MODE is 4, register to the first IPv4 address in SIP_CONTROLLER_LIST_2. ## IPv4 only phones use SIP_CONTROLLER_LIST. Dual mode phones use SIP_CONTROLLER_LIST if SIP_CONTROLLER_LIST_2 is not provided. ## SIP_CONTROLLER_LIST_2 should only be used if IPv6 addresses (FQDN is not supported) may be used for SIP signaling. ## SIP_CONTROLLER_LIST_2 should not be used if FQDN (DNS name format) is used for sip controllers. ## ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Example: ## Dual mode SIP controllers: ## SET SIP_CONTROLLER_LIST_2 "{[2007:7::5054:ff:fe35:c6e]:5060;transport=tcp,47.11.15.142:5060;transport=tcp}, ## {[2007:7::5054:ff:fe80:d4b0]:5060;transport=tcp, 47.11.15.174 :5060;transport=tcp}" ## IPv6-only mode SIPv6 controllers: ##  SET SIP_CONTROLLER_LIST_2 "[2007:7::5054:ff:fe35:c6e]:5060;transport=tcp,[2007:7::5054:ff:fe80:d4b0]:5060;transport=tcp" ## ## SIP Transport UDP ## Determines whether SIP Transport = UDP can be manually configured on the phone. ## 0 for No (default) ## 1 for Yes ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_UDP_TRANSPORT 1 ## ## SIPREGPROXYPOLICY specifies whether the telephone will attempt to maintain ## one or multiple simultaneous registrations. ## Value Operation ## alternate Only a single registration will be attempted and maintained. ## simultaneous Simultaneous registrations will be attempted and maintained with all available controllers. ## This parameter is supported by: ## J129 SIP R1.0.0.0 or R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, the default is simultaneous. ## The parameter shall be configured to "alternate" in IP Office and 3PCC environments only. ## Not supported in 96x1 SIP R6.2 and later; the default value is simultaneous. ## 96x1 SIP R6.0.x; the default value is alternate. ## 96x0 SIP R2.4.1 and later; the default value is alternate. ## SET SIPREGPROXYPOLICY simultaneous ## ## SIMULTANEOUS_REGISTRATIONS specifies the number of Session Managers ## with which the telephone will simultaneously register. ## Valid values are 1, 2 or 3; the default value is 3. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Basic Application SIP R1.1.0.1 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.6 and later ## H1xx SIP R1.0 and later; For IP office environment this parameter shall be set to 1. ## SET SIMULTANEOUS_REGISTRATIONS 3 ## ## CONNECTION_REUSE specifies whether the telephone will use two UDP/TCP/TLS connection (for both outbound ## and inbound) or one UDP/TCP/TLS connection. ## Value Operation ## 0 - disabled, the phone will open oubound connection to the SIP Proxy and listening socket for inbound connection ## from SIP proxy in parallel. This is the only and default behavior for pre-6.4 releases. ## 1 - enabled, the phone will not open a listening socket and will maintain and re-use the sockets it creates with ## the outbound proxies (default) ## For IP office environment this parameter shall be set to 1 (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - only value 1 is supported. ## 96x1 SIP R6.4 and later up to R7.1.0.0 (excluded) - values 0 and 1 are supported, R7.1.0.0 and later only value 1 is supported. ## H1xx SIP R1.0 and later ## SET CONNECTION_REUSE 0 ## ## ENABLE_PPM_SOURCED_SIPPROXYSRVR parameter enables PPM as a source of SIP proxy server information. ## Value Operation ## 0 Proxy server information received from PPM will not be used. ## 1 Proxy server information received from PPM will be used (default). ## This parameter is not supported in IP Office environment as PPM is not supported. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.4.1 and later ## H1xx SIP R1.0 and later ## SET ENABLE_PPM_SOURCED_SIPPROXYSRVR 1 ## ## CONFIG_SERVER specifies the address of the Avaya configuration server. ## Zero or one IP address in dotted decimal or DNS name format, ## optionally followed by a colon and a TCP port number. ## The value may contain 0 to 255 characters; the default value is null (""). ## This parameter is not supported in IP Office environment as PPM is not supported. ## This parameter is supported by: ## 96x0 SIP R2.6.7 and later ## H1xx SIP R1.0 and later ## SET CONFIG_SERVER ppm.myco.com ## ## CONFIG_SERVER_SECURE_MODE specifies whether HTTP or HTTPS is used to access the configuration server. ## Value Operation ## 0 use HTTP (default for 96x0 R2.0 through R2.5) ## 1 use HTTPS (default for other releases and products) ## 2 use HTTPS if SIP transport mode is TLS, otherwise use HTTP ## This parameter is not supported in IP Office environment as PPM is not supported. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.0 and later ## SET CONFIG_SERVER_SECURE_MODE 1 ## ## VOLUME_UPDATE_DELAY specifies the minimum interval, in seconds, between backups of the volume levels to PPM service ## when the phone registered to Avaya Aura Session Manager. If no change to volume levels, there will be no backup to PPM service. ## Valid values are 2 through 900; the default value is 2. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.0.1 and later ## SET VOLUME_UPDATE_DELAY 20 ## ## SIPPROXYSRVR specifies a list of addresses of SIP proxy servers. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## 96x0 SIP R1.0 through R2.4 ## SET SIPPROXYSRVR 192.168.0.8 ## ## SIPSIGNAL specifies the type of transport used for SIP signaling. ## Value Operation ## 0 UDP ## 1 TCP ## 2 TLS (default) ## This parameter is supported by: ## 96x0 SIP R1.0 through R2.4 ## SET SIPSIGNAL 2 ## ## SIP_PORT_SECURE specifies the destination TCP port for SIP messages sent over TLS. ## Valid values are 1024 through 65535; the default value is 5061. ## The parameter is used in non-Avaya environment. In Avaya environment, this ## parameter will be overwritten by PPM configuration. ## This parameter is supported by: ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 through R2.4 ## H1xx SIP R1.0 and later ## SET SIP_PORT_SECURE 5061 ## ## ENABLE_AVAYA_ENVIRONMENT specifies whether the telephone is configured ## for use in an Avaya (SES) or a third-party proxy environment. ## Value Operation ## 0 3rd party proxy with "SIPPING 19" features ## 1 Avaya SES with AST features and PPM (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later; for IP office and 3PCC environments this parameter shall be set to 0. ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; for IP office environment this parameter shall be set to 0. ## 96x0 SIP R1.0 through R2.4 ## SET ENABLE_AVAYA_ENVIRONMENT 1 ## ## ######### NON-AVAYA ENVIRONMENT SETTINGS (SIP ONLY) ######## ## ## MWISRVR specifies a list of addresses of Message Waiting Indicator servers. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## 96x0 SIP R2.0 and later ## SET MWISRVR 192.168.0.7 ## ## DIALPLAN specifies the dial plan used in the telephone. ## It accelerates dialing by eliminating the need to wait for ## the INTER_DIGIT_TIMEOUT timer to expire. ## The value can contain 0 to 1023 characters; the default value is null (""). ## See the telephone Administrator's Guide for format and setting alternatives. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.0 and later ## H1xx SIP R1.0 and later ## SET DIALPLAN [23]xxxx|91xxxxxxxxxx|9[2-9]xxxxxxxxx ## ## PHNNUMOFSA specifies the number of Session Appearances the telephone ## should support while operating in a non-Avaya environment. ## Valid values are 1 through 10; the default value is 3. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.0 and later ## H1xx SIP R1.0 and later ## SET PHNNUMOFSA 3 ## ################## TIME SETTINGS (SIP ONLY) ################# ## ## SNTPSRVR specifies a list of addresses of SNTP servers. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later; ## the default is changed to "0.avaya.pool.ntp.org,1.avaya.pool.ntp.org,2.avaya.pool.ntp.org,3.avaya.pool.ntp.org" in R2.0.0.0 and later. ## Avaya Vantage Devices SIP R1.0.0.0 and later; the default is changed to "0.avaya.pool.ntp.org,1.avaya.pool.ntp.org,2.avaya.pool.ntp.org,3.avaya.pool.ntp.org" in R2.0.0.0 and later. FQDN is supported in R2.0.0.0 and later. ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET SNTPSRVR 192.168.0.5 ## ## SNTP_SYNC_INTERVAL specifies the time interval in minutes at which the phone will attempt to synchronize its time with configured NTP servers. ## Valid values: 60-2880 (minutes), Default: 1440 minutes (1 day). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET SNTP_SYNC_INTERVAL 100 ## ## GMTOFFSET specifies the time offset from GMT in hours and minutes. ## The format begins with an optional "+" or "-" ("+" is assumed if omitted), ## followed by 0 through 12 (hours), followed by a colon (:), ## followed by 00 through 59 (minutes). The default value is 0:00. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 only (TIMEZONE shall be used in R1.0.0.1 and later) ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET GMTOFFSET 0:00 ## ## DSTOFFSET specifies the time offset in hours of daylight savings time from local standard time. ## Valid values are 0, 1, or 2; the default value is 1. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 only (TIMEZONE shall be used in R1.0.0.1 and later) ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET DSTOFFSET 1 ## ## DSTSTART specifies when to apply the offset for daylight savings time. ## The default value for all telephones is 2SunMar2L ## (the second Sunday in March at 2AM local time). ## See the Administrator's Guide for format and setting alternatives. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 only (TIMEZONE shall be used in R1.0.0.1 and later) ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET DSTSTART 2SunMar2L ## ## DSTSTOP specifies when to stop applying the offset for daylight savings time. ## The default value for all telephones is 1SunNov2L ## (the first Sunday in November at 2AM local time). ## See the Administrator's Guide for format and setting alternatives. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 only (TIMEZONE shall be used in R1.0.0.1 and later) ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET DSTSTOP 1SunNov2L ## ## TIMEZONE specifies timezone configuration in Olson name format as appears in the tzone database ## maintained by IANA. Default value is "Etc/GMT" which means +00:00. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0.0.1 and later ## SET TIMEZONE America/New_York ## SET TIMEZONE Asia/Jerusalem ## ################## TIMER SETTINGS (SIP ONLY) ############### ## ## WAIT_FOR_REGISTRATION_TIMER specifies the number of seconds that the telephone will wait ## for a response to a REGISTER request. If no response message is received within this time, ## registration will be retried based on the value of RECOVERYREGISTERWAIT. ## Valid values are 4 through 3600; the default value is 32. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.5 and later ## Note: For Avaya Distributed Office configurations prior to R2.0, this parameter must be set to 60. ## SET WAIT_FOR_REGISTRATION_TIMER 60 ## ## REGISTERWAIT specifies the number of seconds between re-registrations with the current server. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later; valid values are 30 to 86400; the default value is 900. ## Avaya Vantage Basic Application SIP R1.1.0.0 and later ## J169/J179 SIP R1.5.0; valid values are 30 to 86400; the default value is 900. ## 96x1 SIP R6.0 and later; valid values are 30 to 86400; the default value is 900. ## H1xx SIP R1.0 and later; valid values are 30 to 86400; the default value is 900. ## 96x0 SIP R2.4.1 and later; valid values are 30 to 86400; the default value is 900. ## 96x0 SIP R1.0 through R2.2; valid values are 10 to 1000000000; the default value is 3600. ## SET REGISTERWAIT 1000 ## ## RECOVERYREGISTERWAIT specifies a number of seconds. ## If no response is received to a REGISTER request within the number of seconds specified ## by WAIT_FOR_REGISTRATION_TIMER, the telephone will try again after a randomly selected ## delay of 50% to 90% of the value of RECOVERYREGISTERWAIT. ## Valid values are 10 through 36000; the default value is 60. ## This parameter is supported by: ## 96x1 SIP R6.0 and later; not supported in R6.2 and later. ## 96x0 SIP R2.4.1 and later ## SET RECOVERYREGISTERWAIT 90 ## ## WAIT_FOR_UNREGISTRATION_TIMER specifies the number of seconds that the telephone will wait ## before assuming that an un-registration request is complete. ## Un-registration includes termination of registration and all active dialogs. ## Valid values are 4 through 3600; the default value is 32. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.5 and later ## SET WAIT_FOR_UNREGISTRATION_TIMER 45 ## ## WAIT_FOR_INVITE_RESPONSE_TIMEOUT specifies the maximum number of seconds that the ## telephone will wait for another response after receiving a SIP 100 Trying response. ## Valid values are 30 through 180; the default value is 60. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later. ## H1xx SIP R1.0 and later. ## SET WAIT_FOR_INVITE_RESPONSE_TIMEOUT 90 ## ## OUTBOUND_SUBSCRIPTION_REQUEST_DURATION specifies the duration in seconds requested by the ## telephone in SUBSCRIBE messages, which may be decreased in the response from the server. ## Valid values are 60 through 31536000 (one year); the default value is 86400 (one day). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later. ## H1xx SIP R1.0 and later ## 96x0 SIP R2.0 and later. ## SET OUTBOUND_SUBSCRIPTION_REQUEST_DURATION 604800 ## ## NO_DIGITS_TIMEOUT specifies the number of seconds that the telephone will wait ## for a digit to be dialed after going off-hook before generating a warning tone. ## Valid values are 1 through 60; the default value is 20. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.0 and later ## SET NO_DIGITS_TIMEOUT 15 ## ## INTER_DIGIT_TIMEOUT specifies the number of seconds that the telephone will wait ## after a digit is dialed before sending a SIP INVITE. ## Valid values are 1 through 10; the default value is 5. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.0 and later ## SET INTER_DIGIT_TIMEOUT 6 ## ## FAILED_SESSION_REMOVAL_TIMER specifies the number of seconds the telephone will ## display a session line appearance and generate re-order tone after an invalid ## extension has been dialed if the user does not press the End Call softkey. ## Valid values are 5 through 999; the default value is 30. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET FAILED_SESSION_REMOVAL_TIMER 15 ## ## TCP_KEEP_ALIVE_STATUS specifies whether or not the telephone sends TCP keep alive messages. ## Value Operation ## 0 Keep-alive messages are not sent ## 1 Keep-alive messages are sent (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET TCP_KEEP_ALIVE_STATUS 0 ## ## TCP_KEEP_ALIVE_TIME specifies the number of seconds that the telephone will wait ## before sending out a TCP keep-alive (TCP ACK) message. ## Valid values are 10 through 3600; the default value is 60. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET TCP_KEEP_ALIVE_TIME 45 ## ## TCP_KEEP_ALIVE_INTERVAL specifies the number of seconds that the telephone will wait ## before re-transmitting a TCP keep-alive (TCP ACK) message. ## Valid values are 5 through 60; the default value is 10. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET TCP_KEEP_ALIVE_INTERVAL 15 ## ## CONTROLLER_SEARCH_INTERVAL specifies the number of seconds the telephone will wait ## to complete the maintenance check for monitored controllers. ## Valid values are 4 through 3600. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (default value is 16) ## 96x1 SIP R6.0 and later (default value is 16) ## H1xx SIP R1.0 and later (default value is 16) ## 96x0 SIP R2.6.5 and later (default value is 16) ## 96x0 SIP R2.4.1 - R2.6.4 (default value is 4) ## SET CONTROLLER_SEARCH_INTERVAL 20 ## ## ASTCONFIRMATION specifies the number of seconds that the telephone will wait to validate ## an active subscription when it SUBSCRIBEs to the "avaya-cm-feature-status" package. ## Valid values are 16 through 3600. ## This parameter is not supported in IP Office and 3PCC environments as there is no subscription to avaya-cm-feature-status. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later; the default value is 32. ## 96x1 SIP R6.0 and later; the default value is 32. ## H1xx SIP R1.0 and later; the default value is 32. ## 96x0 SIP R2.6 and later; the default value is 60. ## SET ASTCONFIRMATION 90 ## ## FAST_RESPONSE_TIMEOUT specifies the number of seconds that the telephone will wait ## before terminating an INVITE transaction if no response is received. ## However, a value of 0 means that this timer is disabled. ## Valid values are 0 through 32; the default value is 4. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later - it is provided by SMGR for phones connected to Avaya Aura however the settings file ## configuration is still applicable for non-Avaya Aura systems. ## 96x1 SIP 6.0 and later. In 96x1 SIP R6.2 it is provided by SMGR for phones connected to Avaya Aura however the settings file ## configuration is still applicable for non-Avaya Aura systems. ## 96x0 SIP R2.4.1 and later ## SET FAST_RESPONSE_TIMEOUT 5 ## ## RDS_INITIAL_RETRY_TIME specifies the number of seconds that the telephone will wait ## the first time before trying to contact the PPM server again after a failed attempt. ## Each subsequent retry will be delayed by double the previous delay. ## Valid values are 2 through 60, the default value is 2. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.4.1 and later ## SET RDS_INITIAL_RETRY_TIME 4 ## ## RDS_MAX_RETRY_TIME specifies the maximum delay interval in seconds after which ## the telephone will abandon its attempt to contact the PPM server. ## Valid values are 2 through 3600, the default value is 600. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.4.1 and later ## SET RDS_MAX_RETRY_TIME 600 ## ## RDS_INITIAL_RETRY_ATTEMPTS specifies the number of retries after which ## the telephone will abandon its attempt to contact the PPM server. ## Valid values are 1 through 30, the default value is 15. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.4.1 and later ## SET RDS_INITIAL_RETRY_ATTEMPTS 20 ## ## SIP Timer T1 is an estimate of the Round Trip Time (RTT) and is defined in milliseconds. ## Valid values are 500 through 10000 milliseconds; the default value is 500. ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET SIP_TIMER_T1 2000 ## ## SIP Timer T2 is maximum retransmit interval for non-INVITE requests and INVITE responses and is defined in milliseconds. ## Valid values are 2000 through 40000 milliseconds; the default value is 4000. ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET SIP_TIMER_T2 5000 ## ## SIP Timer T4 is maximum duration a message will remain in the network and is defined in milliseconds. ## Valid values are 2500 through 60000 milliseconds; the default value is 5000. ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET SIP_TIMER_T4 6000 ## ## FORBIDDEN_SESSION_REMOVAL_TIMER specifies the duration of an off-hook ## session before call is automatically ended in case no more call appearances ## is available on the called/remote party. ## Value: 5 - 20 seconds; Default 10 seconds ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET FORBIDDEN_SESSION_REMOVAL_TIMER 5 ## ############# CONFERENCING SETTINGS (SIP ONLY) ############# ## ## CONFERENCE_FACTORY_URI specifies the URI for Avaya Aura Conferencing or Network Conferencing in 3PCC environments. ## Valid values contain zero or one URI, ## where a URI consists of a dial string followed by "@" followed by a domain, ## which must match the routing pattern configured in System Manager for Adhoc Conferencing. ## Depending on the dial plan, the dial string may need a prefix code, such as a 9 to get an outside line. ## The domain portion of the URI can be in the form of an IP address or an FQDN. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## Avaya Vantage Basic application SIP R1.1.0.1 and later (for IPO environment only). ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## 96x1 SIP R6.2.1 and later ## H1xx SIP R1.0 and later ## SET CONFERENCE_FACTORY_URI "93375000@avaya.com" ## ## CONFERENCE_ACCESS_NUMBER specifies the default Conference Access Number. The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_ACCESS_NUMBER "93375000" ## ## CONFERENCE_PORTAL_URI specifies the URI of the Conference Portal. The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_PORTAL_URI "https://10.10.10.10:8043/aacpa/" ## SET CONFERENCE_PORTAL_URI "https://conf.portal.com:8043/aacpa/" ## ## CONFERENCE_MODERATOR_CODE specifies the conference moderator code. ## The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_MODERATOR_CODE "20111" ## ## CONFERENCE_PARTICIPANT_CODE specifies the conference participant code. ## The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_PARTICIPANT_CODE "2011" ## ## CONFERENCE_VIRTUAL_ROOM specifies the Scopia Virtual Room ID for the virtual room owner. ## The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_VIRTUAL_ROOM "2011" ## ## CONFERENCE_FQDN_SIP_DIAL_LIST specifies a list of Scopia conferences bridges that can support SIP Enhanced Conference Experience. ## The default value is null (""). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET CONFERENCE_FQDN_SIP_DIAL_LIST "scopia.company.com,alphascopia.company.com,lab.company.com,scopia.partner.com" ## ## UCCPENABLED specifies whether to to enable or disable UCCP Conferencing protocol. ## Value Operation ## 0 UCCP Conferencing protocol is disabled. SIP CCMP is used for conferencing. ## 1 UCCP Conferencing protocol is enabled (default). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET UCCPENABLED 0 ## ## EVENT_NOTIFY_AVAYA_MAX_USERS specifies the maximum number of users to be included in ## an event notification message from CM/AST-II or Avaya Aura Conferencing R6.0 or later. ## Valid values are 0 through 1000; the default value is 20. ## It is used only for development and debugging purposes. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## SET EVENT_NOTIFY_AVAYA_MAX_USERS 10 ## ## SIGNAL_P_CONFERENCE_SIP_HEADER specifies whether P-Conference header shall be sent in SIP 200 OK message ## to the AAC conferencing server. ## Value Operation ## 0 P-Conference header will not be sent ## 1 P-Conference header will be sent (Default) ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET SIGNAL_P_CONFERENCE_SIP_HEADER 0 ## ################ PRESENCE SETTINGS (SIP ONLY) ############## ## ## ENABLE_PRESENCE specifies whether presence will be supported. ## Value Operation ## 0 Disabled ## 1 Enabled ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later, (default is 1); For IP office and 3PCC environments this parameter shall be set to 0 as presence is not supported. ## J169/J179 SIP R1.5.0 (default is 1) ## 96x1 SIP R6.2 and later (default is 1) ## 96x0 SIP R2.6.8 and later (default is 1) ## 96x0 SIP R2.6.6 and R2.6.7 (default is 0) ## H1xx SIP R1.0 and later (default is 1); For IP office environment this parameter shall be set to 0 as presence is not supported. ## SET ENABLE_PRESENCE 1 ## ## PRESENCE_SERVER specifies the address of the Presence server. ## Zero or one IP address in dotted decimal, ## optionally followed by a colon and a TCP port number. ## The default value is null (""). ## Note: Starting with 96x1 R6.5 SIP, if the phone is deployed with Aura Platform 6.2 FP4 and later, ## the value of this parameter is used from PPM and not from the settings file. ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## 96x0 SIP R2.6.6 and later ## H1xx SIP R1.0 and later; this parameter is not supported in IP Office environment as presence is not supported. ## SET PRESENCE_SERVER 192.168.0.5:8090 ## ## PRESENCE_ACL_CONFIRM specifies the handling of a Presence ACL update with pending watchers. ## Value Operation ## 0 Auto confirm - automatically send a PUBLISH to allow presence monitoring (Default) ## 1 Ignore - take no action ## 2 Prompt - the phone directly prompting the user to Allow or Deny the watcher’s request. ## This parameter is not supported in IP Office environment as presence is not supported. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later (values 0-1) ## 96x1 SIP R6.3 and later (values 0-1) ## H1xx SIP R1.0 and later (values 0-2) ## SET PRESENCE_ACL_CONFIRM 1 ## ## ALLOW_DND_SAC_LINK_CHANGE determines if the user will be allowed to change the DND and SAC button link. ## If the change is allowed, the menu to set the DND and SAC link is displayed. ## Value Operation ## 0 - do not allow a user to change default behavior (Default) ## 1 - allow a user change default behavior; parameter will be included in the "A" menu ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.4 and later. ## SET ALLOW_DND_SAC_LINK_CHANGE 1 ## ## DND_SAC_LINK specifies whether to activate the SendAllCall when user enables DoNotDisturb. ## Value Operation ## 0 Do not activate the SendAllCall when user enables DoNotDisturb (default) ## 1 Activate the SendAllCall when user enables DoNotDisturb ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); The value of this parameter is used if the ## ALLOW_DND_SAC_LINK_CHANGE is set to 0. ## 96x1 SIP R6.4 and later; The value of this parameter is used if the ALLOW_DND_SAC_LINK_CHANGE is set to 0. ## Avaya Equinox 3.1.2 and later ## SET DND_SAC_LINK 1 ## ## ENABLE_AUTOMATIC_ON_THE_PHONE_PRESENCE controls whether "on the phone" presence status ## is sent out automatically when user is on a call (or goes off-hook). ## Note that calls on bridged line appearances (that local user has not bridged to) ## do not affect the trigger of the "on the phone" presence update. ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## H1xx SIP R1.0 and later; this parameter is not supported in IP Office environment as presence is not supported. ## SET ENABLE_AUTOMATIC_ON_THE_PHONE_PRESENCE 0 ## ## AWAY_TIMER_VALUE controls the amount of time in minutes where there was no interaction ## with the device after which the device assumes that the user is away from the device. ## The range is 1-1500 minutes. The default value is 30 minutes. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later. ## 96x1 SIP R6.4 and later. ## H1xx SIP R1.0 and later; this parameter is not supported in IP Office environment as presence is not supported. ## SET AWAY_TIMER_VALUE 10 ## ## AWAY_TIMER controls whether the device report an ‘away’ state. ## When this parameter is set to 1, the device will automatically report an ‘away’ state. ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later. ## 96x1 SIP R6.4 and later. ## H1xx SIP R1.0 and later; this parameter is not supported in IP Office environment as presence is not supported. ## SET AWAY_TIMER 0 ## ## AUTO_AWAY_TIME specifies the idle time (in minutes) until presence automatically changes to 'away'. ## Value is normalized (downwards) to one of: [0, 5, 10, 15, 30, 60, 90, 120]. A value of 0 disables the feature. The default value is 30 minutes. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## SET AUTO_AWAY_TIME 10 ## ########### INSTANT MESSAGING SETTINGS (SIP ONLY) ########## ## ## INSTANT_MSG_ENABLED specifies whether Instant Messaging will be enabled or disabled. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## SET INSTANT_MSG_ENABLED 1 ## ########### MLPP SETTINGS (SIP ONLY) ########## ## ## ENABLE_MLPP specifies whether MLPP feature is enabled or not. ## Value Operation ## 0 Disable MLPP feature (default) ## 1 Enable MLPP feature ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET ENABLE_MLPP 1 ## ## MLPP_NET_DOMAIN specifies MLPP Network Domain ## Value Operation ## "" No MLPP Network Domain is configured (default) ## "dsn" DSN Network ## "uc" UC Network ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET MLPP_NET_DOMAIN "dsn" ## ## MLPP_MAX_PREC_LEVEL specifies maximum allowed precedence level for the user ## Value Operation ## 1 Maximum allowed precedence level is Routine (default) ## 2 Maximum allowed precedence level is Priority ## 3 Maximum allowed precedence level is Immediate ## 4 Maximum allowed precedence level is Flash ## 5 Maximum allowed precedence level is Flash Override ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET MLPP_MAX_PREC_LEVEL 2 ## ## ENABLE_PRECEDENCE_SOFTKEY indicates whether precedence soft key should be enabled on idle line appearances on Phone Screen. ## Value Operation ## 0 Disable precedence soft key ## 1 Enable precedence soft key (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET ENABLE_PRECEDENCE_SOFTKEY 0 ## ## DSCPAUD_FO specifies the DSCP value for Flash Override precedence/priority level voice call (0-63). Default value is 41. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DSCPAUD_FO 42 ## ## DSCPAUD_FL specifies the DSCP value for Flash precedence/priority level voice call (0-63). Default value is 43. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DSCPAUD_FL 44 ## ## DSCPAUD_IM specifies the DSCP value for Immediate precedence/priority level voice call (0-63). Default value is 45. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DSCPAUD_IM 43 ## ## DSCPAUD_PR specifies the DSCP value for Priority precedence/priority level voice call (0-63). Default value is 47. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DSCPAUD_PR 48 ## ## DSCPMGMT specifies the DSCP value for OA&M management packet (0-63). The default value is 16. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DSCPMGMT 15 ## ############### EXCHANGE SETTINGS (SIP ONLY) ############### ## ## EXCHANGE_SERVER_LIST specifies a list of one or more Exchange server IP addresses. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## H1xx SIP R1.0 and later ## SET EXCHANGE_SERVER_LIST exch1.myco.com,exch2.myco.com,exch3.myco.com ## ## EXCHANGE_SERVER_SECURE_MODE specifies whether to use HTTPS to contact Exchange servers. ## Value Operation ## 0 use HTTP ## 1 use HTTPS (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.3 and later. ## H1xx SIP R1.0 and later ## SET EXCHANGE_SERVER_SECURE_MODE 0 ## ## EXCHANGE_SERVER_MODE specifies the protocol(s) to be used to contact Exchange servers. ## Value Operation ## 1 use WebDAV ## 2 use Exchange Web Services (EWS) ## 3 try EWS first, if that fails, try WebDAV (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.3 and later. ## SET EXCHANGE_SERVER_MODE 1 ## ## PROVIDE_EXCHANGE_CONTACTS specifies whether menu item(s) for Exchange Contacts are displayed. ## Value Operation ## 0 Not displayed ## 1 Displayed (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.2 and later ## 96x0 SIP R2.0 through R2.4 only ## SET PROVIDE_EXCHANGE_CONTACTS 0 ## ## PROVIDE_EXCHANGE_CALENDAR specifies whether menu item(s) for Exchange Calendar are displayed. ## Value Operation ## 0 Not displayed ## 1 Displayed (default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## SET PROVIDE_EXCHANGE_CALENDAR 0 ## ## USE_EXCHANGE_CALENDAR specifies whether calendar data will be retrieved from Microsoft Exchange. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.0.x only (set only by user option in R6.2 and later) ## 96x0 SIP R2.5 and later ## SET USE_EXCHANGE_CALENDAR 1 ## ## EXCHANGE_USER_DOMAIN specifies the domain for the URL ## used to obtain Exchange contacts and calendar data. The EXCHANGE_USER_DOMAIN is used as part of the ## user authentication. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169/J179 only); Users can change this value in the "Options & Settings...". Refer to EXCHANGE_AUTH_USERNAME_FORMAT for how EXCHANGE_USER_DOMAIN is used. ## J169/J179 SIP R1.5.0; Users can change this value in the "Options & Settings...". Refer to EXCHANGE_AUTH_USERNAME_FORMAT for how EXCHANGE_USER_DOMAIN is used. ## 96x1 SIP R6.0 and later; Users can change this value in the "Options & Settings...". Refer to EXCHANGE_AUTH_USERNAME_FORMAT for how EXCHANGE_USER_DOMAIN is used. ## H1xx SIP R1.0 and later ## 96x0 SIP R2.5 and later ## SET EXCHANGE_USER_DOMAIN exchange.myco.com ## ## EXCHANGE_AUTH_USERNAME_FORMAT specifies the format of the username for user authentication. ## Value Operation ## 0 Office 2003/Office2016 username format - "EXCHANGE_USER_DOMAIN\Exchange Username" or "Exchange Username" if EXCHANGE_USER_DOMAIN is "". ## This is the default value. ## 1 Office 365 username format - "Exchange Username@EXCHANGE_USER_DOMAIN" or "Exchange Username" if EXCHANGE_USER_DOMAIN is "". ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later(J169/J179 only) ## 96x1 SIP R7.1.0.0 and later. ## SET EXCHANGE_AUTH_USERNAME_FORMAT 1 ## ## EXCHANGE_EMAIL_DOMAIN specifies the Exchange email domain. ## Exchange Username with EXCHANGE_EMAIL_DOMAIN defines the email address: Exchange Username@EXCHANGE_EMAIL_DOMAIN. ## This parameter cannot be changed by end users in the "Options & Settings..." menu. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.3 and later ## SET EXCHANGE_EMAIL_DOMAIN avaya.com ## ## ENABLE_EXCHANGE_REMINDER specifies whether or not Exchange reminders will be displayed. ## Value Operation ## 0 Not displayed (default) ## 1 Displayed ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## SET ENABLE_EXCHANGE_REMINDER 1 ## ## EXCHANGE_REMINDER_TIME specifies the number of minutes before an appointment ## at which a reminder will be displayed. ## Valid values are 0 through 60; the default value is 5. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## SET EXCHANGE_REMINDER_TIME 7 ## ## EXCHANGE_SNOOZE_TIME specifies the number of minutes after a reminder has been ## temporarily dismissed at which the reminder will be redisplayed. ## Valid values are 0 through 60; the default value is 5. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## SET EXCHANGE_SNOOZE_TIME 4 ## ## EXCHANGE_REMINDER_TONE specifies whether or not a tone will be generated ## the first time an Exchange reminder is displayed. ## Value Operation ## 0 Tone not generated ## 1 Tone generated (default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## SET EXCHANGE_REMINDER_TONE 0 ## ## EXCHANGE_NOTIFY_SUBSCRIPTION_PERIOD specifies the number of seconds between re-syncs ## with the Exchange server. ## This parameter is supported by: ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (J169/J179 only); valid values are 60 through 3600; the default value is 180. ## 96x1 SIP R6.2 and later; valid values are 60 through 3600; the default value is 180. ## 96x1 SIP R6.0.x; valid values are 0 through 3600; the default value is 180. ## 96x0 SIP R2.5 and later; valid values are 0 through 3600; the default value is 180. ## SET EXCHANGE_NOTIFY_SUBSCRIPTION_PERIOD 200 ## ############### CALENDAR SETTINGS ############### ## ## CALENDAR_PARTICIPANT_CODE_STRING specifies a list of semicolon separated values representing ## the phrase "participant code". The string to be recognized by the Calendar application before ## the participant code appears for click to dial functionality. ## The default value is: participant;participant code;participant-code;code;pc ## The parameter is used with AVaya Aura Conferencing. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CALENDAR_PARTICIPANT_CODE_STRING participant;participant code;participant-code;code ## ## CALENDAR_HOST_CODE_STRING specifies a list of semicolon separated values representing the phrase ## "host code". The string to be recognized by the Calendar application before the host code appears ## for click to dial functionality. ## The default value is: host;host code;host-code;hc ## The parameter is used with AVaya Aura Conferencing. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CALENDAR_HOST_CODE_STRING host;host code;host-code ## ## CALENDAR_MEETING_ID_STRING specifies a list of semicolon separated values representing the phrase ## "meetingid". The string to be recognized by the Calendar application before the meeting id appears ## for click to dial functionality. ## The default value is: meeting;meeting id;meeting-id;mid;id ## The parameter is used with Avaya Scopia. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CALENDAR_MEETING_ID_STRING meeting;meeting id;meeting-id;mid ## ## CALENDAR_MEETING_PIN_STRING specifies a list of semicolon separated values representing the phrase ## "meeting pin". The string to be recognized by the Calendar application before the meeting pin appears ## for click to dial functionality. ## The default value is: meeting pin;pin;meeting-pin ## The parameter is used with Avaya Scopia. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CALENDAR_MEETING_PIN_STRING meeting pin;pin ## ## CALENDAR_PHONE_NUM_MIN_DIGITS specifies the minimal number of digits required for the device to identify ## a number in the location or body of the message. ## The range is 4-21, where 4 is the default. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CALENDAR_PHONE_NUM_MIN_DIGITS 10 ## ################### OTHER SIP-ONLY SETTINGS ################ ## ## SPEAKERSTAT specifies the operation of the speakerphone. ## Value Operation ## 0 Speakerphone disabled ## 1 One-way speaker (also called "monitor") enabled ## 2 Full (two-way) speakerphone enabled (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; the parameter is not supported by J129. ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET SPEAKERSTAT 1 ## ## MUTE_ON_REMOTE_OFF_HOOK controls the speakerphone muting for a remote-initiated ## (a shared control or OOD-REFER) speakerphone off-hook. ## ## Valid values are 0 and 1 ## 0 - the speakerphone is Unmuted ## 1 - the speakerphone is Muted ## ## The default value is 1 (Muted) for 96x1 SIP R6.3 ## The default value is 0 (Unmuted) for 96x1 SIP R6.3.1 and later, J129 SIP R1.0.0.0 and later and H1xx SIP R1.0 and later ## ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.3 and later ## H1xx SIP R1.0 and later; R1.0.2 and later - this parameter is also supported in IPO environment where it is used ## to control auto answer calls whether they start muted (1) or not (0). ## ## The value of the parameter MUTE_ON_REMOTE_OFF_HOOK will be applied to the phone only when the phone is ## deployed with a CM 6.2.2 and earlier releases. ## ## If the phone is deployed with CM 6.3 or later, the MUTE_ON_REMOTE_OFF_HOOK variable is ignored and instead ## the feature is delivered via PPM by enabling the Turn on mute for remote off-hook attempt parameter in the station form ## via the Session Manager (System Manager) or Communication Manager (SAT) administrative interfaces. ## ## SET MUTE_ON_REMOTE_OFF_HOOK 0 ## ## AUTO_UNMUTE specifies whether the call will be unmuted on a transducer changing. This applies to all calls. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R7.1.0.0 and later ## ## SDPCAPNEG specifies whether or not SDP capability negotiation is enabled. ## Value Operation ## 0 SDP capability negotiation is disabled ## 1 SDP capability negotiation is enabled (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.6 and later ## SET SDPCAPNEG 0 ## ## ENFORCE_SIPS_URI specifies whether a SIPS URI must be used for SRTP. ## Value Operation ## 0 Not enforced ## 1 Enforced (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later; not applicable for 3PCC environment ## Avaya Equinox 3.4 and later ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.6 and later ## SET ENFORCE_SIPS_URI 1 ## ## 100REL_SUPPORT specifies whether the 100rel option tag is included in the SIP INVITE header field. ## Value Operation ## 0 The tag will not be included. ## 1 The tag will be included (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.6 and later ## SET 100REL_SUPPORT 1 ## ## DISPLAY_NAME_NUMBER specifies whether the name and/or number will be displayed for ## incoming calls, and if both are displayed, the order in which they are displayed. ## Value Operation ## 0: display calling party name only ## 1: display calling party name followed by calling party number ## 2: display calling party number only ## 3: display calling party number followed by calling party name ## This parameter is supported by: ## J169/J179 SIP R1.5.0; valid values 0 through 3; the default value is 0. ## 96x1 SIP R6.2 and later; valid values 0 through 3; the default value is 0. ## 96x1 SIP R6.0.x; valid values 0 through 1; the default value is 0. ## 96x0 SIP R2.6.5 and later; valid values 0 through 3; the default value is 0. ## 96x0 SIP R2.0 through R2.6.4; valid values 0 through 1; the default value is 0. ## SET DISPLAY_NAME_NUMBER 0 ## ## HOTLINE specifies zero or one hotline number. ## Valid values can contain up to 30 dialable characters (0-9, *, #). ## The default value is null (""). ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## SET HOTLINE "" ## ## PLAY_TONE_UNTIL_RTP specifies whether locally-generated ringback tone will stop ## as soon as SDP is received for an early media session, or whether it will continue ## until RTP is actually received from the far-end party. ## Value Operation ## 0 Stop ringback tone as soon as SDP is received ## 1 Continue ringback tone until RTP is received (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## H1xx SIP R1.0 and later ## SET PLAY_TONE_UNTIL_RTP 0 ## ## PLUS_ONE specifies whether pressing the 1 key during dialing will alternate between 1 and +. ## Value Operation ## 0 1 key only dials 1 (default). ## 1 1 key alternates between 1 and +. ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## SET PLUS_ONE 1 ## ## POUND_KEY_AS_CALL_TRIGGER specifies in case of off-hook dialing whether pressing "#" triggers the call or used as dialed digit. ## Value Operation ## 0 Pound/Hash key is used as a dialed digit . ## 1 Pound/Hash key triggers a call (default). ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.1.0.0 and later ## SET POUND_KEY_AS_CALL_TRIGGER 0 ## Note: For IP Office Environment, POUND_KEY_AS_CALL_TRIGGER shall be set to 0 for proper operation of the pound key. ## ## TEAM_BUTTON_RING_TYPE specifies the alerting pattern to use for team buttons. ## Valid values are 1 through 8, the default value is 1. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.2 and later ## SET TEAM_BUTTON_RING_TYPE 3 ## ## QTP_BUTTON_COMPRESS specifies the range of features which can be assigned to Quick Touch Panel on Phone Screen. ## Value Operation ## 0 buttons will be compressed and all features depicted in SMGR buttons 4 to 11 will be assigned to QTP without blanks. (default) ## 1 buttons will be compressed and blanks removed from the QTP panel. ## Features and Autodials configured on SMGR buttons 4 through 24 will show up on the QTP (up to a maximum of 8 buttons). ## Moreover Call Appearances, and Bridged Call Appearances will be excluded from QTP. ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R7.1.0.0 and later. ## SET QTP_BUTTON_COMPRESS 1 ## ## SECURECALL specifies whether an icon will be displayed when SRTP is being used. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## 96x1 SIP R6.2 up to R7.0.0 (excluded). The parameter has been obsoleted in 96x1 SIP R7.0.0. ## SET SECURECALL 1 ## ## LOCALLY_ENFORCE_PRIVACY_HEADER specifies whether the telephone will display ## "Restricted" (in the current language) instead of CallerId information when ## a Privacy header is received in a SIP INVITE message for an incoming call. ## Value Operation ## 0 Disabled (default): CallerID information will be displayed ## 1 Enabled: "Restricted" will be displayed ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## H1xx SIP R1.0 and later ## SET LOCALLY_ENFORCE_PRIVACY_HEADER 1 ## ## ENABLE_SIP_USER_ID controls the display of the user ID input field on the Login Screen ## Value Operation ## 0 SIP User ID field is not available to user during Login (default) ## 1 SIP User ID field is available to user during Login ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_SIP_USER_ID 1 ## ## ENABLE_STRICT_USER_VALIDATION specifies whether AOR received in 'Request-URI' of incoming call should be validated or not with 'contact' header published by phone in REGISTRATION. ## Value Operation ## 0 validation is not done (Default) ## 1 validation is done ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_STRICT_USER_VALIDATION 1 ## ## BRANDING_VOLUME specifies the volume level at which the Avaya audio brand is played. ## Value Operation ## 8 9db above nominal ## 7 6db above nominal ## 6 3db above nominal ## 5 nominal (default) ## 4 3db below nominal ## 3 6db below nominal ## 2 9db below nominal ## 1 12db below nominal ## 0 No Volume ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (Values 0-8) ## J169/J179 SIP R1.5.0 (Values 1-8) ## Avaya Vantage Devices SIP R1.0.0.0 and later (Values 1-8) ## J129 SIP R1.0.0.0 (or R1.1.0.0) (Values 1-8) ## 96x1 SIP R6.2 and later (Values 1-8) ## H1xx SIP R1.0 and later (Values 1-8) ## SET BRANDING_VOLUME 2 ## ## ENABLE_OOD_MSG_TLS_ONLY specifies whether an Out-Of-Dialog (OOD) REFER ## must be received over TLS transport to be accepted. ## Value Operation ## 0 No, TLS is not required ## 1 Yes, TLS is required (default) ## Note: A value of 0 is only intended for testing purposes. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## H1xx SIP R1.0 and later ## SET ENABLE_OOD_MSG_TLS_ONLY 1 ## ## PROVIDE_EDITED_DIALING specifies control for editied dialing for user. ## Value Operation ## 0 Dialing Options is not displayed. Edit dialing is disabled. ## The user cannot change edit dialing and the phone defaults to on-hook dialing. ## 1 Dialing Options is not displayed. On hook dialing is disabled. ## The user cannot change edit dialing and the phone defaults to edit dialing. ## 2 Dialing Options is displayed (default). ## The user can change edit dialing and the phone defaults to on-hook dialing. ## 3 Dialing Options is displayed. ## The user can change edit dialing and the phone defaults to edit dialing. ## This parameter is supported by: ## 96x1 SIP R6.0.x only ## 96x0 SIP R2.0 and later ## SET PROVIDE_EDITED_DIALING 2 ## ## VU_MODE specifies visiting user mode capabilities. ## Value Operation ## 0 No visiting user support (default). ## 1 User is prompted at registration time as to whether or not they are visiting. ## 2 Only visiting user registrations are allowed. ## This parameter is supported by: ## 96x1 SIP R6.0.x only ## 96x0 SIP R2.0 up to R2.6.13 (excluded). R2.6.13+ do not support SES. Visiting user feature is supported by SES only. ## SET VU_MODE 0 ## ## TEAM_BUTTON_REDIRECT_INDICATION controls if the redirection indication should be shown on ## a Team Button (on a monitoring station) in case it is not a redirect destination of the monitored station. ## Value Operation ## 0 - disabled; the redirect indication will be shown only on a monitoring station which is redirection destination (default). ## 1 - enabled; the redirection icon is displayed on all monitoring stations ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.4 and later ## SET TEAM_BUTTON_REDIRECT_INDICATION 1 ## ## ENABLE_BLIND_TRANSFER indicates whether enable blind transfer or not ## Value Operation ## 0 Disable blind transfer ## 1 Enable blind transfer (default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET ENABLE_BLIND_TRANSFER 0 ## ############ ACCESSIBILITY SETTINGS (SIP ONLY) ############# ## ## PROVIDE_KEY_REPEAT_DELAY specifies how long a navigation button must be held down ## before it begins to auto-repeat, and whether an option will be provided by which ## the user can change this value. ## Value Operation ## 0 Default (500ms) with user option (default) ## 1 Short (250ms) with user option ## 2 Long (1000ms) with user option ## 3 Very Long (2000ms) with user option ## 4 No Repeat with user option ## 5 Default (500ms) without user option ## 6 Short (250ms) without user option ## 7 Long (1000ms) without user option ## 8 Very Long (2000ms) without user option ## 9 No Repeat without user option ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## SET PROVIDE_KEY_REPEAT_DELAY 2 ## ################### HANDSET EQUALIZATION ################### ## ## ADMIN_HSEQUAL specifies handset audio equalization standards compliance ## Note that this value will only have an effect on a telephone if the handset equalization ## has not been set by the user or by the HSEQUAL local procedure for that telephone. ## Value Operation ## 1 Use handset equalization that is compliant with TIA 810/920 (default) ## 2 Use handset equalization that is compliant with FCC Part 68 HAC requirements ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2 and later ## 96x1 SIP R6.0.4 and later ## 96x0 H.323 R3.1.4 and later ## 96x0 SIP R2.6.7 and later ## SET ADMIN_HSEQUAL 2 ## ###################### HEADSET PROFILES #################### ## ## HEADSET_PROFILE_DEFAULT specifies the number of the default headset audio profile. ## Valid values are 1 through 20; the default value is 1. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x1 SIP R6.3 and later. ## H1xx SIP R1.0 and later ## SET HEADSET_PROFILE_DEFAULT 1 ## ## HEADSET_PROFILE_NAMES specifies an ordered list of names to be displayed for headset audio profile selection. ## See support.avaya.com for the list of headset profiles. ## The list can contain 0 to 255 UTF-8 characters; the default value is null (""). ## Names are separated by commas without any intervening spaces. ## Two commas in succession indicate a null name, ## which means that the default name should be displayed for the corresponding profile. ## Names may contain spaces, but if any do, the entire list must be quoted. ## There is no way to prevent a profile from being displayed. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x1 SIP R6.3 and later. ## H1xx SIP R1.0 and later ## SET HEADSET_PROFILE_NAMES "Avaya L100 Headset,Vendor A,Vendor B,Vendor C,,,,” ## ###################### HANDSET PROFILES #################### ## ## HANDSET_PROFILE_DEFAULT specifies the number of the default handset audio profile. ## Valid values are 1 through 20; the default value is 1. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET HANDSET_PROFILE_DEFAULT 1 ## ## HANDSET_PROFILE_NAMES specifies an ordered list of names to be displayed for handset audio profile selection. ## The list can contain 0 to 255 UTF-8 characters; the default value is null (""). ## Names are separated by commas without any intervening spaces. ## Two commas in succession indicate a null name, ## which means that the default name should be displayed for the corresponding profile. ## Names may contain spaces, but if any do, the entire list must be quoted. ## There is no way to prevent a profile from being displayed. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET HANDSET_PROFILE_NAMES "Acme Earwigs,,Spinco Ear Horns" ## ################ EMERGENCY TELEPHONE NUMBER ################ ## ## PHNEMERGNUM specifies an emergency telephone number to be dialed if the associated button is selected. ## Valid values may contain up to 30 dialable characters (0-9, *, #); the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.1 and later for IP Office Environment only ## 96x1 H.323 R6.0 and later; the parameter is supported when the phone is registered to Avaya Communication Manager only. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.5 and later; the parameter is supported when the phone is registered to Avaya Communication Manager only. ## 96x0 SIP R2.0 and later ## 4630 H.323 R1.0 and later ## SET PHNEMERGNUM 9911 ## ## PHNMOREEMERGNUMS specifies list of comma separated emergency numbers ## Valid values may contain up to 30 dialable characters (0-9, *, #); the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.1 and later for IP Office Environment only ## H1xx SIP R1.0.2 and later ## SET PHNMOREEMERGNUMS "911,109,115" ## ############ EMERGENCY NUMBER SOFTKEY (SIP ONLY) ########### ## ## ENABLE_SHOW_EMERG_SK specifies whether an emergency softkey, ## with or without a confirmation screen, will be displayed when the phone is registered. ## All emergency numbers will always be supported. ## Value Operation ## 0 An emergency softkey will not be displayed. ## 1 An emergency softkey will be displayed, without a confirmation screen. ## 2 An emergency softkey will be displayed, with a confirmation screen (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## SET ENABLE_SHOW_EMERG_SK 1 ## ## ENABLE_SHOW_EMERG_SK_UNREG specifies whether an emergency softkey, ## with or without a confirmation screen, will be displayed when the phone is not registered. ## All emergency numbers will always be supported. ## Value Operation ## 0 An emergency softkey will not be displayed. ## 1 An emergency softkey will be displayed, without a confirmation screen. ## 2 An emergency softkey will be displayed, with a confirmation screen (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## SET ENABLE_SHOW_EMERG_SK_UNREG 1 ## ############## APPLICATION ACCESS SETTINGS ############### ## ## APPSTAT restricts access to certain applications. ## Value Operation ## 0 Call Log and Redial are suppressed and changes to Speed Dial/Contacts are not allowed ## 1 Call Log, Redial and Speed Dial/Contacts work without restrictions (default) ## 2 Call Log is suppressed, the Last-6-numbers Redial option is suppressed, ## and changes to Speed Dial/Contacts are not allowed ## 3 Changes to Speed Dial/Contacts are not allowed; other applications work without restrictions ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R1.0 and later ## B189 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET APPSTAT 1 ## ################## OPTION ACCESS SETTINGS ################## ## ## OPSTAT restricts access to certain user options. ## Value Operation ## 000 user options are not accessible ## 001 user can only access the Log-Off Option. ## 010 user can only access view-only options, they cannot change any settings ## 011 user can only access view-only options and the Log-Off Option ## 100 user can access all options except the view-only options and the Log-Off option ## 101 user can access all options except the view-only options ## 110 user can access all the options except the Log-Off option ## 111 user can invoke any or all of the user options (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R1.0 and later ## B189 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET OPSTAT 101 ## ## OPSTAT2 specifies whether customized labels from a backup file will be used ## even if the first digit of the value of OPSTAT is "0". ## Value Operation ## 0 Customized labels from a backup file will not be used (default) ## 1 Customized labels from a backup file will be used ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## B189 H.323 R1.0 and later ## SET OPSTAT2 1 ## ## SYSAUDIOPATH specifies whether the Audio Path option will be displayed ## for user selection or whether the audio path used for a server-initiated ## off-hook command will be determined by this parameter. ## Value Operation ## 0 The Audio Path option is displayed for user selection (default) ## 1 The audio path is set to Speaker ## 2 The audio path is set to Headset ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SYSAUDIOPATH 1 ## ## LOCKED_PREFERENCES specifies list of parameters configured in the Avaya Equinox Application under "User preferences" menus ## which shall be blocked for user configuration. I.e. users can only view their values, but not change them. The default value is "". ## In case of a parameter mentioned in the OBSCURE_PREFERENCES, then the administrator only will be able to configure it using this file. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET LOCKED_PREFERENCES "NAME_SORT_ORDER,NAME_DISPLAY_ORDER" ## ## OBSCURE_PREFERENCES specifies list of parameters configured in the Avaya Equinox Application under "User preferences" menus ## which shall be hidden for users. I.e. users cannot see them. The default value is "". ## In case of a parameter mentioned in the OBSCURE_PREFERENCES, then the administrator only will be able to configure it using this file. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET OBSCURE_PREFERENCES "NAME_SORT_ORDER,NAME_DISPLAY_ORDER" ## ################ PHONE SETTINGS (H.323 ONLY) ############### ## ## PHNSCRALL specifies whether separate screens will be displayed for call appearances and features. ## Value Operation ## 0 Separate screens will be displayed for call appearances and features (default) ## 1 A consolidated screen will be displayed that includes call appearances and features ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET PHNSCRALL 1 ## ## EOEDITDIAL specifies whether a # character will be inserted at the end of Edit Dialing strings. ## Value Operation ## 0 # will not be inserted ## 1 # will be inserted (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x0 H.323 R3.2 and later ## SET EOEDITDIAL 0 ## ## FBONCASCREEN specifies whether features will be displayed on the same screen as call appearances ## when the value of PHNSCRALL is 0. ## Value Operation ## 0 Features will not be displayed on the same screen as call appearances (default) ## 1 As many features as will fit will be displayed on the same screen as call appearances, ## in addition to being displayed on a separate feature screen. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later (9608, 9608G and 9611G models only) ## 96x0 H.323 R3.0 and later (9630 and 9640 models only) ## SET FBONCASCREEN 1 ## ## PHNSCRIDLE specifies the content displayed in Main Display Area when the phone is in Idle state. ## Value Operation ## 0 Dial Pad Window is displayed (Default) ## 1 Call Appearance Window is displayed. ## This parameter is supported by: ## B189 H.323 R1.0 and later ## SET PHNSCRIDLE 1 ## ## PHNSCRCOLUMNS specifies whether the Phone Screen is presented with ## one (full-width) or two (each half-width) columns. ## This parameter is relevant only for 9608, 9608G and 9611G phones only. ## Note: The field "Phone Screen Width" in HOME-> Options & Settings -> Screen & Sound Options menu allows ## also users to change the way phone screen is presented as described above. PHNSCRCOLUMNS will be enforced only if ## user did not change at all the field "Phone Screen Width" value. Please note that user changes are stored in backup/restore ## file as "Phone Screen Width" field (if BRURI has a valid value) which means that if the restored file include "Phone Screen Width" parameter then it ## will take precedence over PHNSCRCOLUMNS. If BRURI is not valid, but user still change the content of "Phone Screen Width" field, then ## user value will take precedence over PHNSCRCOLUMNS (The only way to clear user configuration in this case is by doing "CLEAR" operation in CRAFT menu). ## Value Operation ## 0 call appearance and feature button occupies the entire width of the Line (default) ## 1 call appearance and feature button occupies half the width of the Line ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later SET PHNSCRCOLUMNS 1 ## ## CADISPMODE specifies whether to keep the display of the call appearance label in call state idle mode as it is ## without dependency on call state (ringing, dialing, etc) and whether to add prefix or suffix in order ## to identify the bridge/line number. The parameter is supported with Avaya Communication Manager only. ## Value Operation ## 0 Labels are changed according to call state where Avaya Communication Manager provides the labels. ## This is the behavior in pre 6.6 releases. This is the default value. ## 1 The idle call label is presented independent on call states. In addition, "a." through "z." lowercase (and then "A."-"Z.") are added ## as prefix in full width screen or as a suffix on the right column and a prefix on the left column in half width screen. ## "a." through "z." are added to bridged and line appearances according to the bridged/line button order. ## 2 The idle call label is presented independent on call states as in 1, but without addition of "a." through "z." lowercase (and then "A."-"Z.") strings. ## If personalized label is configured for line/bridged appearance then it will be used instead of the idle call label assigned by Avaya Communication Manager. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## SET CADISPMODE 1 ## ## CALLAPPRSELMODE controls highlight of call appearance when there is incoming call. ## Value Operation ## 0 When there is incoming call, the call appearance of incoming call is highlighted and applicable softkeys ## are presented for incoming call ("Answer", "Ignore" if no other call exists or "Ans Hold", "Ans Drop", ## "Ignore" if another call exists). This is the behavior in pre 6.6 releases. This is the default value. ## 1 When there is incoming call, the highlight remains on the active/hold call appearance and therefore ## presenting the softkeys for the active/hold call (and not for the incoming call). ## CALLAPPRSELMODE is supported when the phone is registered to Avaya Communication Manager only. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## SET CALLAPPRSELMODE 1 ## ## HIDEDTMFDIGITS - defines if the DTMF digits will be displayed when they are entered. ## Value Operation ## 0 DTMF digits will be displayed when entered (default) ## 1 DTMF digits will not be displayed when entered; they will be replaced by '*' ## This parameter is supported by: ## B189 H.323 R1.0 SP1 and later ## SET HIDEDTMFDIGITS 1 ## ##################### CALL LOG SETTINGS #################### ## ## CLDISPCONTENT specifies whether the name, the number, or both will be displayed for Call Log entries. ## Value Operation ## 0 Both the name and the number will be displayed ## 1 Only the name will be displayed (default) ## 2 Only the number will be displayed ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 (only supports values of 0 or 1) ## 96x1 H.323 R6.0 and up to R6.6.3 (excluded) (only supports values of 0 or 1) ## 96x1 H.323 R6.6.3 and later ## 96x1 SIP R6.0 and later (only supports values of 0 or 1) ## 96x0 H.323 R3.2 and later ## SET CLDISPCONTENT 0 ## ## LOG_DIALED_DIGITS specifies if the call log will contain digits dialed by a user or ## information about a remote party in case where the user dialed a FAC code. ## The FAC code is identified by * or # entered as a first character. ## ## Value Operation ## 0 Allow dialed FAC code to be replaced with a remote party number in the call History ## 1 Dialed digits are logged in call History exactly as they were entered by the user (default). ## ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.5 and later ## ## SET LOG_DIALED_DIGITS 0 ## ############## CALL LOG SETTINGS (H.323 ONLY) ############## ## ## CLDELCALLBK specifies whether a Call Log entry will be deleted when a callback is initiated ## by pressing the Call softkey from the entry's Details screen. ## Value Operation ## 0 Entries will not be deleted when a callback in initiated (default) ## 1 Entries will be deleted when a callback in initiated ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## SET CLDELCALLBK 1 ## ## LOGMISSEDONCE specifies whether Call Log will display more than one ## missed Call Log entry from the same caller. ## Value Operation ## 0 Multiple Call Log entries will be displayed per caller (default) ## 1 Only one missed Call Log entry will be displayed per caller ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## SET LOGMISSEDONCE 1 ## ## LOGUNSEEN specifies whether incoming calls that did not cause alerting will be logged ## as missed calls (e.g., calls that are forwarded because the phone is busy). ## Value Operation ## 0 Unseen calls will not be logged (default) ## 1 Unseen calls will be logged ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## SET LOGUNSEEN 1 ## ## LOGBACKUP specifies whether Call Log entries will be backed up to, ## and restored from, the backup/restore file. ## Value Operation ## 0 Call Log entries will not be backed up and restored ## 1 Call Log entries will be backed up and restored (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## SET LOGBACKUP 0 ## ## CLBACKUPTIMESTAT specifies whether Call Log entries will be backed up only after ## a minimum interval as specified by the value of CLBACKUPTIME. ## Note that this parameter only has an effect if the value of LOGBACKUP is 1. ## Value Operation ## 0 Call Log entries will be backed up as they are created (default) ## 1 Call Log entries will be backed up after the interval specified by CLBACKUPTIME ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x0 H.323 R3.1.3 and later ## 96x1 H.323 R6.6.3 and later ## SET CLBACKUPTIMESTAT 1 ## ## CLBACKUPTIME specifies the minimum interval, in minutes, between backups of the Call Log, ## if the values of LOGBACKUP and CLBACKUPTIMESTAT are both 1. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later; Valid values are 1 through 60; the default value is 15. ## 96x0 H.323 R3.1.3 and later; Valid values are 10 through 60; the default value is 15. ## 96x1 H.323 R6.6.3 and later; Valid values are 1 through 60; the default value is 15. ## SET CLBACKUPTIME 20 ## ## CALL_LOG_JOURNAL specifies whether retrieving calls while offline feature is supported. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## SET CALL_LOG_JOURNAL 1 ## ############## RING TONE SETTING (H.323 ONLY) ############ ## ## DEFAULTRING specifies the default ring tone. ## Valid values are 1 through 14; the default value is 9. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2 and later ## SET DEFAULTRING 12 ## ################ TIMER SETTING (H.323 ONLY) ############## ## ## TIMERSTAT specifies whether Timer On and Timer Off softkeys will be presented to the user. ## Value Operation ## 0 Timer On and Timer Off softkeys will not be presented to the user (default) ## 1 Timer On and Timer Off softkeys will be presented to the user ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## B189 H.323 R1.0 and later ## SET TIMERSTAT 1 ## ####################### USB SETTINGS ####################### ## ## Please note that USB mass storage device is supported when the phone is registered ## to Avaya Communication Manager only. ## ## USBPOWER controls when power is provided to the USB interface. ## Value Operation ## 0 Turn off USB power regardless of power source. ## 1 Turn on USB power only if Aux powered. ## 2 Turn on USB power regardless of power source (default). ## 3 Turn on USB power if Aux powered or PoE Class 3 power. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 SIP 7.1.0.0 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R2.0 and later ## SET USBPOWER 0 ## ## USBLOGINSTAT specifies whether the USB Login/Logout feature is enabled ## Value Operation ## 0 USB Login/Logout feature is disabled. ## 1 USB Login/Logout feature is enabled (default). ## This parameter is supported by: ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.0 and later ## SET USBLOGINSTAT 1 ## ## ENABLE_USB_GENERAL_PURPOSE controls whether the USB general purpose port is enabled. ## Value Operation ## 0 USB port is disabled. ## 1 USB port is enabled (default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET ENABLE_USB_GENERAL_PURPOSE 0 ## ########### BLUETOOTH SETTINGS ############## ## ## BLUETOOTHSTAT specifies whether the user is given an option to enable Bluetooth. ## Value Operation ## 0 Bluetooth is disabled and the user is not given an option to enable it ## 1 The user is given an option to enable Bluetooth (default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; Up to R1.0.0.0 build 2304 (excluded), default is 0 and shall remain 0 (disabled) as BLUETOOTH is not officially supported. ## from build 2304 and later Bluetooth is officially supported as described in the parameter description. ## 96x1 SIP 7.0.0 and later ## 96x1 H.323 R6.2 and later ## H1xx SIP R1.0 and later SET BLUETOOTHSTAT 0 ## ## BLUETOOTH_FEATURES_SHARED_VIA_STAT which specifies whether "Shared via Bluetooth" option will be offered to the users or not. ## Value Operation ## 0 "Shared via Bluetooth" option is disabled (default) ## 1 "Shared via Bluetooth" option is enabled ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later ## SET BLUETOOTH_FEATURES_SHARED_VIA_STAT 1 ## ########### DECT SETTINGS ############## ## ## DECTSTAT specifies whether the user is given an option to enable DECT. DECT is used ## for cordless handset. ## Value Operation ## 0 DECT is disabled and the user is not given an option to enable it ## 1 The user is given an option to enable DECT (default) ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET DECTSTAT 0 ## ############################################################ ## ## WI-FI SETTINGS ## ############################################################ ## ########## NETWORK MODE OF OPERATION ########## ## ## WIFISTAT specifies whether the user is given an option to enable Wi-Fi. ## Value Operation ## 0 Wi-Fi is disabled and the user is not given an option to enable it; the phone will only use the Ethernet interface. ## 1 The user is given an option to enable Wi-Fi; the phone will connect to Ethernet (Default), unless the UI is used to manually switch to Wi-Fi. ## 2 Wi-Fi is the preferred interface, but manual override to a different SSID or to Ethernet is allowed; the phone will connect to WLAN_ESSID (i.e., the pre-configured Wi-Fi ## network) unless the phone UI is used to manually switch to another SSID or to Ethernet. Associated pre-configured Wi-Fi network security parameters must also be specified. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; values 0-2 are supported. ## Only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT). If the phone does not support Wi-Fi/BT, WIFISTAT will be internally set ## to 0, regardless of the value received from 46xxsettings.txt. ## The Administrator should always test a new settings file configuration on a single phone before committing it to several phones, as a configuration error (such as ## specifying an incorrect WLAN_ESSID or Wi-Fi security settings) will cause phones to become disconnected from the network, necessitating manual correction on each ## phone's local User Interface, as the phone will not be reachable via any other means. ## If the phone is using the pre-configured network (i.e., Ethernet or a specific Wi-Fi WLAN_ESSID), and then the phone UI is used to manually switch to a different ## network, the phone will enter Manual Network Configuration Mode, which will cause the phone to continue to connect to the manually-configured network on subsequent ## reboots, regardless of the pre-configured network specified by WIFISTAT and any associated parameters. ## There are 2 ways to return the phone to Automatic Network Configuration mode (i.e., to comply again with WIFISTAT, and if WIFISTAT=2, also WLAN_ESSID and ## associated pre-configured Wi-Fi network security parameters): ## - Use the phone's UI to explicitly toggle "Network config" from "Manual" to "Auto". ## - Change WIFISTAT to 0 and reboot the phone, which will force the phone to use Ethernet, after which, WIFISTAT can be changed to the desired value and the phone ## rebooted again. ## Avaya Vantage Devices SIP R1.0.0.0 and later; values 0-1 are supported. ## H1xx SIP R1.0 and later; values 0-1 are supported. ## SET WIFISTAT 0 ## ## WIFIAPSTAT specifies whether the user is given an option to enable Wi-Fi hotspot. ## Value Operation ## 0 Wi-Fi hotspot is disabled and the user is not given an option to enable it (default) ## 1 The user is given an option to enable Wi-Fi hotspot ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET WIFIAPSTAT 1 ## ## WIFI_CON_STATUS_ON_LOGOUT specifies whether ALL wireless connections will be forgotten (including static networks) when the device is logout. ## Value Operation ## 0 ALL Wi-Fi connections are forgotten (including static networks and all authentication options (802.1x, WEP/WPA)) when the device moves to logout state ## 1 ALL Wi-Fi connections are preserved when the device moves to logout state (and in particular, the active Wi-Fi connection remains as it is)(default) ## Note: when WIFI_CON_STATUS_ON_LOGOUT is set to 1, then the Wi-Fi credentials are shared across all users. When WIFI_CON_STATUS_ON_LOGOUT is set to 0 (and the network mode is Wi-Fi), ## after each logout, then the new/same user is required to enter Wi-Fi credentials before being able to login. When there is no Wi-Fi connectivity, emergency calls cannot be established. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## SET WIFI_CON_STATUS_ON_LOGOUT 0 ## ## WLAN_MAX_AUTH_FAIL_RETRIES specifies how many times the phone will retry a secure connection upon receiving (possibly successive) auth failures. ## Value range is 0-4. The default is 3. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET WLAN_MAX_AUTH_FAIL_RETRIES 2 ## ########## NETWORK CONFIGURATION USER PRIVILEGE ########## ## ## ENABLE_NETWORK_CONFIG_BY_USER specifies whether network configuration can be modified by the end user, either via the Settings menu, or when there is a network issue that ## could be remedied by the user. ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET ENABLE_NETWORK_CONFIG_BY_USER 0 ## ########## WI-FI REGULATORY DOMAIN SETTINGS ########## ## ## WLAN_COUNTRY specifies the 2-character ISO 3166 Alpha-2 Country code representing the Wi-Fi regulatory domain. ## The default value is "US". ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_COUNTRY CA ## ## WLAN_ENABLE_80211D specifies whether 802.11d is used or not. When enabled, the Wi-Fi regulatory domain will be used according to the 802.11d Country IE provided by the ## connected Wi-Fi Access Point. When disabled, the Wi-Fi regulatory domain will be used according to WLAN_COUNTRY. ## Note: The use of 802.11d is banned in the United States, so this parameter must NOT be set to 1 in this regulatory domain. ## Value Operation ## 0 Disabled (Default) ## 1 Enabled ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_ENABLE_80211D 1 ## ########## PRE-CONFIGURED WI-FI NETWORK ########## ## ## WLAN_ESSID specifies the SSID string of the pre-configured Wi-Fi network. ## The value can contain 1 to 32 characters; the default value is null (""). ## Valid characters are: ## A-Z, a-z, 0-9, and the following: *.-!$%&'()+,:;/\=@~# ## The space character, ASCII 0x20, is NOT supported. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_ESSID mywlanSSID ## ## WLAN_SECURITY specifies the pre-configured Wi-Fi network Security Method. ## Value Operation ## none No security (Default) ## wep WEP security ## wpa2psk WPA/WPA2 PSK (pre-shared key) security ## wpa2e WPA2 Enterprise security (802.1x authentication) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_SECURITY wpa2e ## ##### Pre-configured Wi-Fi network WEP security settings ##### ## ## WEP_DEFAULT_KEY specifies the pre-configured Wi-Fi network index of the WEP default key. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wep. ## The range of valid values is 1-4; the default value is 1. ## Only Shared Key authentication is supported. Open authentication is NOT supported. ## Some Wi-Fi Routers can only be configured with 1 WEP key, in which case ONLY WEP_KEY1 should be set. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WEP_DEFAULT_KEY 2 ## ## WEP_KEY_LEN specifies the pre-configured Wi-Fi network WEP key length. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wep. ## Value Operation ## 64bit WEP keys of 64 bits ## 128bit WEP keys of 128 bits (default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WEP_KEY_LEN 64bit ## ## WEP_KEY1/2/3/4 specifies the pre-configured Wi-Fi network WEP Keys 1 to 4. ## These parameters are only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wep. ## The value can contain 10 (for 64-bit WEP) or 26 (for 128-bit WEP) ASCII-Hex digits; the default value is null (""). ## Valid characters are: ## 0-9, A-F ## These parameters are supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WEP_KEY1 0123456789ABCDEF0123456789 ## SET WEP_KEY2 123456789ABCDEF01234567890 ## SET WEP_KEY3 23456789ABCDEF012345678901 ## SET WEP_KEY4 3456789ABCDEF0123456789012 ## ##### Pre-configured Wi-Fi network WPA/WPA2 PSK or 802.1X EAP security settings ##### ## ## WLAN_PASSWORD specifies the pre-configured Wi-Fi network password. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and: ## - WLAN_SECURITY is wpa2psk ## or ## - WLAN_SECURITY is wpa2e and WLAN_WPA2E_EAP_METHOD is PEAP and WLAN_WPA2E_EAP_PHASE2 is MSCHAPV2 ## If WLAN_SECURITY is wpa2psk, the value can contain 8 to 63 characters. ## If WLAN_SECURITY is wpa2e, the value can contain 1 to 32 characters. ## The default value is null (""). ## Valid characters are: ## A-Z, a-z, 0-9, and the following: *.-!$%&'()+,:;/\=@~# ## The space character, ASCII 0x20, is NOT supported. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_PASSWORD Avaya123 ## ## WLAN_WPA2E_EAP_METHOD specifies the pre-configured Wi-Fi network 802.1x EAP method. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e. ## Value Operation ## PEAP Connect using PEAP (Default) ## TLS Connect using TLS ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_WPA2E_EAP_METHOD TLS ## ## WLAN_WPA2E_EAP_PHASE2 is the pre-configured Wi-Fi network 802.1x phase 2 Method. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e and WLAN_WPA2E_EAP_METHOD is PEAP. ## Value Operation ## none No phase 2 authentication (Default, but not currently supported) ## MSCHAPV2 As of J100 2.0, MUST be set to this value for forward compatibility ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_WPA2E_EAP_PHASE2 MSCHAPV2 ## ## WLAN_WPA2E_IDENTITY specifies the pre-configured Wi-Fi network 802.1x identity. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e and: ## - WLAN_WPA2E_EAP_METHOD is PEAP and WLAN_WPA2E_EAP_PHASE2 is MSCHAPV2 ## or ## - WLAN_WPA2E_EAP_METHOD is TLS ## The value can contain 1 to 32 characters; the default value is null (""). ## Valid characters are: ## A-Z, a-z, 0-9, and the following: *.-!$%&'()+,:;/\=@~# ## The space character, ASCII 0x20, is NOT supported. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_WPA2E_IDENTITY User123 ## ## WLAN_WPA2E_ANONYMOUS_IDENTITY specifies the pre-configured Wi-Fi network 802.1x anonymous identity. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e and ## WLAN_WPA2E_EAP_METHOD is PEAP and WLAN_WPA2E_EAP_PHASE2 is MSCHAPV2. ## The value can contain 1 to 32 characters; the default value is null (""). ## Valid characters are: ## A-Z, a-z, 0-9, and the following: *.-!$%&'()+,:;/\=@~# ## The space character, ASCII 0x20, is NOT supported. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_WPA2E_ANONYMOUS_IDENTITY foo@example ## ########## WLAN LAYER 2 QOS SETTINGS ######## ## ## WLAN_L2QUAD specifies the layer 2 priority value for audio frames generated by the telephone when Wi-Fi interface is used. ## Valid values are 0 through 7; the default value is 6. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_L2QUAD 1 ## ## WLAN_L2QSIG specifies the layer 2 priority value for signaling frames generated by the telephone when Wi-Fi interface is used. ## Valid values are 0 through 7; the default value is 3. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_L2QSIG 1 ## ########## WLAN LAYER 3 QOS SETTINGS ######## ## ## WLAN_DSCPAUD specifies the layer 3 Differentiated Services (DiffServ) Code Point for audio frames generated by the telephone when Wi-Fi interface is used. ## Valid values are 0 through 63; the default value is 46. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_DSCPAUD 1 ## ## WLAN_DSCPSIG specifies the layer 3 Differentiated Services (DiffServ) Code Point for signaling frames generated by the telephone when Wi-Fi interface is used. ## Valid values are 0 through 63; the default value is 34. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT) ## SET WLAN_DSCPSIG 1 ## ########### Headset Signaling ############## ## ## HEADSETBIDIR specifies whether bidirectional signaling ## on the headset interface will be enabled or disabled. ## This parameter shall only be used in case of using wireless headset and the base station is connected to headset jack of the phone. In all other cases (such as use of wired headset), ## the parameter shall remain with the default value "Disabled". ## Note: The user has an option to change the value of this parameter through UI (96x1 H.323 - "Headset Signaling..." field in HOME-> Options & Settings -> Call Settings", ## H175 SIP - "Headset Signaling..." field in Settings application -> Call Settings"). This parameter is backup/restore to file server if BRURI is valid (96x1 H.323) or PPM (H175 SIP). ## In case of 96x1 H.323 the parameter precedence is according to the last source (which means backup/restore value has higher precedence compare to the 46xxsettings.txt file). If BRURI ## is invalid then the value from the settings file will be used. ## In case of H175 SIP, the value configured in the 46xxsettings.txt file will be used as initial configuration in case no such parameter is stored in PPM in Aura ## Environment. ## Value Operation ## 0 Disabled (default) ## 1 Switchhook and alerting signaling are both enabled ## 2 Only switchhook signaling is enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later (values 0-2) ## Avaya Vantage Devices SIP R1.0.0.0 and later - only value 2 is supported. ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.3 and later (values 0-2) ## 96x1 H.323 R6.2.1 and later (values 0-1) ## Note that 96x1 H.323 R6.2 only supported signaling for alerting. ## SET HEADSETBIDIR 1 ## ########## AUTO-ANSWER SETTINGS (H.323 ONLY) ############# ## ## AUTOANSSTAT specifies the operation of the local auto-answer capability. ## Value Operation ## 0 Auto-answer is disabled (default). ## 1 Auto-answer is always enabled. ## 2 Auto-answer is enabled if the incoming call is on a primary call appearance. ## 3 Auto-answer is always enabled if the user is logged into a call center. ## 4 Auto-answer is enabled if the user is logged into a call center ## and if the incoming call is on a primary call appearance. ## Note: Auto-answer also depends on the value of AUTOANSSTRING (see below). ## Also, if the call server is administered to provide auto-answer capability, ## the call server administration will take precedence over AUTOANSSTAT. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2.2 and later ## SET AUTOANSSTAT 1 ## ## AUTOANSSTRING specifies a substring that must appear in the call-associated ## display message for an incoming call if that call is to be auto-answered. ## If the value of AUTOANSSTRING is null, no substring is required. ## The value can contain 0 to 15 characters; the default value is null (""). ## Note: Auto-answer also depends on the value of AUTOANSSTAT (see above). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2.2 and later ## SET AUTOANSSTRING ## ## AUTOANSALERT specifies whether the telephone will audibly alert for auto-answered calls. ## Value Operation ## 0 Auto-answered calls will not alert (default). ## 1 Auto-answered calls will alert. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2.2 and later ## SET AUTOANSALERT 1 ## ################### CALL CENTER SETTINGS ################# ## ## HEADSYS specifies whether the telephone will go on-hook if the headset is active ## when a Disconnect message is received. ## Value Operation ## 0 The telephone will go on-hook if a Disconnect message is received when the headset is active ## 1 Disconnect messages are ignored when the headset is active ## Note: a value of 2 has the same effect as a value of 0, and ## a value of 3 has the same effect as a value of 1. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later (the default value is 0 unless the value ## of CALLCTRSTAT is set to 1, in which case the default value is 1) ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later (the default value is 0) ## 96x1 H.323 R6.2.1 and later (the default value is 0 unless the value ## of CALLCTRSTAT is set to 1, in which case the default value is 1) ## 96x1 H.323 R6.1 and R6.2 ignore this parameter, and will ignore Disconnect messages ## if the user is logged in as a call center agent. If the user is not logged in ## as a call center agent, the telephone will go on-hook if a Disconnect message ## is received when the headset is active. ## 96x1 H.323 releases prior to R6.1 (the default value is 1) ## 96x1 SIP R6.4 and later (the default value is 0) ## 96x1 SIP R6.0 and later up to R6.4 (not included) (the default value is 1) ## 96x0 H.323 R1.2 and later (the default value is 1) ## 96x0 SIP R1.0 and later (the default value is 1) ## 16xx H.323 R1.3 and later (the default value is 1) ## SET HEADSYS 0 ## ########## CALL CENTER SETTINGS (96x1/J100 SIP ONLY) ############ ## ## SKILLSCREENTIME specifies the duration, in seconds, that the Skills screen will be displayed. ## Valid values are 0 through 60; the default value is 5. ## A value of 0 means that the Skills screen will not be removed automatically when the agent logs in. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.2 and later ## SET SKILLSCREENTIME 5 ## ## UUIDISPLAYTIME specifies the duration, in seconds, that the UUI Information screen will be displayed. ## Valid values are 5 through 60; the default value is 10. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.2 and later ## SET UUIDISPLAYTIME 10 ## ## ENTRYNAME specifies whether the Calling Party Name or the VDN/Skill Name will be used in History entries. ## Value Operation ## 0 Calling Party Name will be used (default) ## 1 VDN/Skill Name will be used ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## SET ENTRYNAME 1 ## ## BUTTON_MAPPINGS specifies a list of Button=Status pairs that change the operation ## of some of the buttons on the telephone. ## Button=Status pairs are separated by commas without any intervening spaces. ## Valid Button values are "Forward", "Speaker", "Hookswitch", and "Headset". ## Valid Status values are "na" and "cc-release". ## Value Operation ## na The corresponding button will be disabled. ## cc-release The button will invoke the cc-release feature. ## If the value of BUTTON_MAPPINGS is null (the default), all buttons will operate normally. ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## SET BUTTON_MAPPINGS Forward=na,Speaker=cc-release,Hookswitch=na,Headset=na ## ## CC_INFO_TIMER specifies the duration, in hours, of the subscription to the SIP CC-Info event package. ## Valid values are 1 through 24; the default value is 8. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.2 and later ## SET CC_INFO_TIMER 8 ## ########## CALL CENTER SETTINGS (96x1 H.323 ONLY) ########## ## ## CALLCTRSTAT specifies whether Call Center features will be enabled or disabled. ## Value Operation ## 0 Call Center features will be disabled (default) ## 1 Call Center features will be enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET CALLCTRSTAT 1 ## ## OPSTATCC specifies whether Call Center options such as Greetings will be presented ## to the user even if the value of OPSTAT is set to disable user options. ## Note that the value of CALLCTRSTAT must be 1 for OPSTATCC to be used. ## Value Operation ## 0 Call Center options will be displayed based on the value of OPSTAT (default) ## 1 Call Center options will be displayed based on the value of OPSTATCC ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET OPSTATCC 1 ## ## AGTACTIVESK specifies which softkeys will be displayed for active call center calls ## Value Operation ## 0 Softkeys will be displayed in the default order, depending on administered features (default) ## 1 The positions of the Release and Transfer softkeys will be interchanged, otherwise the same as 0 ## 2 The Release softkey will not be displayed, otherwise the same as 0 ## 3 The soft keys will be labeled as an active call in a non-call center environment from left to right: Hold, Conf, Transfer, Drop. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later (values 0-3) ## 96x1 H.323 R6.2.1 and later (values 0-2); value 3 is added in 96x1 H.323 R6.4 and later. ## SET AGTACTIVESK 2 ## ## AGTCALLINFOSTAT specifies whether the caller-information line will be displayed. ## Value Operation ## 0 The caller-information line will not be displayed ## 1 The caller-information line will be displayed (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTCALLINFOSTAT 0 ## ## AGTFWDBTNSTAT specifies whether the Forward button will be disabled for call center agents. ## Note that the value of CALLCTRSTAT must be 1 for AGTFWDBTNSTAT to be used. ## Value Operation ## 0 The Forward button will operate normally ## 1 The Forward button will be disabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTFWDBTNSTAT 0 ## ## AGTGREETINGSTAT specifies whether or not an agent greeting may be created, modified, played and deleted. ## Note that the value of CALLCTRSTAT must be 1 for AGTGREETINGSTAT to be used. ## Value Operation ## 0 Agent greetings will be disabled ## 1 Agent greetings will be enabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTGREETINGSTAT 0 ## ## AGTGREETLOGOUTDEL specifies whether agent greetings will be deleted when the agent logs out ## Value Operation ## 0 Agent greetings will not be deleted when the agent logs out ## 1 Agent greetings will be deleted when the agent logs out (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2.1 and later ## SET AGTGREETLOGOUTDEL 0 ## ## AGTVUSTATID specifies the Vu-stat format number for Agent ID determination. ## Valid values are 0 through 50; the default value is 0. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2 and later ## SET AGTVUSTATID 33 ## ## AGTLOGINFAC specifies the Feature Access Code to be used for logging in Call Center agents. ## Valid values are 1 to 4 dialable characters (0-9, * and #); the default value is "#94". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTLOGINFAC #33 ## ## AGTLOGOUTFAC specifies the Feature Access Code to be used for logging out Call Center agents. ## Valid values are 1 to 4 dialable characters (0-9, * and #); the default value is "#95". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTLOGOUTFAC #34 ## ## AGTSPKRSTAT specifies how the Speaker button functions for call center agents. ## by default (if AGTSPKRSTAT is configured to 1) the Speaker button, will function ## normally unless CALLCTRSTAT is 1 and a call center agent is logged in. In ## latter case it will be disabled (not function at all). ## Value Operation ## 0 The Speaker button functions normally ## 1 The Speaker button will be disabled (default) ## This value only applies if the following conditions are also met: ## the value of CALLCTRSTAT is 1 and a call center agent is logged in. ## 2 The Speaker button functions as a Release button ## This value only applies if the following conditions are also met: ## the value of CALLCTRSTAT is 1, a call center agent is logged in, ## the telephone is a 9641 and has a Release button administered ## (otherwise the default behavior will apply). ## 3 The Speaker button functions as a Release button ## This value only applies if the following conditions are also met: ## the value of CALLCTRSTAT is 1, a call center agent is logged in and ## the telephone has a Release button administered. ## (otherwise the default behavior will apply). ## 4 The Speaker button functions as a Release button ## This value only applies if the following condition is also met: ## the telephone has a Release button administered. ## (otherwise the default behavior will apply). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later (values 0-4) ## 96x1 H.323 R6.3 and later (values 0-4) ## 96x1 H.323 R6.1 and later (values 0-2) ## SET AGTSPKRSTAT 0 ## ## AGTTIMESTAT specifies whether the date and time will be displayed on the top line for call center agents. ## Note that the value of CALLCTRSTAT must be 1 for AGTTIMESTAT to be used. ## Value Operation ## 0 The date and time will be displayed normally ## 1 The display of the date and time will be disabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTIMESTAT 0 ## ## AGTTRANSLTO specifies the text string used by the call server as a translation of the English ## string "to" in call-associated display messages. This string is used by the telephone when ## parsing received display messages to decide whether to play an agent greeting. ## Valid values are 1 to 6 UTF-8 characters; the default value is "to". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTRANSLTO to ## ## AGTTRANSLCLBK specifies the text string used by the call server as a translation of the English ## string "callback" in call-associated display messages. This string is used by the telephone ## when parsing received display messages to decide whether to play an agent greeting. ## Valid values are 1 to 10 UTF-8 characters; the default value is "callback". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTRANSLCLBK callback ## ## AGTTRANSLPRI specifies the text string used by the call server as a translation of the English ## string "priority" in call-associated display messages. This string is used by the telephone ## when parsing received display messages to decide whether to play an agent greeting. ## Valid values are 1 to 8 UTF-8 characters; the default value is "priority". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTRANSLPRI priority ## ## AGTTRANSLPK specifies the text string used by the call server as a translation of the English ## string "park" in call-associated display messages. This string is used by the telephone ## when parsing received display messages to decide whether to play an agent greeting. ## Valid values are 1 to 6 UTF-8 characters; the default value is "park". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTRANSLPK park ## ## AGTTRANSLICOM specifies the text string used by the call server as a translation of the English ## string "ICOM" in call-associated display messages. This string is used by the telephone ## when parsing received display messages to decide whether to play an agent greeting. ## Valid values are 1 to 6 UTF-8 characters; the default value is "ICOM". ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET AGTTRANSLICOM ICOM ## ## CCLOGOUTIDLESTAT specifies whether the Headset audio path and LED ## will be turned off or left on when a call center agent logs out, ## if the agent is not on a call. ## Value Operation ## 0 The Headset audio path and LED will be turned off (default) ## 1 The Headset audio path and LED will be left on if they are already on ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET CCLOGOUTIDLESTAT 1 ## ## LOCALZIPTONEATT specifies the attenuation of zip tone level. ## The possible values are in the range of 0-95 dB. ## The default value is 35 dB. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET LOCALZIPTONEATT 35 ## ## AGENTGREETINGSDELAY specifies the time in milisecconds between call ## autoanswer and playing of agent greeting. ## The default is 700 ms and valid values are 0 - 3000 ms ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## SET AGENTGREETINGSDELAY 1000 ## ## AGTCAINFOLINE controls presentation of call associated information in the agent information line ## when the phone is in half width screen mode. ## Value Operation ## 0 the Agent Information Line presents agent-oriented information only ## 1 the Agent Information Line presents agent-oriented information as well ## to call associated information (as supported in pre 6.6 release) (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## SET AGTCAINFOLINE 0 ## ########## RECORDING TONE SETTINGS ####### ## ## RECORDINGTONE specifies whether Call Recording Tone will be generated on active calls. ## Value Operation ## 0 Call Recording Tone will not be generated (default) ## 1 Call Recording Tone will be generated ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2 and later ## B189 H.323 R1.0 and later ## SET RECORDINGTONE 1 ## ## RECORDINGTONE_INTERVAL specifies the number of seconds between Call Recording Tones. ## Valid values are 1 through 60; the default value is 15. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.0.0 and later ## 96x1 H.323 R6.2 and later ## B189 H.323 R1.0 and later ## SET RECORDINGTONE_INTERVAL 10 ## ## RECORDINGTONE_VOLUME specifies the volume of the Call Recording Tone in 5dB steps. ## Value Operation ## 0 The tone volume is equal to the transmit audio level (default) ## 1 The tone volume is 45dB below the transmit audio level ## 2 The tone volume is 40dB below the transmit audio level ## 3 The tone volume is 35dB below the transmit audio level ## 4 The tone volume is 30dB below the transmit audio level ## 5 The tone volume is 25dB below the transmit audio level ## 6 The tone volume is 20dB below the transmit audio level ## 7 The tone volume is 15dB below the transmit audio level ## 8 The tone volume is 10dB below the transmit audio level ## 9 The tone volume is 5dB below the transmit audio level ## 10 The tone volume is equal to the transmit audio level ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.0.0 and later ## 96x1 H.323 R6.2 and later ## B189 H.323 R1.0 and later ## SET RECORDINGTONE_VOLUME 8 ## ########## CALL CENTER AND SKS SETTINGS (16xx, 96x1 H.323, J169/J179 H.323 and Avaya Vantage Basic Application SIP) ########## ## ## Note for 96x1 H.323 phones: The below parameters are supported by 96x1 H.323 for Call Center Agent registered to ## Avaya Communication Manager. 96x1 H.323 telephone recognizes the user has logged into the call center if the LEDs associated with at least one of the ## following buttons are On: any Auxiliary Work buttons (buttonType 52), Manual In (buttonType 93), Auto In (buttonType 92), or ## After Call Work (buttonType 91) buttons AND the value of CALLCTRSTAT is 1. ## ## CCBTNSTAT specifies whether the values of ## CONFSTAT, DROPSTAT, HOLDSTAT, MUTESTAT, and XFERSTAT ## are used for enabling and disabling the buttons associated with those parameters. ## Value Operation ## 0 The telephone uses the values of those parameters ## 1 The telephone ignores the values those parameters (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## 16xx H.323 R1.3.3 and later ## SET CCBTNSTAT 1 ## ## CONFSTAT specifies whether the Conference button is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The Conference button is disabled when CCBTNSTAT is 0 (default for 16xx) ## 1 The Conference button is enabled (default for 96x1 and Avaya Vantage Basic Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## 16xx H.323 R1.3.3 and later ## SET CONFSTAT 1 ## ## DROPSTAT specifies whether the Drop button is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The Drop button is disabled when CCBTNSTAT is 0 (default for 16xx) ## 1 The Drop button is enabled (default for 96x1) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## 16xx H.323 R1.3.3 and later ## SET DROPSTAT 1 ## ## HEADSTAT specifies whether the Headset button is enabled or disabled when CCBTNSTAT is 0. ## It is ignored by telephones that do not have a Headset button. ## Value Operation ## 0 The Headset button is disabled when CCBTNSTAT is 0 (default) ## 1 The Headset button is enabled. ## This parameter is supported by: ## 16xx H.323 R1.3.3 and later ## SET HEADSTAT 1 ## ## HOLDSTAT specifies whether the Hold button is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The Hold button is disabled when CCBTNSTAT is 0 (default for 16xx) ## 1 The Hold button is enabled (default for 96x1 and Avaya Vantage Basic Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## 16xx H.323 R1.3.3 and later ## SET HOLDSTAT 1 ## ## HOOKSTAT specifies whether the switchhook is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The switchhook is disabled when CCBTNSTAT is 0 (default) ## 1 The switchhook is enabled. ## This parameter is supported by: ## 16xx H.323 R1.3.3 and later ## SET HOOKSTAT 1 ## ## MUTESTAT specifies whether the Mute button is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The Mute button is disabled when CCBTNSTAT is 0 (default for 16xx) ## 1 The Mute button is enabled (Default for Avaya Vantage Basic Application SIP) ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 16xx H.323 R1.3.3 and later ## SET MUTESTAT 1 ## ## XFERSTAT specifies whether the Transfer button is enabled or disabled when CCBTNSTAT is 0. ## Value Operation ## 0 The Transfer button is disabled when CCBTNSTAT is 0 (default for 16xx) ## 1 The Transfer button is enabled (default for 96x1 and Avaya Vantage Basic Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## 16xx H.323 R1.3.3 and later ## SET XFERSTAT 1 ## ################## TRUSTED CERTIFICATES AND GENERAL CERTIFICATES SETTINGS ################## ## ## TRUSTCERTS specifies a list of names of files that contain copies of CA certificates ## (in PEM format) that will be downloaded, saved in non-volatile memory, ## and used by the telephone to authenticate received identity certificates. ## The list can contain up to 255 characters. ## Values are separated by commas without intervening spaces. ## The default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; support of up to 100 PEM and DER format root and intermediate trusted certificates. ## The list can contain up to 1024 characters. Avaya Vantage Open application does not use the downloaded trusted certificates. ## However, when Avaya Vantage Open application is installed, this parameter is used to download trusted certificates for ## to be used Avaya Vantage device (for example, 802.1x EAP-TLS) or by other applications (for example, Android Browser, etc.). ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R2.0 and later ## 96x0 SIP R1.0 and later ## ## SET TRUSTCERTS av_prca_pem_2033.txt,av_sipca_pem_2027.txt,av_csca_pem_2032.txt ## Note: The above is list of Avaya trusted certificates. You shall only use ## the ones that are required for your setup. ## Note: 96x1 H.323 R6.6 and later supports also intermediate certificates download for cases ## where servers do not provided the full certificate chain up to the root CA. There is no support ## for certificate signature validation up to intermediate certificate. Certificate signature validation ## is always supported up to the root CA. ## Note: Avaya Vantage Basic application and Avaya Equinox uses the Android trusted certificate repository and the downloaded certificates. ## using TRUSTCERTS. ## ## MAX_TRUSTCERTS specifies the maximum number of trusted ## certificates, which are defined by TRUSTCERTS ## parameter, can be downloaded to the phone. ## Note: each trusted certificate file may contain more than one certificate. MAX_TRUSTCERTS enforces the number of certificates. ## Valid value: 1 to 10, default: 6 ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP 7.1.0.0 and later ## SET MAX_TRUSTCERTS 8 ## ## ENABLE_PUBLIC_CA_CERTS specifies whether the embedded public root CA certificates are used for services other than Device Enrollment Service (DES) ## and re-directed file server using DES. DES always used the embedded public root CA certificates (even if ENABLE_PUBLIC_CA_CERTS is 0). ## For the re-directed file server using DES, there is use of embedded public root certificates if DES service did not provide private CA. If DES provides private CA, then the ## embedded public root CA certificates are ignored (however if DES is re-triggered from admin menu and private CA is provided from DES then the embedded public root CA certificates will be used according to ENABLE_PUBLIC_CA_CERTS). ## For rest of the services, this parameter controls whether embedded public root CA certificates are used (in addition, to downloaded trusted certificates) or not (only downloaded trusted certificates are used). ## If DES did not provide private CA, then the ENABLE_PUBLIC_CA_CERTS is set to "1" without ability to change it. If DES provides private CA, then this parameter is configurable (in such case, TRUSTCERTS shall include ## DES service private CA, else the phone will not be able to re-connect to the re-directed file server). ## For cases where DES is not used, then the parameter is fully configurable and if ENABLE_PUBLIC_CA_CERTS is "0" and no downloaded trusted certificates (TRUSTCERTS=="") then the phone trusts for any HTTP/S file server ## for configuration / image download and fails with rest of services (PPM/SIP, AADS, etc). If either ENABLE_PUBLIC_CA_CERTS is "1" and/or TRUSTCERTS<> "" then the service must have identity certificate that can be validated ## using the embedded public root CA certificates (if ENABLE_PUBLIC_CA_CERTS is "1") or downloaded trusted certificates (if TRUSTCERTS <>"") - there is no exception to configuration and software files download from the HTTP/S file server ## in such case. ## Value Operation ## 0 Embedded public CA certs are not trusted (Default). ## 1 Embedded public CA certs are always trusted (in addition to trusted certificates downloaded according to TRUSTCERTS) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later; the embedded public root CA certificates are Android public root certificates which can be viewed in the settings application --> Security --> Trusted credentials --> SYSTEM. ## SET ENABLE_PUBLIC_CA_CERTS 1 ## Note: This parameter is used on Avaya Vantage devices to enable all Android root CA certificates for non-Android applications such as AADS, configuration and firmware download using HTTPS, PPM, 802.1x EAP-TLS, SCEP over HTTPS. ## This parameter cannot be used to disable Android root CA certificates for Android applications. CA_CERT_BLACKLIST shall be used to disable Android root CA certificates for both Android and non-Android applications. ## ## TLSSRVRID specifies whether a certificate will be trusted only if the ## identity of the device from which it is received matches the certificate, ## per Section 3.1 of RFC 2818. ## Value Operation ## 0 Identity matching is not performed ## 1 Identity matching is performed (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Not used by Avaya Vantage Open application. ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; Supported by SIP/PPM and file downloads. ## B189 H.323 R1.0 and later ## 96x0 H.323 R2.0 and later ## 96x0 SIP R2.0 and later ## TLSSRVRID is not supported by 96x1 H.323 phones and instead ## TLSSRVRVERIFYID is supported (see below) ## SET TLSSRVRID 0 ## ## TLSSRVRVERIFYID Specifies whether the identity of a TLS server is checked against its certificate. ## This parameter obsoletes TLSSRVRID for 96x1 H.323 phones. ## 0 Identity of a TLS server is NOT checked against its certificate (default). ## 1 Identity of a TLS server is checked against its certificate. The validation of server identity ## is applicable for IPSec VPN with certificate based authentication (using NVSGIP) , Backup/restore over ## HTTPS (using BRURI), HTTPS file server (using TLSSRVR), WML browser (using WMLHOME), ## H.323 over TLS signaling (using MCIPADD). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## B189 H.323 R6.6 and later ## SET TLSSRVRVERIFYID 1 ## ## FQDN_IP_MAP specifies a comma separated list of name/value pairs where the name is an FQDN and the value is an IP address. ## The IP address may be IPv6 or IPv4 but the value can only contain one IP address. Default is "". String length is up to 255 ## characters. No spaces are allowed inside the string. ## The purpose of this parameter is to support cases where the server certificate Subject Common Name of Subject Alternative Names ## include FQDN (instead of IP address) and the SIP_CONTROLLER_LIST is defined using IP address. The main use case is for Avaya Aura SM/PPM connectivity ## where the SIP controller list returned from Aura (PPM) to the endpoint is IP address only while server certificate is defined with FQDN. ## Internet trusted CAs prefer signing of Internet public server certificates with FQDN only. ## This parameter is supported with any phone service running over TLS. Though, the main use case if for Avaya Aura SM/PPM services. ## This parameter is not to be used as an alternative to a DNS lookup or reverse DNS lookup. ## The reverse case will not be supported. If the phone is accessing a server using an FQDN and the server’s certificate only contains an IP address, ## this will be considered a failure and the FQDN_IP_MAP will not be used. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0) (IPv6 is not yet supported) ## J169/J179 SIP R1.5.0 ## 96x1 SIP R7.1.0.0 and later ## SET FQDN_IP_MAP "sm1.avaya.com=135.20.230.199,sm1.avaya.com=2000::204,sm2.avaya.com=135.20.230.201,ppm.ottawa.avaya.com=2000::207" ## ## SERVER_CERT_RECHECK_HOURS specifies the number of hours after which certificate expiration ## and OCSP will be used (if OCSP is enabled) to recheck the revocation and expiration status ## of the certificates that were used to establish a TLS connection. ## SERVER_CERT_RECHECK_HOURS is applicable for H.323 over TLS signaling only in 96x1 H.323 R6.6. ## SERVER_CERT_RECHECK_HOURS is applicable for SIP and 802.1x EAP-TLS when used by J129 SIP R1.0.0.0 and later. ## Valid values are: 0-32767. A value of 0 means that periodic checks will not be done. ## The default is 24. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET SERVER_CERT_RECHECK_HOURS 30 ## ## CERT_WARNING_DAYS specifies how many days before the expiration of a certificate that a warning ## should first appear on the phone screen. This includes trusted certificates, OCSP certificates and identity certificate. ## Log and syslog message will be generated as well. The warning will reappear every 7 days. ## Valid values are: 0-99 (60 is default), where 0 means no certificate expiration warning will be generated. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## SET CERT_WARNING_DAYS 30 ## ## DELETE_MY_CERT specifies whether the installed identity certificate (using SCEP or PKCS12 file download) will be deleted. ## Value Operation ## 0 Installed Identity certificate remain valid (Default) ## 1 Installed Identity certificate is removed. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET DELETE_MY_CERT 1 ## ## CA_CERT_BLACKLIST specifies comma separated list of SHA-1 signatures of public keys of embedded Android trusted certificates that shall not be trusted. ## The default value is "". String length is up to 1024 characters. This parameter can be used in case of one the embedded Android trusted certificates is revoked ## without a need for software upgrade of the device. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET CA_CERT_BLACKLIST ceffe60cb8a6cd49fad49ac6e09e8ed329c6e633 ## ## BLOCK_CERTIFICATE_WILDCARDS specifies whether the endpoint will accept server identity certificates with wildcards. ## Value Operation ## 0 Accept wildcards in certificate (default) ## 1 Do not accept wildcards in certificates ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET BLOCK_CERTIFICATE_WILDCARDS 1 ## ################## TLS SETTINGS ################## ## ## TLS_SECURE_RENEG Specifies whether a TLS session will be terminated if the peer does not support secure renegotiation. ## Value Operation ## 0 TLS secure renegotiation is not required from peer (Default) ## 1 TLS secure renegotiation is required from peer ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## SET TLS_SECURE_RENEG 1 ## ## TLS_VERSION controls TLS version used for all TLS connections (except SLA monitor agent) ## Value Operation ## 0 TLS versions 1.0 and 1.2 are supported (default). ## 1 TLS version 1.2 only is permitted. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Not used by Avaya Vantage Open application. ## 96x1 SIP R7.0.1.0 and later releases ## 96x1 H.323 R6.6.2 and later releases ## B189 H.323 R6.6.2 and later releases ## SET TLS_VERSION 1 ## ################ HTTP PROXY SERVER SETTINGS ############## ## ## HTTPPROXY specifies the address of the HTTP proxy server used by SIP ## telephones to access an SCEP server that is not on the enterprise network. ## Zero or one IP address in dotted decimal or DNS name format, ## optionally followed by a colon and a TCP port number. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; HTTPPROXY is NOT supported for SCEP, but for Android HTTP based applications. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; HTTPPROXY is NOT supported for SCEP, but for WEB Browser and Exchange. ## 96x0 SIP R1.0 and later ## Note that in H.323 telephones, SCEP uses WMLPROXY. ## SET HTTPPROXY proxy.mycompany.com ## ## HTTPEXCEPTIONDOMAINS specifies a list of one or more domains, ## separated by commas without any intervening spaces, ## for which HTTPPROXY will not be used. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; HTTPPROXY is NOT supported for SCEP, but for Android HTTP based applications. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; HTTPEXCEPTIONDOMAINS is NOT supported for SCEP, but for WEB Browser and Exchange. ## 96x0 SIP R1.0 and later ## Note that in H.323 telephones, SCEP uses WMLEXCEPT. ## SET HTTPEXCEPTIONDOMAINS mycompany.com ## ## HTTPPROXYAUTOCONFIGURL specifies the Proxy Auto configuration URL (PAC URL). The value can be zero or one PAC URL. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.0.0 and later; ## SET HTTPPROXYAUTOCONFIGURL http://example.net/example.com/inet ## SET HTTPPROXYAUTOCONFIGURL example.com/inet ## ## HTTPPROXYSOURCE specifies the HTTP Proxy Source (Manual, None or Proxy Auto-Config). ## Value Operation ## 0 "None" - No HTTP Proxy is configured. ## 1 "Manual" - HTTP Proxy is configured according to HTTPPROXY and HTTPEXCEPTIONDOMAINS (default) ## 2 "Proxy Auto-configuration" - HTTP Proxy is configured according to HTTPPROXYAUTOCONFIGURL. ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.0.0 and later; ## SET HTTPPROXYSOURCE 2 ## ## HTTP_PROXY_CSDK_ENABLE specifies whether CSDK shall use the OS reverse proxy settings if exist. ## Value Operation ## 0 No use of any HTTP proxy by CSDK ## 1 Enable CSDK to use the HTTP proxy configured in OS, and enforce the HTTP Tunneling in GME without going through the STUN check ## if HTTPUA for the call is going through a HTTP proxy. ## 2 Enable CSDK to use the HTTP proxy configured in OS, and still enable the GME for STUN check before HTTP Tunneling (Default) ## This parameter is supported by: ## Avaya Equinox 3.4 and later ## SET HTTP_PROXY_CSDK_ENABLE 0 ## ################ HTTP TUNNELING SETTINGS ############## ## ## ENABLE_MEDIA_HTTP_TUNNEL specifies whether Media HTTP Tunneling feature is enabled or disabled. ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## Avaya Equinox 3.2 and later ## SET ENABLE_MEDIA_HTTP_TUNNEL 1 ## ################ CAPTIVE PORTAL SETTINGS ############## ## ## CAPTIVE_PORTAL_SERVER specifies the URL of the captive portal server. ## Android supports a detection mechanism of whether the device is behind captive portal which requires HTTP authentication ## in order to access the Internet. The mechanism is based on sending HTTP requests to http://clients3.google.com/generate_204. ## If HTTP response 204 is returned with null content then the device assumes that it is correctly connected to the Internet, ## else captive portal is assumed. There may be customers who block access to the Internet (or to certain Internet pages) ## and for these customers the detection mechanism will notice that the device is not connected to the Internet and raise notification. ## For these customers, the CAPTIVE_PORTAL_SERVER parameter shall be configured to "" (default). "" (null string) implies that the detection mechanism is disabled. ## There may be other customers that want to have their own captive portal server. These customers can configure their own HTTP server. ## As long their HTTP server does NOT return 204 with null content then the device will assume it behind captive portal ## and redirect the user to the relevant HTTP authentication page. As the default of CAPTIVE_PORTAL_SERVER is null, ## then customers may need to do staging of the device and set CAPTIVE_PORTAL_SERVER to the relevant captive portal server ## in the settings file before deploying the device in the field (mainly for cases where captive portal prevent download of ## configuration files in the field) OR configure CAPTIVE_PORTAL_SERVER in the local DHCP server deployed in the field. ## Captive portal is supported over both Wi-Fi and Ethernet interfaces. ## Zero or one URL in the following format: ## [http://]hostname[:port][/path] ## [https://]hostname[:port][/path] ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; the default value is "connectivitycheck.gstatic.com". ## H1xx SIP R1.0.1 and later ## SET CAPTIVE_PORTAL_SERVER http://clients3.google.com/generate_204 ## ###################### SCEP SETTINGS ##################### ## ## Note: When FIPS_ENABLED is set to 1 (for endpoints which support FIPS mode), SCEP shall not be used. ## If identity certificate was generated before FIPS_ENABLED is set to 1, the phone will keep using it. ## However, it is NOT recommended to use identity certificate generated using SCEP when FIPS_ENABLED is 0 when ## the phone is configured to work in FIPS mode (FIPS_ENABLED==1). It is recommended to CLEAR (return to factory defaults) ## before configuring the phone to FIPS mode (FIPS_ENABLED==1). ## ## MYCERTURL specifies the URL of the SCEP server from which ## the telephone should obtain an identity certificate, ## if it does not already have one from that server. ## Zero to 255 ASCII characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - the URL can be https or http. ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; the URL can be https or http. Avaya Vantage Open application does not use identity certificate. ## J129 SIP R1.0.0.0 up to R1.1.0.0 (excluded) - the URL can be only http; R1.1.0.0 and later - the URL can be https or http. ## 96x1 H.323 R6.0 and later ## H1xx SIP R1.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 up to R7.1.0.0 (excluded) - the URL can be only http; R7.1.0.0 and later - the URL can be https or http. ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTURL http://certsrvr.trustus.com/mscep/mscep.dll ## SET MYCERTURL https://10.10.10.10/certsrv/mscep/mscep.dll ## ## MYCERTCN specifies the Common Name (CN) used in the SUBJECT of an SCEP ## certificate request. The value must be a string that contains either ## "$SERIALNO" (which will be replaced by the telephone's serial number) or ## "$MACADDR" (which will be replaced by the telephone's MAC address), ## but it may contain other characters as well, including spaces. ## Eight ("$MACADDR") to 255 characters; the default value is "$SERIALNO". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Avaya Vantage Open application does not use identity certificate. ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## Note that prior to R2.6.8, 96x0 SIP releases only support ## "$MACADDR" or "$SERIALNO" as a value, not additional characters. ## SET MYCERTCN "Avaya telephone with MAC address $MACADDR" ## ## MYCERTDN specifies the part the SUBJECT of an SCEP certificate request ## that is common for all telephones. It must begin with a "/" and may ## include Organizational Unit, Organization, Location, State and Country. ## Zero to 255 ASCII characters; the default value is null (""). ## Note: It is recommended that "/" be used as a separator between components. ## Commas have been found to not work with some servers. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Avaya Vantage Open application does not use identity certificate. ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTDN /C=US/ST=NJ/L=MyTown/O=MyCompany ## ## MYCERTCAID specifies an identifier for the CA certificate with which ## the SCEP certificate request is to be be signed, if the server hosts ## multiple Certificate Authorities. ## Zero to 255 ASCII characters; the default value is "CAIdentifier". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Avaya Vantage Open application does not use identity certificate. ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTCAID EjbSubCA ## ## MYCERTKEYLEN specifies the bit length of the public and private keys ## generated for the SCEP certificate request. ## 4 ASCII numeric digits, "1024" through "2048"; the default value is "1024". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; only "2048" is supported ## J169/J179 H.323 R6.7 and later; only value "2048" is supported. ## Avaya Vantage Devices SIP R1.0.0.0 and later; only value "2048" is supported. ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later; default value is "2048" in 96x1 SIP R6.5+. 96x1 SIP R7.1.0.0 and later - only "2048" is supported. ## H1xx SIP R1.0 and later; default value is "2048" in H1xx SIP R1.0.1+. ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTKEYLEN 1024 ## ## MYCERTRENEW specifies the percentage of the identity certificate's ## Validity interval after which renewal procedures will be initiated. ## Valid values are 1 through 99; the default value is 90. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTRENEW 90 ## ## MYCERTREPLACE specifies the percentage of the identity certificate's ## Validity interval after which replacement procedures will be initiated. ## Replacement procedure generates new public/private keys for the identity certificate. ## Valid values are 1 through 99; the default value is 90. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; Avaya Vantage Open application does not use identity certificate. ## SET MYCERTREPLACE 90 ## ## MYCERTWAIT specifies the telephone's behavior if the SCEP server ## indicates that the certificate request is pending manual approval. ## Value Operation ## 0 Poll the SCEP server periodically in the background ## 1 Wait until a certificate is received or the request is rejected (default) ## This parameter is supported by: ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## SET MYCERTWAIT 1 ## ## SCEPPASSWORD specifies the password to be included (if not null) ## in the challengePassword attribute of an SCEP certificate request. ## Values may contain 0 to 32 ASCII characters (50 ASCII characters in 96x1/B189 H.323 6.6 and later); ## the default value is "$SERIALNO". ## If the value contains "$SERIALNO", it will be replaced by the telephone's serial number. ## If the value contains "$MACADDR", it will be replaced by the telephone's MAC address in hex. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; please note that if SCEP is configured and SCEPPASSWORD is empty, ## the user will be prompted to enter the SCEP password. ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## Avaya Vantage Devices SIP R1.0.0.0 and later; please note that if SCEP is configured and SCEPPASSWORD is empty, ## the user will be prompted to enter the SCEP password. Avaya Vantage Open application does not use identity certificate. ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## B189 H.323 R6.6 and later ## Note that to maintain the security of the password, this parameter should ## not be set in a file that is accessible on an enterprise network, ## it should only be set in a restricted staging configuration. ## SET SCEPPASSWORD "$SERIALNO" ## ## MYCERTKEYUSAGE specifies the purpose(s) for which a certificate is issued. ## 0 to 255 ASCII characters. List of text strings, separated by commas without any intervening spaces, ## that will be compared to the values specified for the X.509 KeyUsage extension ## and for each matching value, the corresponding bit will be set in the SCEP PKCSReq; ## Invalid strings will be ignored; Possible values are: "digitalSignature", "nonRepudiation", ## "keyEncipherment", "dataEncipherment", "keyAgreement", "keyCertSign", ## "cRLSign", "encipherOnly", "decipherOnly". The default is null string (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## B189 H.323 R6.6 and later ## SET MYCERTKEYUSAGE digitalSignature, keyEncipherment ## ## SCEPENCALG specifies SCEP Encryption Algorithm. ## Value Operation ## 0 DES (default) ## 1 AES-256 ## This parameter is supported by: ## J129 SIP R4.0.0.0 and later ## SET SCEPENCALG 1 ## ###################### PKCS12 SETTINGS ##################### ## ## PKCS12URL specifies the URL to be used to download a PKCS #12 file ## containing an identity certificate and its private key. ## 0 to 255 ASCII characters, zero or one URL. The value can be a string that contains either ## "$SERIALNO" (which will be replaced by the telephone's serial number) or "$MACADDR" ## (which will be replaced by the telephone's MAC address), but it may contain other characters as well. ## If $MACADDR is added to the URL then the PKCS12 filename on the file server shall include MAC address ## without colons (i.e., 6 pairs of ASCII hexadecimal characters AABBCCDDEEFF with hex characters A-F ## encoded as upper-case characters). For example, if Ethernet MAC address of a specific phone ## is: 00-24-D7-E4-2E-98 and the PKCS12URL is: http://pkc12file_$MACADDR.cer, then the filename of the ## PKCS12 file for this phone on the file server shall be: pkc12file_0024D7E42E98.cer. ## PKCS12 file download is preferred over SCEP if PKCS12URL is defined. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; Same note as for 96x1 SIP R7.1.0.0 below. ## 96x1 SIP R7.1.0.0 and later; The URL can specify the file server using an IPv4/IPv6 address or an FQDN. ## An empty parameter value means that PKCS#12 identity certificate download is disabled (if there is ## an already existing PKCS12 file on the phone then it will not be deleted if PKCS12URL is set to "". ## DELETE_MY_CERT shall be set to 1 or CLEAR procedure shall be used to delete existing PKCS12 file). ## If the parameter is not empty, PKCS#12 file installation is preferred over SCEP. ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later; Avaya Vantage Open application does not use identity certificate. ## J129 SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## SET PKCS12URL http://pkc12file_$MACADDR.cer ## ## PKCS12_PASSWD_RETRY specifies the number of retries for entering PKCS12 file password. ## Values: 0-100 and the default is 3. 0 means no retry. ## If user failed to enter the correct PKCS12 file password after PKCS12_PASSWD_RETRY retries, then the ## phone will continue the startup sequence without installation of PKCS12 file. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later; Avaya Vantage Open application does not use identity certificate. ## SET PKCS12_PASSWD_RETRY 4 ## ## PKCS12PASSWORD specifies the PKCS12 file password. The default value is "". It is recommended to set this parameter in 46xxsettings file when this file ## is downloaded in secure network such as in staging center. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later; Avaya Vantage Open application does not use identity certificate. ## SET PKCS12PASSWORD "PKCS12PASS" ## ##################### 802.1X SETTINGS #################### ## ## DOT1XSTAT specifies the 802.1X Supplicant operating mode. ## Value Operation ## 0 Supplicant disabled (default, unless indicated otherwise below) ## 1 Supplicant enabled, but responds only to received unicast EAPOL messages ## 2 Supplicant enabled; responds to received unicast and multicast EAPOL messages ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R2.0 and later ## 96x0 SIP R2.0 and later (default was 1 prior to R2.4.1) ## SET DOT1XSTAT 1 ## ## DOT1X specifies the 802.1X pass-through operating mode. ## Pass-through is the forwarding of EAPOL frames between the telephone's ## Ethernet line interface and its secondary (PC) Ethernet interface ## Value Operation ## 0 EAPOL multicast pass-through enabled without proxy logoff (default) ## 1 EAPOL multicast pass-through enabled with proxy logoff ## 2 EAPOL multicast pass-through disabled ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch, All K155 devices have embedded Ethernet switch. ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R2.0 and later ## 96x0 SIP R2.0 and later ## Note: In 96x0 H.323 releases 1.0 through 1.5, DOT1X is supported, but it controls both Supplicant and pass-through operation. ## In these releases, operation is as follows: ## Value Operation ## 0 Unicast Supplicant and multicast pass-through enabled without proxy logoff (default) ## 1 Unicast Supplicant and multicast pass-through enabled with proxy logoff ## 2 Unicast or multicast Supplicant operation enabled, without pass-through ## SET DOT1X 1 ## ## DOT1XEAPS specifies the authentication method to be used by 802.1X. ## Valid values are "MD5" (the default) and "TLS". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.2.1 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R3.1.4 and later ## 96x0 SIP R2.0 and later ## B189 H.323 R6.6 and later ## SET DOT1XEAPS MD5 ## ## DOT1XWAIT specifies whether the telephone will wait for ## 802.1X to complete before proceeding with startup. ## Value Operation ## 0 Does not wait for 802.1X to complete (default) ## 1 Waits for 802.1X to complete ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R3.2.2 and later ## SET DOT1XWAIT 1 ## ################ FIPS SETTINGS ########################### ## ## FIPS_ENABLED specifies whether only FIPS-approved cryptographic algorithms will be supported. ## Value Operation ## 0 No restriction on using non FIPS-approved cryptographic algorithms (default) ## 1 Use only FIPS-approved cryptographic algorithms using embedded FIPS 140-2-validated cryptographic module (Per NIST Certificate #1747, ## for the exact operational environment used by the endpoint please refer to the Avaya support team). ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP 7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET FIPS_ENABLED 1 ## ################ OCSP (Online Certificate Status Protocol) SETTINGS ########################### ## ## OCSP_ENABLED specifies whether OCSP will be used to check revocation status of certificates. ## Value Operation ## 0 OCSP is disabled (default) ## 1 OCSP is enabled. OCSP will be used to check revocation status for the certificates ## presented by peers for any TLS connection (H.323 signaling over TLS, HTTPS, ## 802.1x with EAP-TLS, SLA Mon agent, IPSec VPN, SSO) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_ENABLED 1 ## ## OCSP_ACCEPT_UNK specifies whether in cases where certificate revocation status for a specific certificate ## cannot be determined to bypass certificate revocation operation for this certificate. ## Value Operation ## 0 Certificate is considered to be revoked if the certificate revocation status is unknown. TLS connection ## will be closed. ## 1 Certificate revocation operation will accept certificates for which the certificate revocation ## status is unknown (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_ACCEPT_UNK 1 ## ## OCSP_NONCE specifies whether a nonce will be included in OCSP requests and expected in OCSP responses. ## Value Operation ## 0 Nonce is NOT added to OCSP packets ## 1 Nonce is added to OCSP packets (Default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_NONCE 1 ## ## OCSP_URI specifies the URI of an OCSP responder. The URI can be an IP address or hostname. ## The default is "". 0 to 255 ASCII characters - zero or one URI. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_URI http://clients1.google.com/ocsp ## ## OCSP_URI_PREF specifies the preferred URI to use for OCSP requests if more than one is available. ## Value Operation ## 1 Use the OCSP_URI first and then the OCSP field of the Authority Information Access (AIA) extension ## of the certificate being checked (Default) ## 2 Use the OCSP field of the Authority Information Access (AIA) extension of the ## certificate being checked first and then OCSP_URI ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_URI_PREF 0 ## ## OCSP_TRUSTCERTS specifies list of OCSP trusted certificates which are used as ## OCSP signing authority for the certificate that its revocation status is being checked. ## This is needed in case the OCSP responder uses a different CA than the root CA of the certificate that ## its revocation status is being checked. ## 0 to 255 ASCII characters: zero or more file names or URLs, separated by commas without any intervening spaces ## The default is "". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later ## 96x1 H.323 R6.6 and later ## SET OCSP_TRUSTCERTS ocsp.cer ## ## OCSP_HASH_ALGORITHM specifies the hashing algorithm for OCSP request. ## Value Operation ## 0 SHA-1 (default) ## 1 SHA-256 ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP 7.1.0.0 and later ## SET OCSP_HASH_ALGORITHM 1 ## ## OCSP_USE_CACHE specifies if OCSP caching is used. ## Value Operation ## 0 Do not to use OCSP caching. Always check with OCSP responder. ## 1 Use OCSP cache caching (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP 7.1.0.0 and later ## SET OCSP_USE_CACHE 1 ## ## OCSP_CACHE_EXPIRY specifies the cache expiry in minutes. ## Valid values: 60 to 10080 (60 min to 7 days) with default 2880 (2 days). ## Note that OCSP response cache expiry uses nextUpdate value in OCSP response message. Only if nextUpdate is not present will the OCSP_CACHE_EXPIRY parameter value be used. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET OCSP_CACHE_EXPIRY 1 ## ################ PUSH INTERFACE SETTINGS ################# ## ## TPSLIST (Trusted Push Server List) specifies a list of URI authority components ## (optionally including scheme and path components) to be trusted. ## A URI received in a Push Request will only be used to obtain Push content ## if it matches one of these values. The list can contain up to 255 characters. ## Values are separated by commas without any intervening spaces. ## If the value of TPSLIST is null (the default), Push will be disabled. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0.1 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.2, R2.5 and later ## 16xx H.323 R1.0 and later ## SET TPSLIST 135.20.21.20,push.avaya.com,http://135.20.21.33:80,http://apps.avaya.com/push ## ## SUBSCRIBELIST specifies a list of URIs to which the telephone will send a Subscribe ## message (an HTTP GET for the URI with the telephone's MAC address, extension number, ## IP address and model number appended as query values) after the telephone successfully ## registers with a call server, or when a "subscribe" Push Request is received with ## a type attribute of "all". The list can contain up to 255 characters. ## Values are separated by commas without any intervening spaces. ## If the value of SUBSCRIBELIST is null (the default), Subscribe messages will not be sent ## after registration or in response to a Push Request with a type attribute of "all". ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0.1 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.2, R2.5 and later ## 16xx H.323 R1.0 and later ## SET SUBSCRIBELIST http://135.20.21.21/subscribe,http://push.avaya.com/clients ## ## PUSHCAP allows the modes of individual Push types supported by the telephone to be controlled. ## The value is a 3, 4 or 5 digit number, of which each digit controls a Push type and can have a ## value of 0, 1 or 2. A digit of 0 means that all Push requests will be rejected for that push type. ## A digit of 1 means that only Push requests with a mode of "barge" will be accepted for that push type. ## A digit of 2 means that Push requests with a mode of "barge" or "normal" will be accepted for that push type. ## The Push types controlled by each digit are as follows: ## 11111 ## ||||+- The rightmost digit controls top line Push requests. ## |||+-- The next digit to the left controls display (WML browser) Push requests. ## ||+--- The next digit to the left controls receive audio Push requests. ## |+---- The next digit to the left controls transmit audio Push requests. ## +----- The next digit to the left controls phonexml Push requests. ## J100 SIP R3.0.0.0 and later support 5-digit values (default is 00000), Only receive audio Push requests and top line Push requests are supported. Both "22222" and "00202" values have the same meaning. ## J169/J179 H.323 R6.7 and later support 3 and 4-digit values (default is 2222). ## J169/J179 SIP R1.5.0 support 4-digit values (default is 0000) ## 96x1 H.323 R6.0 and later support 3 and 4-digit values (default is 2222). ## 96x1 SIP R6.2 and later support 4-digit values (default is 0000). ## 96x1 SIP R6.0.x support 4-digit values (default is 0000) ## but not display Push, so valid values are 0000 through 2202. ## 96x0 H.323 R2.0 and later support 3 and 4-digit values (default is 2222). ## 96x0 SIP R2.2, R2.5 and later support 5-digit values (default is 00000). ## SET PUSHCAP 2222 ## ## PUSHPORT specifies the TCP port number to be used by the HTTP server in the telephone for Push. ## Valid values are 80 through 65535; the default value is 80 ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0.1 and later ## 96x0 H.323 R2.0 and later ## 96x0 SIP R2.2, R2.5 and later ## SET PUSHPORT 80 ## ################ HTTP/S WEB SERVER ################# ## ## ENABLE_WEBSERVER specifies whether the HTTP/S WEB Server is enabled or disabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; If the phone boots up in 3PCC environment and ENABLE_WEBSERVER is not explicitly set 0, it will be internally set to 1. ## This is to enable web server by default in 3PCC environments. SET ENABLE_WEBSERVER 1 ## ## WEBSERVER_ON_HTTP specifies whether HTTP access to the Web Interface is enabled or disabled. ## The WEB Server will be accessible using HTTP as long ENABLE_WEBSERVER and WEBSERVER_ON_HTTP are set to 1. ## The WEB Server will be accessible using HTTPS as long ENABLE_WEBSERVER is set to 1 AND (Identity certificate is installed in factory or ## using WEB/SCEP/PKCS12 file download). ## Value Operation ## 0 Web Server will not be accessible via HTTP ## 1 Web Server will be accessible via HTTP (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later SET WEBSERVER_ON_HTTP 1 ## ## WEB_HTTP_PORT specifies the HTTP port on which the Web Server running on the phone will be accessed using HTTP. ## Valid values are 80-65535. The default value is 80. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later ## SET WEB_HTTP_PORT 81 ## ## WEB_HTTPS_PORT specifies the HTTPS port on which the Web Server running on the phone will be accessed using HTTPS. ## Valid values are 443-65535. The default value is 443. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## SET WEB_HTTPS_PORT 444 ## ## FORCE_WEB_ADMIN_PASSWORD specifies the password to access the phone through Web as Administrator. ## From settings file, FORCE_WEB_ADMIN_PASSWORD will be used instead of WEB_ADMIN_PASSWORD (configured from the Web Interface). ## As long as FORCE_WEB_ADMIN_PASSWORD is configured in the Settings file, it will be used as the Web admin password. ## It will overwrite any password user might have configured from the Web Interface. ## Valid values are: 8 to 31 alphanumeric characters including upper, lower and special characters. ## Special characters allowed:~!@#$%^&*_-+=`|\(){}[]:;'<>,.?/. The default is "27238". ## Note: WEB_ADMIN_PASSWORD has no interaction with PROCPSWD or ADMIN_PASSWORD. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later SET FORCE_WEB_ADMIN_PASSWORD Avaya@123 ## ################# WML BROWSER SETTINGS ################### ## ## Note that if WMLHOME and WMLIDLEURI are set here, the web pages that they specify ## will be used by all telephones that support these parameters. ## If it is desired to use web pages that are customized to the display capabilities ## of a specific telephone model, the parameter should be set in the model-specific ## section for that telephone model located near the end of this file. ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLPROXY specifies zero or one address for an HTTP proxy server that will be used by the ## WML browser, and by the Weather and World Clock applications on the 9621, 9641 and 9670. ## The address can be in dotted-decimal (IPv4), or DNS name format, ## separated by commas without any intervening spaces. ## The value can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET WMLPROXY proxy.company.com ## ## WMLPORT specifies the TCP port number of the HTTP proxy server specified by WMLPROXY. ## Valid values are 0 through 65535. ## The default value for H.323 software is 8000. ## The default value for SIP software is 8080. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET WMLPORT 9000 ## ## WMLEXCEPT specifies zero or more IP addresses or domains for which ## the HTTP proxy server specified by WMLPROXY will not be used. ## The values are separated by commas without any intervening spaces. ## The value can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET WMLEXCEPT mycompany.com,135.20.21.20 ## ## WMLHELPSTAT specifies whether a web application help item will be displayed on the ## Home screen if no WML applications are administered and if the value of WMLHOME is null. ## Value Operation ## 0 A web application help item will not be displayed ## 1 A web application help item will be displayed (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later. ## 96x1 SIP R6.2 and later ## SET WMLHELPSTAT 0 ## ################# IDLE TIMER SETTINGS #################### ## ## BAKLIGHTOFF specifies the number of minutes of idle time after which the display backlight will be turned off. ## Phones with gray-scale displays do not completely turn backlight off, they set it to the lowest non-off level. ## Valid values are 0 through 999; the default value is 120 (2 hours). ## A value of 0 means that the display backlight will not be turned off automatically when the phone is idle. ## For ENERGY STAR® compliance on applicable phones, a value of 20 is recommended. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later. ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; default is changed to 10 minutes in R2.0.0.0. The range for K155 is 1-60. ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## B189 H.323 R1.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET BAKLIGHTOFF 60 ## ## HOMEIDLETIME specifies the number of minutes of idle time after which the Home screen will be displayed. ## A value of 0 means that the Home screen will not be displayed automatically when the phone is idle. ## This parameter is supported by: ## 9621 and 9641 H.323 R6.0 and later (valid values are 0 through 30; the default value is 10) ## 9621 and 9641 SIP R6.0 and later (valid values are 0 through 30; the default value is 10) ## 9670 H.323 R2.0 and later (valid values are 5 through 30; the default value is 10) ## J129 R1.0.0.0 and later (valid values are 0 through 30; the default value is 10) ## Note: The parameter is not supported by J179/J169 SIP phones. ## SET HOMEIDLETIME 5 ## ## SCREENSAVERON specifies the number of minutes of idle time after which the screen saver will be displayed. ## If an image file has been downloaded based on the SCREENSAVER (H.323), LOGOS and CURRENT_LOGO ## (for 96x0 R1.0 SIP and later, 96x1 R6.0 SIP and later and J169/J179 SIP R1.5.0) or SCREENSAVER_IMAGE ## (for J100 SIP R2.0 and later) parameters, it will be used as the screen saver. ## Otherwise, the built-in Avaya one-X(TM) screen saver will be used. ## Valid values are 0 through 999; the default value is 240 (4 hours). ## A value of 0 means that the screen saver will not be displayed automatically when the phone is idle. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J139 SIP R3.0.0.0 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later, but not supported by the 9610 ## 96x0 SIP R1.0 and later ## SET SCREENSAVERON 480 ## ## SCREENSAVER specifies the filename of a JPEG image to be used as a customized screen saver. ## Valid values are 0 through 32 ASCII characters; the default value is null (""). ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later; Supported format is JPEG file. Resolution is 480 x 800 pixels and color depth of 24 bits. ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R2.0 and later, but not supported by the 9610 ## SET SCREENSAVER filename ## ## SCREENSAVERURL specifies the URL content presented in screen saver mode. ## Zero to 255 ASCII characters; the default value is null (""). ## In case the SCREENSAVERURL parameter is configured and it includes a valid URL address, H1xx should display ## this URL when the H1xx is in screen saver mode. The URL can contain link to web pages and also to Video files. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET SCREENSAVERURL http://www.xyz.com/H1xxScreenSaver/ ## ## SCREENSAVER_IMAGE specifies a list of screensaver images. The default value is "". ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## Up to 5 background images are supported. Only jpeg/jpg files are supported. ## The maximum size of any jpeg file is 256 KB. The filenames are case insensitive. ## J169/J179 screen resolution is 320 pixels x 240 pixels. J179 color depth is 16 bits. ## J129 screen resolution is 128 pixels x 32 pixels. ## The files shall be stored in the same directory defined by HTTPDIR / TLSDIR. ## SET SCREENSAVER_IMAGE "screensaver_example1.jpg,screensaver_example2.jpeg" ## ## SCREENSAVER_IMAGE_DISPLAY specifies the administrator choice of screensaver image. ## The filename shall be one of the filenames listed in SCREENSAVER_IMAGE. ## If SCREENSAVER_IMAGE_SELECTABLE is set to 1 then the end user may override this setting. ## The default value is "". ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## SET SCREENSAVER_IMAGE_DISPLAY screensaver_example1.jpg ## ## SCREENSAVER_IMAGE_SELECTABLE specifies whether end users are allowed to choose screensaver images ## (and overrides administrator choice as configured using SCREENSAVER_IMAGE_DISPLAY parameter). ## Value Operation ## 0 End user is not allowed to choose screensaver image and will not see the screensaver image selection in the Settings -> Display menu. ## 1 End user is allowed to choose the screensaver image from the Settings -> Display menu (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## SET SCREENSAVER_IMAGE_SELECTABLE 0 ## ## BACKLIGHT_SELECTABLE specifies whether backlight timer will be determined per administrator (BAKLIGHTOFF) or user configuration. ## Value Operation ## 0 "Backlight timer" value (BAKLIGHTOFF) will be obtained from 46xxsettings. ## 1 "Backlight timer" value (BAKLIGHTOFF) can be set by user using "User Menu->Settings->Display" submenu (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J139 SIP R3.0.0.0 and later ## SET BACKLIGHT_SELECTABLE 0 ## ## WMLIDLETIME specifies the number of minutes of idle time after which ## the web page specified by the value of WMLIDLEURI will be displayed. ## If WMLIDLEURI is null, a web page will not be displayed when the phone is idle. ## On the 9610, WMLIDLETIME specifies the number of minutes of idle time after which the ## idle application (configured by IDLEAPP in the 9610 backup/restore file) will be displayed. ## Valid values are 1 through 999; the default value is 10. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET WMLIDLETIME 60 ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R1.0 and later, but not supported by the 9610 ## 96x0 SIP R2.0 and later ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############# PHONE LOCK SETTINGS (SIP ONLY) ############# ## ## ENABLE_PHONE_LOCK specifies whether a softkey (on the idle Phone screen) and ## a feature button will be displayed to allow the user to manually lock the phone. ## Value Operation ## 0 Disabled: Lock softkey and feature button will not be displayed (default) ## 1 Enabled: Lock softkey and feature button will be displayed ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; Please note that on J129 the Lock option appears ## in the main menu. There is no Lock softkey or feature button. ## Avaya Vantage Devices SIP R1.0.0.0 and later; Please note that ENABLE_PHONE_LOCK is used as enable/disable ## of lock screen. ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.5 and later ## H1xx SIP R1.0 and later, Please note that ENABLE_PHONE_LOCK is used on H1xx as enable/disable ## of lock screen. ## SET ENABLE_PHONE_LOCK 1 ## ## PHONE_LOCK_IDLETIME specifies the interval of idle time, in minutes, after which ## the phone will automatically lock if the value of ENABLE_PHONE_LOCK is 1. ## A value of 0 means that the phone will not lock automatically. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; valid values are 0 through 10080; the default value is 0. The parameter is supported ## no matter what is the ENABLE_PHONE_LOCK value is. ## J169/J179 SIP R1.5.0; valid values are 0 through 10080; the default value is 0. ## Avaya Vantage Devices SIP R1.0.0.0 and later; valid values are 1 through 10080; the default value is 60. Please note PHONE_LOCK_IDLETIME ## specifies the maximum interval of idle time, in minutes, allowed for user configuration (unless exchange policy enforces lower number). ## User can choose smaller value than this value in the settings application. By default, user choice is 5 minutes. ## 96x1 SIP R6.2 and later; valid values are 0 through 10080; the default value is 0. ## 96x1 SIP R6.0.x; valid values are 0 through 999; the default value is 0. ## 96x0 SIP R2.5 and later; valid values are 0 through 999; the default value is 0. ## H1xx SIP R1.0 and later; valid values are 1 through 10080; the default value is 60. Please note PHONE_LOCK_IDLETIME ## specifies the maximum interval of idle time, in minutes, allowed for user configuration (unless exchange policy enforces lower number). ## User can choose smaller value than this value in the settings application. By default, user choice is 5 minutes. ## SET PHONE_LOCK_IDLETIME 30 ## ## PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS specifies the number of failed attempts ## before the the device is lockout. The range is 0, 5-20. If 0, then no limit on number of failed attempts. ## Otherwise, it defines the number of failed attempts. The default value is 8. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; ## H1xx SIP R1.0 and later; ## SET PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS 10 ## ## LOCKSCREENURL specifies the URL content presented in lock screen mode. ## Zero to 255 ASCII characters; the default value is null (""). ## In case the LOCKSCREENURL parameter is configured and it includes a valid URL and in case the H1xx is registered ## to the SIP Controller and it is locked, then content of this URL shall be presented. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET LOCKSCREENURL http://www.xyz.com/H1xxlock/ ## ############# TRUST AGENTS SETTINGS (SIP ONLY) ############# ## ## TRUST_AGENTS_STAT specifies whether user can configure "Trust Agents" or not. ## Value Operation ## 0 Trust agents are disabled and user is not able to define trust agents. The menu "Trust agents" under Settings application --> security is hidden ## and all trust agents defined are disabled. ## 1 The user is given an option to enable/disable trust agents. The menu "Trust agents" under Settings application --> security is shown ## and by default all trust agents defined are disabled (default). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET TRUST_AGENTS_STAT 0 ## ## TRUST_AGENTS_SMARTLOCK_STAT specifies whether user can configure "Smart Lock Agent" or not. ## Value Operation ## 0 The "Smart Lock" menu under settings application --> security is not shown to the user and "Smart Lock" is disabled. ## 1 The "Smart Lock" menu in the settings application --> security is shown to the user and user can define "Smart Lock" and enable/disable "Smart Lock" (default). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET TRUST_AGENTS_SMARTLOCK_STAT 0 ## ## TRUST_AGENTS_AVAYA_SMARTLOCK_STAT specifies whether user can configure "Avaya Smart Lock Agent" or not. ## Value Operation ## 0 The "Avaya Smart Lock" menu under settings application --> security is not shown to the user and "Avaya Smart Lock" is disabled. ## 1 The "Avaya Smart Lock" menu in the settings application --> security is shown to the user and user can define "Avaya Smart Lock" and enable/disable "Avaya Smart Lock" (default). ## Note: User can enable only one Smart Lock ("Avaya Smart Lock" or "Android Smart Lock"). "Avaya Smart Lock" is mainly used to "unlock/lock" or "login and unlock/logout" ## according to Bluetooth pairing connection status of user's mobile phone. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.1.0.0 and later ## SET TRUST_AGENTS_AVAYA_SMARTLOCK_STAT 0 ## ############ CODEC AND RTP SETTINGS (SIP ONLY) ########### ## ## ENABLE_G711A specifies whether the G.711 a-law codec is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET ENABLE_G711A 0 ## ## ENABLE_G711U specifies whether the G.711 mu-law codec is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET ENABLE_G711U 0 ## ## ENABLE_G722 specifies whether the G.722 codec is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; the default value is 1. ## 96x1 SIP R6.2 and later; the default value is 1. ## 96x1 SIP R6.0.x; the default value is 0. ## 96x0 SIP R2.0 and later; the default value is 0. ## H1xx SIP R1.0 and later; the default value is 1. ## SET ENABLE_G722 1 ## ## ENABLE_G726 specifies whether the G.726 codec is enabled. ## Value Operation ## 0 Disabled (default for 96x0 R1.0) ## 1 Enabled (default for all other releases and models) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later; For IP office environment this parameter shall be set to 0 as G.726 is not supported by IP Office. ## SET ENABLE_G726 0 ## ## G726_PAYLOAD_TYPE specifies the RTP payload type to be used for the G.726 codec. ## Valid values are 96 through 127; the default value is 110. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET G726_PAYLOAD_TYPE 111 ## ## ENABLE_G729 specifies whether the G.729A codec is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled without Annex B support (default) ## 2 Enabled with Annex B support ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET ENABLE_G729 0 ## ## ENABLE_OPUS specifies whether the OPUS codec is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled WIDEBAND_20K (default value). ## 2 Enabled NARROWBAND_16K ## 3 Enabled NARROWBAND_12K ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later; ## Up to R2.0.0.0 (excluded) for IP office and 3PCC environments this parameter shall be set to 0 (As OPUS is supported in Avaya Aura environment only). ## R2.0.0.0 and later supports OPUS in all environments. ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; supported in both Aura and IPO environments. ## SET ENABLE_OPUS 0 ## ## OPUS_PAYLOAD_TYPE specifies the RTP payload type to be used for the OPUS codec. ## Valid values are 96 through 127; the default value is 116. ## This parameter is used when media offer is sent to the far end ## in an INVITE (or 200 OK when INVITE with no SDP is received). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.1 and later ## SET OPUS_PAYLOAD_TYPE 111 ## ## SEND_DTMF_TYPE specifies whether DTMF tones are sent in-band (as regular audio), ## or out-of-band (using RFC 2833 procedures). ## Value Operation ## 1 in-band ## 2 out-of-band (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET SEND_DTMF_TYPE 1 ## ## DTMF_PAYLOAD_TYPE specifies the RTP payload type to be used for RFC 2833 signaling. ## Valid values are 96 through 127; the default value is 120. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET DTMF_PAYLOAD_TYPE 121 ## ## SYMMETRIC_RTP specifies whether or not the telephone should discard ## received RTP/SRTP datagrams if their UDP Source Port number is not ## the same as the UDP Destination Port number that the telephone is ## including in RTP/SRTP datagrams intended for that endpoint. ## Value Operation ## 0 Ignore the UDP Source Port number in received RTP/SRTP datagrams. ## 1 Discard received RTP/SRTP datagrams if their UDP Source Port number ## does not match the UDP Destination Port number that the telephone is ## including in RTP/SRTP datagrams intended for that endpoint (default). ## This parameter is supported by: ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.1.0 and later (9608 and SIP9611 HW version 3 and higher) ## 96x1 SIP R6.0 and later (hardware version below 3). ## H1xx SIP R1.0 and later ## 96x0 SIP R2.4 and later ## SET SYMMETRIC_RTP 0 ## ############ VIDEO SETTINGS ########### ## ## ENABLE_VIDEO specifies whether video is enabled or disabled. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.0.0.1 and later ## Avaya Equinox 3.1.2 and later ## H1xx SIP R1.0 and later ## SET ENABLE_VIDEO 0 ## ## VIDEO_H264_PROFILE specifies the maximal profile level that can be used by the device. ## Value Operation ## 66 Baseline profile ## 100 High profile and Baseline profile (Default) ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_H264_PROFILE 66 ## ## VIDEO_PAYLOAD_LENGTH specifies the video packets payload length (bytes) ## Valid values are 0, 1200 through 1460; where 0 means that the video packets payload length is calculated ## according to MTU_SIZE parameter. If MTU_SIZE is 1500 bytes then video payload length will be: ## 1460 == 1500 Bytes (Ethernet) - 20 (IP) - 8 (UDP) - 12 (RTP). In similar way if MTU_SIZE is 1496 bytes ## then video payload length will be: 1456. ## The default value is 0. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_PAYLOAD_LENGTH 1460 ## ## PINHOLE_KEEPALIVE_INTERVAL specifies the maximal time in seconds between consecutive video RTP packets. ## Valid values are 0-60; where 0 means no RTP keepalives; 1-60 refers to keepalive interval in seconds. ## The default is 15 seconds. ## If the timer expires the device will send out RTP packet with PT=0 and no payload in order to keep ## NAT/Firewall pinhole open. The use case is when video is MUTE and the device is behind NAT/firewall device. ## Keepalives are sent on video RTP ports only (not video RTCP). Audio RTP packets are kept sending even if there is audio MUTE. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET PINHOLE_KEEPALIVE_INTERVAL 60 ## ## ENABLE_FIR specifies whether key frame requests are supported using RTCP FIR (Full Intra Requests ## according to RFC 5104). ## Value Operation ## 0 RTCP FIR is not SDP negotiated with remote peer ## 1 RTCP FIR is SDP negotiated with remote peer (default). Only if both peers support ## RTCP FIR, then RTCP FIR messages will be generated and received. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET ENABLE_FIR 0 ## ## ENABLE_PLI specifies whether key frame requests are supported using RTCP PLI (Picture Loss Indication ## according to RFC 4585). ## Value Operation ## 0 RTCP PLI is not SDP negotiated with remote peer ## 1 RTCP PLI is SDP negotiated with remote peer (default). Only if both peers support ## RTCP PLI, then RTCP PLI messages will be generated and received. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET ENABLE_PLI 0 ## ## ENABLE_TMMBR specifies whether TMMBR (Temporary Maximum Media Stream Bit Rate Requests) RTCP requests ## (according to RFC 5104) are sent to remote peer for bit rate adaptation and whether the device ## responds to TMMBR RTCP requests received. ## Value Operation ## 0 RTCP TMMBR is not SDP negotiated with remote peer ## 1 RTCP TMMBR is SDP negotiated with remote peer (default). Only if both peers support ## RTCP TMMBR, then RTCP TMMBR messages will be generated and received. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET ENABLE_TMMBR 0 ## ## DYNAMIC_VIDEO_SIZE_REQUEST specifies whether the device notifies the other side that the local video window ## size has been changed so it can change the transmitted video accordingly. ## Value Operation ## 0 Disabled ## 1 TMMBR - the notification will be done by changing the incoming video bandwidth ## request using RTCP TMMBR (default) ## The parameter is only applicable if ENABLE_TMMBR is set to 1. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET DYNAMIC_VIDEO_SIZE_REQUEST 0 ## ## DYNAMIC_VIDEO_SIZE_REQUEST_DELAY specifies the amount of time (in seconds) the device will wait before ## asking the remote party to reduce resolution to match a newly selected video window size. ## The range is 1-600. The default value is 20 seconds. ## The parameter is only applicable if ENABLE_TMMBR and DYNAMIC_VIDEO_SIZE_REQUEST are set to 1. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET DYNAMIC_VIDEO_SIZE_REQUEST_DELAY 60 ## ## VIDEO_MAX_RX_RESOLUTION specifies the maximum video resolution that the device will request from the other side. ## Value Operation ## 4 480p ## 5 720p (1280 x 720) ## 6 1080p (1920x1080), Default ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_MAX_RX_RESOLUTION 5 ## ## VIDEO_MAX_TX_RESOLUTION specifies the maximum video resolution that the device encodes and sends. ## This value is enforced locally without signaling to the remote party. ## Value Operation ## 1 180p (320 x 180) ## 2 240p ## 3 360p (640 x 360) ## 4 480p ## 5 720p (1280 x 720) ## 6 1080p (1920x1080), Default ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_MAX_TX_RESOLUTION 5 ## ## VIDEO_MAX_RX_BANDWIDTH specifies the overall SDP requested bandwidth for Video RTP ## including IP and UDP overheads (but not Ethernet). The range is 80-4300 kbps. ## The default value is 4300 kbps. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_MAX_RX_BANDWIDTH 1000 ## ## VIDEO_MAX_TX_BANDWIDTH specifies the overall bandwidth consumed by transmitted RTP for video ## including IP and UDP overheads (but not Ethernet). The range is 80-4300 kbps. ## The default value is 2500 kbps. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET VIDEO_MAX_TX_BANDWIDTH 1000 ## ## VIDEO_CALL_DISPLAY_MODE specifies whether video call will be presented automatically on external screen or on H175 built-in screen. ## Value Operation ## 0 When a video call is established (no matter whether user initiate a video call, escalate from audio to video call or answer a video call), ## present the video on H175 Built in screen. ## 1 When a video call is established (no matter whether user initiate a video call, escalate from audio to video call or answer a video call), ## present the video on H175 external screen ONLY if H175 is connected to external screen, else H175 built-in screen will be used (default). ## Note: This parameter is also stored/retrieved to/from PPM. Configuration of this parameter using the settings file is useful for initial ## configuration case only where such value is not stored yet in PPM. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET VIDEO_CALL_DISPLAY_MODE 0 ## ## VIDEO_MAX_BANDWIDTH_ANY_NETWORK specifies the maximum bandwidth used for video calls. ## The value range is 0 to 10,000 kbps. Default value is 1280 kbps. 0 means video is blocked. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET VIDEO_MAX_BANDWIDTH_ANY_NETWORK 1000 ## ############ CAMERA SETTINGS ########### ## ## CAMERA_ANTIFLICKER_POWERLINE_FRQ specifies the frequency in Hz of the electrical power lines. ## The camera anti-flicker filter cancels artifacts caused by Florescent lights resonating ## at the power-line frequency. The power-line frequency is dependent on where the phone is deployed, ## and is either 50Hz or 60Hz. The automatic mode means that anti-flicker frequency is adjusted ## automatically according to the COUNTRY settings file parameter (60Hz when COUNTRY is undefined). ## Value Operation ## 0 Auto (default) ## 1 50 Hz ## 2 60 Hz ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET CAMERA_ANTIFLICKER_POWERLINE_FRQ 60 ## ############ EXTERNAL MONITOR SETTINGS ########### ## ## ENFORCE_DVI specifies whether DVI is enforced when PC display passes through H175. ## Value Operation ## 0 When PC display passes through H175, DVI is NOT enforced. As a result, the external monitor EDID information provided by H175 ## to the PC will contain both HDMI and DVI resolutions. HDMI resolutions in the EDID are limited to 720p. ## 1 When PC display passes through H175, DVI is enforced (Default). As a result, the external monitor EDID information provided to ## the PC will only contain DVI resolutions. This should be the preferred operating mode so that output video resolutions generated by H175 ## are not limited to HDMI 720p. In this mode the resulting video output signal will set to the DVI format. ## Note: This parameter has no affect when either the PC video output or external monitor connected to H175 are DVI. ## Note: This parameter has no affect when H175 is not connected to a PC video output. ## This parameter is supported by: ## H1xx SIP R1.0.0.1 and later ## SET ENFORCE_DVI 0 ## ## CLONE_DISPLAY specifies whether clone internal display to external monitor ## Value Operation ## 0 HDMI Pass through, PC screen is pass through the device (default) ## 1 H175 internal display is clone to the external monitor ## Note: This parameter is also stored/retrieved to/from PPM. Configuration of this parameter using the settings file is useful for initial ## configuration case only where such value is not stored yet in PPM. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET CLONE_DISPLAY 1 ## ################## OTHER SIP-ONLY SETTINGS ################# ## ## PHNMUTEALERT_BLOCK specifies whether the Mute Alert feature will be Blocked or Unblocked. ## Value Operation ## 0 Unblocked ## 1 Blocked (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0.1 and later. ## SET PHNMUTEALERT_BLOCK 1 ## ## MATCHTYPE specifies how a calling party number is compared to the numbers ## in the user's Contacts to obtain a name to display for the incoming call. ## Value Operation (for 96x1 SIP R6.2 to R7.0 (excluded), 96x0 R2.6.5 and later) ## 0 The Contact name is displayed if the rightmost 4 digits of the calling ## party number match the rightmost 4 digits of a Contacts number (default) ## 1 The Contact name is displayed if the entire calling party number ## exactly matches the all of the digits in a Contacts number ## Value Operation (for 96x1 SIP R7.0 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later) ## 0 The Contact name is displayed if the entire calling/called party number exactly matches ## the number stored in the contact (ELD rules are applied) (default) ## 1 The Contact name is displayed if all the digits of the shorter number (contacts, calling/called party number) ## match to the rightmost digits of the longer number (contacts, calling/called party number). ## 2 The Contact name is displayed if at least 4 rightmost digits of the calling/called ## party number match the rightmost 4 digits of a Contacts number ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.2 and later ## 96x0 SIP R2.6.5 and later. ## SET MATCHTYPE 0 ## ## ENABLE_HOLD_BUTTON specifies whether a Hold softkey will be displayed during an active call. ## Value Operation ## 0 A Hold softkey will not be displayed ## 1 A Hold softkey will be displayed (default) ## This parameter is supported by: ## 96x0 SIP R2.6.7 and later ## SET ENABLE_HOLD_BUTTON 0 ## ## USE_QUAD_ZEROES_FOR_HOLD specifies how Hold will be signaled in SDP. ## Value Operation ## 0 "a=directional attributes" will be used (default) ## 1 "c=0.0.0.0" will be used ## This parameter is supported by: ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.0 and later ## SET USE_QUAD_ZEROES_FOR_HOLD 1 ## ## SOFTKEY_CONFIGURATION specifies which feature will show up on which softkey on the J129 phone screen. Does not apply to other J100 models. ## The features are defined as follows: ## 0 = Redial ## 1 = Contacts ## 2 = Emergency ## 3 = Recents ## 4 = Voicemail ## The default is "0,1,2". i.e. default softkeys are Redial, Contacts, Emerg. ## Note: RULES: ## If a value is not presented then the softkey is blank ## e.g. SOFTKEY_CONFIGURATION 0,,2 => Redial, Blank, Emerg ## If a value is outside the range then the softkey is blank ## e.g. SOFTKEY_CONFIGURATION 0,1,7 => Redial,Contacts, Blank ## e.g. SOFTKEY_CONFIGURATION 0,&GGI^,2 => Redial, Blank, Emerg ## If there are not enough values in the range then the remaining softkeys will be blank ## e.g. SOFTKEY_CONFIGURATION 4,3 = Voicemail,Recents, Blank ## ADDITIONAL NOTES: Even if PHNEMERGNUM is defined the EMERG softkey must be defined. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J129 only) ## SET SOFTKEY_CONFIGURATION 1,2,3 ## ###################### SIG SETTING ###################### ## ## SIG specifies the type of software to be used by the telephone by ## controlling which upgrade file is requested after a power-up or a reset. ## Value Operation ## 0 Download the upgrade file for the same signaling protocol ## that is supported by the current software (default) ## 1 Download 96x1Hupgrade.txt (for H.323 software) ## 2 Download 96x1Supgrade.txt (for SIP software) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## SET SIG 0 ## ############### ETHERNET INTERFACE SETTINGS ################ ## ## PHY1STAT specifies the speed and duplex settings for the Ethernet line interface. ## Valid values are 1 through 6; the default value is 1. ## Value Operation ## 1 auto-negotiate ## 2 10Mbps half-duplex ## 3 10Mbps full-duplex ## 4 100Mbps half-duplex ## 5 100Mbps full-duplex ## 6 1Gbps full-duplex if supported by hardware, otherwise auto-negotiate ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (values 1-5 only), J139 SIP R3.0.0.0 and later (values 1-5 only) ## J169/J179 H.323 R6.7 and later ## H1xx SIP R1.0 and later (values 1-5 only) ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later (values 1-5 only) ## B189 H.323 R1.0 and later ## 96x1 SIP R6.0.x ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET PHY1STAT 1 ## Note: The parameter is permanently configured to "Auto-Negotiate" on Avaya Vantage SIP R1.1.0.0 and later. ## ## PHY2STAT specifies the speed and duplex settings for the secondary (PC) Ethernet interface. ## Valid values are 0 through 6; the default value is 1. ## Value Operation ## 0 disabled ## 1 auto-negotiate ## 2 10Mbps half-duplex ## 3 10Mbps full-duplex ## 4 100Mbps half-duplex ## 5 100Mbps full-duplex ## 6 1Gbps full-duplex if supported by hardware, otherwise auto-negotiate ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (values 0-5 only), J139 SIP R3.0.0.0 and later (values 1-5 only) ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch (values 0-1 only), All K155 devices have embedded Ethernet switch (values 0-1 are supported only). ## H1xx SIP R1.0 and later (values 0-5 only) ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.2 and later (values 0-5 only) ## 96x1 SIP R6.0.x ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET PHY2STAT 1 ## ## PHY2_AUTOMDIX_ENABLED specifies whether auto-MDIX is enabled on PHY2. ## Valid values are 0 through 1; the default value is 1. ## Value Operation ## 0 auto-MDIX is disabled ## 1 auto-MDIX is enabled (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later for K165/K175 models with embedded Ethernet switch; All K155 devices have embedded Ethernet switch. ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.3 and later ## 96x1 SIP R6.3 and later ## SET PHY2_AUTOMDIX_ENABLED 1 ## ## EEESTAT controls whether Energy-Efficient Ethernet (802.3az) is enabled on PHY1 and PHY2. ## Value Operation ## 0 EEE is disabled ## 1 EEE is enabled (default). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later (J129 only) ## SET EEESTAT 0 ## ############### POWER OVER ETHERNET SETTINGS ################ ## ## ASSUME_SP_POE specifies whether Single port PoE injector is connected to Video Collaboration Station. ## Value Operation ## 0 single port PoE injector is not connected to the Video Collaboration Station.(default) ## 1 single port PoE injector is connected to the Video Collaboration Station. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET ASSUME_SP_POE 1 ## ## SP_POE_POWER specifies how much power is provided when Video Collaboration Station is connected to Single port PoE injector. ## The range is 15-26 watts where 20 is the default. ## This parameter is supported by: ## H1xx SIP R1.0 and later ## SET SP_POE_POWER 21 ## ############# LOCAL PROCEDURE ACCESS SETTINGS ############ ## ## PROCSTAT specifies whether local (craft) procedures can be used to configure the telephone. ## Value Operation ## 0 Local procedures can be used (default) ## 1 Local procedures cannot be used ## Note: Be very careful before setting PROCSTAT to 1 ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET PROCSTAT 1 ## ## PROCPSWD specifies an access code for access to local (craft) procedures. ## Valid values contain 0 through 7 ASCII numeric digits. ## The default value is 27238 (CRAFT) unless indicated otherwise below. ## A null value implies that an access code is not required for access. ## Note: Setting this parameter via CM (for H.323) or PPM (for SIP) is more secure ## because this file can usually be accessed and read by anyone on the network. ## Setting the value in this file is intended primarily for configurations with ## versions of telephone or server software that do not support setting this ## value from the server. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (must contain at least 4 digits ## else default value 27238 is used) ## J169/J179 H.323 R6.7 and later (must contain at least 4 digits, ## else default value 27238 is used) ## Avaya Vantage Devices SIP R1.0.0.0 and later (must contain at least 4 digits) ## 96x1 H.323 R6.0 and later (must contain at least 4 digits for R6.2.4 and later, ## else default value 27238 is used) ## 96x1 SIP R6.0 and later (must contain at least 4 digits for R6.3 and later ## else default value 27238 is used) ## H1xx SIP R1.0 and later (must contain at least 4 digits else default value ## 27238 is used) ## B189 H.323 R1.0 and later (must contain at least 4 digits for R1.0 and later ## else default value 27238 is used) ## 96x0 H.323 R1.0 and later (default is null ("") prior to R1.2, must contain at ## least 4 digits for R3.2.1 and later else default value ## 27238 is used) ## 96x0 SIP R1.0 and later (must contain at least 4 digits for R2.6.10 and later ## else default value 27238 is used) ## 16xx H.323 R1.0 and later (default is null ("") prior to R1.3.3, and ## must contain at least 4 digits for R1.3.3 and later) ## SET PROCPSWD 572958 ## ## ADMIN_PASSWORD specifies a complex access code for access to local (craft) procedures. ## Valid values contain 6 and 31 alphanumeric characters including upper, lower and special characters. ## The default value is 27238 which implies that PROCPSWD is used as access code for access to local (craft) procedures. ## If ADMIN_PASSWORD length is less than 6 or greater than 31, the parameter is treated as not defined. ## If ADMIN_PASSWORD is configured, then PROCPSWD is ignored. ## The special characters supported are: ~!@#$%^&*_-+=`|\(){}[]:;'<>,.?/. " is not supported. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later; the default value is "". ADMIN_PASSWORD is not applied on Boot Recovery Menu (BRM). ## 96x1 SIP R7.1.0.0 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later; default value is "". ## Note: For Avaya Vantage Basic Application and Avaya Vantage Devices, if ADMIN_PASSWORD is not configured and PROCPSWD is not configured, then the ## local (craft) procedures are not accessible. ## Note: The parameter is also used by "Avaya Vantage Basic Application" to allow administrator ## to unpin/pin the application when PIN_APP is defined to "Avaya Vantage Basic Application" package name. ## SET ADMIN_PASSWORD ComPlexPSWD12?! ## ## ADMIN_LOGIN_ATTEMPT_ALLOWED specifies the number of failed attempts for entering the access code (PROCPSWD or ADMIN_PASSWORD) ## before the local (craft) procedures will be locked for a period specified by ADMIN_LOGIN_LOCKED_TIME. ## Valid values are 1 to 20, default 10. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET ADMIN_LOGIN_ATTEMPT_ALLOWED 11 ## ## ADMIN_LOGIN_LOCKED_TIME specifies the time in minutes that local (craft) procedures are locked once the ## number of failed attempts for entering the access code (PROCPSWD or ADMIN_PASSWORD) is reached. ## Valid values 5 min to 1440 min, default 10 min. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET ADMIN_LOGIN_LOCKED_TIME 2 ## ## MUTECRAFTOPTIONS specifies whether CRAFT options can be invoked when pressing MUTE button in case of ## off-hook idle state (I.e. the telephone must have all call appearances in either In-Use or Idle call states ## and switchhook is off-hook) or MUTE button applies to the audio stream. ## Value Operation ## 0 CRAFT options can be invoked when pressing MUTE button in case of off-hook idle state (default) ## 1 MUTE button applies to the audio stream when pressing MUTE button in case of off-hook idle state ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6 and later ## SET MUTECRAFTOPTIONS 1 ## ##################### SNMP SETTINGS ###################### ## ## SNMPSTRING specifies a security string that must be included in SNMP query messages ## for the query to be processed. ## Valid values contain 0 through 32 ASCII alphanumeric characters. ## The default value is null ("") unless indicated otherwise below. ## A null value results in SNMP being disabled. ## Note: Setting this parameter via CM (for H.323) or PPM (for SIP) is more secure ## because this file can usually be accessed and read by anyone on the network. ## Setting the value in this file is intended primarily for configurations with ## older versions of telephone, CM or PPM software that do not support setting ## this value from the server. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET SNMPSTRING mystring ## ## SNMPADD specifies a list of source IP addresses from which SNMP query messages ## will be accepted and processed. ## Addresses can be in dotted-decimal (IPv4), colon-hex (IPv6, if supported), or ## DNS name format, separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET SNMPADD 192.168.0.22,192.168.0.23 ## ####### LINK LAYER DISCOVERY PROTOCOL (LLDP) SETTINGS ###### ## ## LLDP_ENABLED specifies whether LLDP is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled ## 2 Enabled, but only begin transmitting if an LLDP frame is received (default) ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; the default is 1. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; the default is 1. ## 96x0 SIP R2.0 and later ## Note that the following do NOT support the LLDP_ENABLED parameter, ## but they always operate consistent with a value of 2 above: ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.2 and later ## 16xx H.323 R1.2 and later ## SET LLDP_ENABLED 1 ## ## LLDP_XMIT_SECS specifies the rate in seconds at which LLDP messages will be transmitted. ## Valid values are 1 through 3600; the default value is 30. ## The primary intent of this parameter is to allow an SSO application to discover ## the telephone faster, since SSO uses LLDP for discovery, but this parameter ## controls the LLDP transmission rate on both Ethernet interfaces. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## B189 H.323 R1.0 and later ## SET LLDP_XMIT_SECS 10 ## ######## SYNCHRONIZED STATE OPERATION (SSO) SETTINGS ####### ## ## SSO_ENABLED specifies whether SSO will be enabled or disabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_ENABLED 1 ## ## SSO_REGISTERED_MODE specifies what the telephone does if it receives a ## registration request from an SSO application when it already registered. ## Value Operation ## 1 Accept the request if the provided credentials match the credentials ## that were used to establish the existing registration, otherwise, ## unregister and attempt to register using the new credentials (default) ## 2 Accept the request only if the provided credentials match the credentials ## that were used to establish the existing registration ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_REGISTERED_MODE 2 ## ## SSO_LOCK_SYNC specifies whether the telephone will attempt to lock and unlock its ## user interface if it receives a request to lock or unlock from the SSO application. ## Value Operation ## 0 Ignores SSO requests to lock and unlock ## 1 Attempts to lock and unlock based on SSO requests (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_LOCK_SYNC 0 ## ## SSO_DISCONNECT_ACTION specifies what the telephone does if the link to the PC drops ## while an SSO connection is active ## Value Operation ## 1 Unregisters after invoking each FAC specified by SSO_DISCONNECT_FACS (default) ## 2 Locks the user interface ## 3 No action is taken ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_DISCONNECT_ACTION 2 ## ## SSO_DISCONNECT_FACS specifies a list of Feature Access Codes (FACs) to be invoked ## as determined by the value of SSO_DISCONNECT_ACTION above. ## FACs are separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_FACS *69,*35,*22 ## ## SSO_CLIENT_CERT specifies whether the telephone will request a client ## identity certificate during the TLS handshake for an SSO connection. ## Value Operation ## 0 A client certificate will not be requested (default) ## 1 A client certificate will be requested and authenticated ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET SSO_CLIENT_CERT 1 ## ################## EVENT LOGGING SETTINGS ################## ## ## SYSLOG_ENABLED enable or disable sending Syslog messages. ## Value Operation ## 0 Sending Syslog messages is disabled (default) ## 1 Sending Syslog messages is enabled ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET SYSLOG_ENABLED 1 ## ## LOGSRVR specifies one address for a syslog server ## in dotted-decimal (IPv4), colon-hex (IPv6, if supported), or DNS name format. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## H1xx SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET LOGSRVR 192.168.0.15 ## ## Note that different parameters are used to specify the severity levels of events ## logged for H.323 vs. SIP, with different default values, as described below. ## ## SYSLOG_LEVEL specifies the severity level of syslog messages. ## Events with the selected severity level and above will be logged ## (note that lower numeric severity values correspond to higher severity levels). ## Value Operation ## 3 Error, Critical, Alert and Emergency events are logged ## 4 Warning, Error, Critical, Alert and Emergency events are logged (Default) ## 5 Notice, Warning, Error, Critical, Alert and Emergency events are logged ## 6 Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged ## 7 Debug, Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET SYSLOG_LEVEL 4 ## ## LOCAL_LOGS_ENABLED enable or disable local logging storage. ## Value Operation ## 0 Local logging is disabled ## 1 Local logging is enabled (default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET LOCAL_LOGS_ENABLED 0 ## ## LOGLOCAL specifies the severity levels of events logged in the ## endptRecentLog, endptResetLog and endptStartupLog objects in the SNMP MIB. ## Events with the selected severity level and above will be logged ## (note that lower numeric severity values correspond to higher severity levels). ## Value Operation ## 0 Logging to the MIB is disabled ## 1 Emergency events are logged ## 2 Alert and Emergency events are logged ## 3 Critical, Alert and Emergency events are logged ## 4 Error, Critical, Alert and Emergency events are logged ## 5 Warning, Error, Critical, Alert and Emergency events are logged ## 6 Notice, Warning, Error, Critical, Alert and Emergency events are logged ## 7 Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged (default) ## 8 Debug, Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged ## Warning: A setting of 8 can impact the performance of the telephone due to the number of events generated. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x0 H.323 R1.0 and later ## 96x1 H.323 R6.0 and later ## B189 H.323 R1.0 and later ## 16xx H.323 R1.0 and later ## SET LOGLOCAL 5 ## ## LOCAL_LOG_LEVEL specifies the severity levels of events logged in the ## endptRecentLog, endptResetLog and endptStartupLog objects in the SNMP MIB. ## Events with the selected severity level and above will be logged ## (note that lower numeric severity values correspond to higher severity levels). ## Value Operation ## 0 Emergency events are logged ## 1 Alert and Emergency events are logged ## 2 Critical, Alert and Emergency events are logged ## 3 Error, Critical, Alert and Emergency events are logged (default) ## 4 Warning, Error, Critical, Alert and Emergency events are logged ## 5 Notice, Warning, Error, Critical, Alert and Emergency events are logged ## 6 Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged ## 7 Debug, Informational, Notice, Warning, Error, Critical, Alert and Emergency events are logged ## Warning: A setting of 7 can impact the performance of the telephone due to the number of events generated. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or 1.1.0.0), J169/J179 SIP R1.5.0 and J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; No SNMP support in R1.0.0.0. This parameter affects local log files stored on the device. Values 3-7 are supported. ## 96x0 SIP R1.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; No SNMP support in R1.0. This parameter affects local log files stored on the device. ## SET LOCAL_LOG_LEVEL 3 ## ## LOG_CATEGORY specifies a list of categories of events to be logged via syslog and locally. ## This parameter must be specified to log events below the Error level. ## The list can contain up to 255 characters. The default is "". ## Category names are separated by commas without any intervening spaces. ## See Administrator's guide for additional detail. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; the default is "ALL" which implies all categories. ## 96x0 SIP R1.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later; the default is "ALL" which implies all categories. New categories for H1xx compare to ## 96x1 SIP: "ANDROID" and "KERNEL". ## SET LOG_CATEGORY DHCP,NETMGR,AUDIO ## ## LOG_VERBOSITY defines whether or not the verbose logging is enabled or disabled. ## Value Operation ## 0 "Info" log messages are collected (default) ## 1 "Debug" log messages are collected (for debugging purposes). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; the parameter will take effect after reboot. ## Note: The local/remote logging level on Avaya Vantage shall also be configured to "Debug" or "Notice" (for "Info" messages) in order to capture the relevant messages ## from the Avaya Vantage Basic application and Avaya Equinox. ## SET LOG_VERBOSITY 1 ## ## SUPPORTEMAIL defines the default E-mail address to send diagnostic logs. The default value is "". ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.1.0.0 and later; used when sending the debug or audio report to an email application. ## Avaya Equinox 3.1.2 and later ## SET SUPPORTEMAIL support@company.com ## ## ANALYTICSENABLED defines whether to allow data collection by Avaya using Google Analytics on behalf of the administrator's user community or not. ## Value Operation ## 0 Do not allow collection by Avaya using Google Analytics on behalf of the administrator's user community. ## 1 Allow collection by Avaya using Google Analytics on behalf of the administrator's user community (default). ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.1.0.0 and later ## Avaya Equinox 3.1.2 and later ## SET ANALYTICSENABLED 0 ## ######### DEBUGGING SETTINGS (96x1 H.323 only) ########### ## ## LOGTOFILE specifies whether logging of optional debug messages ## to an internal file will be enabled or disabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2.3 and later ## B189 H.323 R1.0 and later ## SET LOGTOFILE 1 ## ################### AUDIO DEBUG RECORDING ################## ## ## ENABLE_RECORDING specifies whether audio debug recording is enabled for users. ## Value Operation ## 0 Audio debug recording is disabled (default) ## 1 Audio debug recording is enabled ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.1.0.0 and later (controls whether audio recording is enabled or disabled as part of the audio report). ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.3 and later ## H1xx SIP R1.0 and later ## SET ENABLE_RECORDING 1 ## ## WARNING_FILE specifies the file name or URL for a custom single-channel WAV file ## coded in ITU-T G.711 u-law or A-law PCM with 8-bit samples at 8kHz to be used ## as a call recording warning instead of the built-in English warning. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.3 and later ## SET WARNING_FILE "Warning.wav" ## ############## SECURE SHELL (SSH) SETTINGS ############### ## ## Note: The SSH server on the endpoints is used by Avaya Services only for debugging purposes only. ## The SSH server supports only Avaya Services Logins ("craft" and "sroot"). ## By enabling Avaya Services Logins you are granting Avaya access to your endpoints. ## This is necessary required to maximize the performance and value of your Avaya ## support entitlements, allowing Avaya to resolve product issues in a timely manner. ## In addition to enabling the Avaya Logins, the Avaya Product that the endpoints register with must be registered ## using the Avaya Global Registration Tool (GRT, see https://grt.avaya.com) to be eligible for Avaya remote connectivity. ## Please see the Avaya support site (support.avaya.com/registration) for additional information for registering products ## and establishing remote access and alarming. ## By disabling Avaya Services Logins you are preventing Avaya access to ## your endpoints. This is not recommended, as it can impacts Avaya’s ability to provide ## support for the product. Unless the customer is well versed in managing the product ## themselves, Avaya Services Logins should not be disabled. ## The access to the SSH server is protected by ASG (Legacy authentication algorithm) ## or EASG (new authentication algorithm). Enhanced Access Security Gateway (EASG) provides a more secure authentication ## compared to ASG for SSH server access. Endpoints that support EASG are no longer support ASG. ## EASG is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ("craft" only) ## J169/J179 H.323 R6.7 and later ("craft" only) ## Avaya Vantage Devices SIP R1.0.0.0 and later ("craft" and "sroot"). ## 96x1 SIP R7.1.0.0 and later ("craft" only) ## ## SSH_ALLOWED specifies whether SSH is supported. ## Value Operation ## 0 Disabled ## 1 Enabled ## 2 Configured using local craft procedure - the SSH server can be enabled or disabled from local craft procedure. ## When this mode is configured, then by default the SSH server is disabled. ## The default of 96x1 H.323 R6.2 up to 6.4 (not included) is 0 (disabled). The default value for 96x1 H.323 6.4 and later is 2. ## The default of 96x1 SIP R6.2 and later is 0 (disabled). ## The default of B189 is 0 (disabled). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (values 0-2), the default is 0. ## J169/J179 H.323 R6.7 and later (values 0-2) ## J169/J179 SIP R1.5.0 (values 0-1) ## Avaya Vantage Devices SIP R1.0.0.0 and later (values 0-1, default value is 0) ## 96x1 H.323 R6.2 and later (values 0-1), value 2 is added in R6.4 and later. ## 96x1 SIP R6.2 and later (values 0-1) ## B189 H.323 R1.0 and later (values 0-1) ## H1xx SIP R1.0 and later (values 0-1, default==0) ## SET SSH_ALLOWED 1 ## ## SSH_ROOT_ALLOWED specifies whether SSH root access is enabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET SSH_ROOT_ALLOWED 1 ## ## SSH_BANNER_FILE specifies the file name or URL for a custom SSH banner file. ## If the value is null, a default English banner will be used for SSH. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x1 H.323 R6.2 and later ## 96x1 SIP R6.2 and later ## B189 H.323 R1.0 and later ## H1xx SIP R1.0 and later ## SET SSH_BANNER_FILE http://security.myco.com/files/SSH-Banner.txt ## ## SSH_IDLE_TIMEOUT specifies the number of minutes of inactivity ## after which an SSH connection will be terminated ## Valid values are 0 through 32767; the default value is 10. ## A value of 0 means that the connection will not be terminated due to inactivity. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## 96x1 H.323 R6.2 and later ## 96x1 SIP R6.2 and later ## B189 H.323 R1.0 and later ## H1xx SIP R1.0 and later ## SET SSH_IDLE_TIMEOUT 30 ## ## EASG_SITE_CERTS specifies list of EASG site certificates which are used by ## technicians when they don't have access to the Avaya network to generate ## EASG responses for SSH login. ## 0 to 255 ASCII characters: zero or more file names or URLs, separated by commas ## without any intervening spaces ## The default is "". ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later ## 96x1 SIP 7.1.0.0 and later ## SET EASG_SITE_CERTS "mySiteCert.p7b" ## ## EASG_SITE_AUTH_FACTOR specifies Site Authentication Factor code associated with ## the EASG site certificate being installed. ## Valid value: a 10 to 20 character alphanumeric string ## Default value: "" ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later ## 96x1 SIP 7.1.0.0 and later ## SET EASG_SITE_AUTH_FACTOR "avaya12345abcd" ## ## CERT_WARNING_DAYS_EASG specifies how many days before the expiration of ## EASG product certificate that a warning should first appear on the phone ## screen. Syslog message will be generated as well. ## Valid values are: 90-730, default is 365 days. ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 SIP 7.1.0.0 and later ## SET CERT_WARNING_DAYS_EASG 100 ## ############## ANDROID DEBUG BRIDGE (ADB) SETTINGS ############### ## ## ADBSTAT specifies whether ADB for developers use is disabled without an option to enable it or whether users/administrators can enable/disable it. ## Value Operation ## 0 ADB is disabled and "ADB mode" field in the "Developer options" menu in the settings application is NOT shown to the user. ## 1 "ADB mode" field in the settings application is shown to the users/administrator and users/administrator can enable/disable it (by default, disable). ## This is the default value. ## Note: As ADB is not secured protocol, it is recommended to disable it. ADB shall only be used for applications development on Avaya Vantage devices. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET ADBSTAT 0 ## ######## Enhanced Debugging Capabilities Support ####### ## ## AUTHCTRLSTAT controls whether enhanced debugging capabilities can be activated from the SSH server by ## Avaya technicians only. The parameter shall only be set to 1 for the debugging period by Avaya technicians and ## shall be configured back to 0 when the debugging period is end. ## Value Operation ## 0 Enhanced debugging capabilities are disabled (default). ## 1 Enhanced debugging capabilities are enabled. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; While this parameter is supported, Avaya Technician is NOT expected ## to use it in the field and customers are encouraged to verify it is remain with its default value. ## 96x1 SIP R7.0.1.0 and later releases (hardware version 3 and up). ## 96x1 H.323 R6.6.2 and later releases (hardware version 3 and up). ## SET AUTHCTRLSTAT 1 ## ######## APPLICATION WATCHDOG SETTING (H.323 ONLY) ####### ## ## APPLICATIONWD specifies whether the application watchdog is enabled or disabled ## Value Operation ## 0 Application watchdog is disabled ## 1 Application watchdog is enabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.2 and later ## B189 H.323 R1.0 and later ## SET APPLICATIONWD 0 ## ##### SERVICE LEVEL AGREEMENT (SLA) MONITOR SETTINGS ##### ## ## Please note that SLA Monitor agent requires specification of the root ## (and intermediate if applicable) trusted certificates using TRUSTCERTS for verifying ## the SLA Monitor server certificate. ## ## SLMSTAT specifies whether or not the SLA Monitor agent is enabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R3.1.4 and later ## SET SLMSTAT 1 ## ## SLMCAP specifies whether the SLA Monitor agent is enabled for packet capture (sniffing). ## Value Operation ## 0 Disabled (default) ## 1 Enabled with payloads are removed from RTP packets ## 2 Enabled with payloads included in RTP packets ## 3 Controlled from craft menu - enable of RTP packets capture or disable packets capture ## using local CRAFT procedures. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (values 0-3) ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later (values 0-3) ## 96x1 SIP R6.2 to R6.5 (values 0-2) ## 96x1 SIP R7.0 and later (values 0-3) ## 96x0 H.323 R3.1.4 and later (values 0-2) ## SET SLMCAP 1 ## ## SLMCTRL specifies whether the SLA Monitor agent is enabled for device control. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## 2 Controlled from craft menu ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (values 0-2) ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later (values 0-2) ## 96x1 SIP R6.2 to R6.5 (values 0-1) ## 96x1 SIP R7.0 and later (values 0-2) ## 96x0 H.323 R3.1.4 and later (values 0-1) ## SET SLMCTRL 1 ## ## SLMPERF specifies whether the SLA Monitor agent is enabled for device performance monitoring. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R3.1.4 and later ## SET SLMPERF 1 ## ## SLMPORT specifies the UDP port that will be opened by the SLA Monitor agent ## to receive discovery and test request messages. ## Valid values are 6000 through 65535; the default value is 50011. ## Important note: If default port is not used, both the SLA Mon agent and server must ## be configured with the SAME port. SLMPORT impacts the phone's SLA Mon agent configuration. ## A corresponding configuration must also be made on the SLA Mon server agentcom-slamon.conf file. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R3.1.4 and later ## SET SLMPORT 43210 ## ## SLMSRVR specifies the IP address and the port number of the SLA Mon server in the ## aaa.bbb.ccc.ddd:n format. ## Set the IP address of the SLA Mon server in the aaa.bbb.ccc.ddd format ## to restrict the registration of agents only to that server. Specifying a port ## number is optional. If you do not specify a port number, the system takes ## 50011 as the default port. If the value of the port number is 0, any port number is acceptable. ## The IP address must be in the dotted decimal format, optionally followed by a ## colon and an integer port number from 0 to 65535. ## To use a non-default port n, set the value of SLMSRVR in the ## aaa.bbb.ccc.ddd:n format, where aaa.bbb.ccc.ddd is the IP address ## of the SLA Mon server. ## Important note: If default port is not used, both the SLA Mon agent and server must ## be configured with the SAME port. SLMSRVR impacts the phone's SLA Mon agent configuration. ## A corresponding configuration must also be made on the SLA Mon server agentcom-slamon.conf file. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## B189 H.323 R6.7.1 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.4 and later ## 96x1 SIP R6.2 and later ## 96x0 H.323 R3.1.4 and later ## SET SLMSRVR 192.168.27.35:50011 ## ####### CONVERGED NETWORK ANALYZER (CNA) SETTINGS ######## ## ## CNASRVR specifies a list of CNA server IP addresses. ## Addresses can be in dotted-decimal (IPv4) or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## 96x1 H.323 R6.0 and R6.1 ## 96x1 SIP R6.0.x only ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET CNASRVR 192.168.0.10 ## ## CNAPORT specifies the TCP destination port used for CNA registration. ## Valid values are 0 through 65535; the default value is 50002. ## This parameter is supported by: ## 96x1 H.323 R6.0 and R6.1 ## 96x1 SIP R6.0.x only ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.0 and later ## SET CNAPORT 65003 ## ############# ENHANCED LOCAL DIALING RULES ############### ## ## These settings affect certain dialing behaviors, such as ## dialing numbers from the incoming Call Log or from web pages ## Please note that the enhanced local dialing rules are not applicable ## when using 96x0/96x1 H.323 phones in IP office environment. ## The parameters below are supported by 16xx H.323 phones. ## ## Dialing Algorithm Status ## Controls whether algorithm defined by parameters in ## this section is used during certain dialing behaviors. ## 0 disables algorithm. ## 1 enables algorithm, but not for Contacts (default). For B189 H.323 only, value 1 has the same meaning as value 0 ## since B189 does not support call log application and WML browser. ## 2 enables algorithm, including Contacts (96xx SIP R2.0 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R6.0 and later, 96x1/B189 H.323 6.6 and later, H1xx SIP R1.0 and later, J169/J179 H.323 R6.7 and later) ## Note: Avaya Vantage Basic Application SIP R1.0.0.1 and later and Avaya Equinox 3.1.2 and later support values 0 (disabled) and 1 where value means that enhanced local dialing rules are applied on ## all outgoing calls (whether originated from contacts, history or dialer). All Enhanced local dialing rules parameters mentioned in this section which are marked as supported ## by Avaya Vantage Basic application SIP R1.0.0.1 and later and Avaya Equinox 3.1.2 and later can be supported by any Avaya Breeze Client SDK based application. ## SET ENHDIALSTAT 1 ## ## Country Code ## For United States the value is '1' ## Valid values 1 to 999. The default value is 1. ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNCC 1 ## ## Internal extension number length ## If your extension is 12345, your dial plan length is 5. ## On 96xx phones, the maximum extension length is 13. ## This value must match the extension length set on your ## call server. ## Valid values are 3-13. The default value is 5. ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNDPLENGTH 5 ## ## International access code ## For the United States, the value is 011. ## Valid values are 0 to 4 dialable characters (0-9,*,#). The default value is "011". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNIC 011 ## ## Long distance access code ## Valid values are 0 through 9 and empty string. The default value is 1. ## if no long distance access code is needed then SET PHNLD "". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNLD 1 ## ## National telephone number Length ## For example, 800-555-1111 has a length of 10. ## Valid values are 5-15. The default value is 10. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNLDLENGTH 10 ## ## Outside line access code ## The number you press to make an outside call. ## Valid values are 0 to 2 dialable characters (0-9, *, #). The default value is 9. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## Avaya Equinox 3.1.2 and later; default is "". ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; default is "". ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R2.5 and later ## SET PHNOL 9 ## ## ELD_SYSNUM ## Controls whether Enhanced Local Dialing algorithm will be ## applied for System Numbers - Busy Indicators and Auto Dials. ## Value Operation ## 0 Disable ELD for System Numbers ## 1 Enable ELD for System Numbers (Default) ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.0.1.2 and later. ## ## SET ELD_SYSNUM 1 ## ## APPLY_DIALINGRULES_TO_PLUS_NUMBERS specifies whether to apply dialing rules on numbers with "+" at the beginning. ## Value Operation ## 0 Dialing rules are not applied on numbers that beginning with "+" (default). ## 1 Dialing rules are applied to strip the "+" and replace with dial plan digits. ## Customers are recommended to configure "+" dialing on Avaya Session Manager as the preferable solution over enabling this feature in the endpoints. ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET APPLY_DIALINGRULES_TO_PLUS_NUMBERS 1 ## ## AUTOAPPLY_ARS_TO_SHORTNUMBERS specifies whether to disable the dialing rules logic that automatically appends the outside line access code (PHNOL) to numbers ## that are shorter than the shortest extension length. ## Value Operation ## 0 Do not append Outside line access code to numbers that are shorter than the shortest extension length. ## 1 Append Outside line access code to numbers that are shorter than the shortest extension length (default). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET AUTOAPPLY_ARS_TO_SHORTNUMBERS 0 ## ## DIALPLANLOCALCALLPREFIX indicates whether the area code must be removed for local calls. ## Value Operation ## 0 Disabled, area code is not removed for local calls (default). ## 1 Enabled, area code is removed for local calls. ## Note: Area code is configured using DIALPLANAREACODE. ## Note: PHNREMOVEAREACODE is obsoleted by DIALPLANLOCALCALLPREFIX and the recommendation is to use DIALPLANLOCALCALLPREFIX (though, PHNREMOVEAREACODE is supported ## for backward compatibility). ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET DIALPLANLOCALCALLPREFIX 1 ## ## DIALPLANNATIONALPHONENUMLENGTHLIST defines the national number length list. List of comma separated integers (Basically a collection of PHNLDLENGTH values). ## If PHNLDLENGTH is also present DIALPLANNATIONALPHONENUMLENGTHLIST takes precedence (PHNLDLENGTH is ignored). The default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET DIALPLANNATIONALPHONENUMLENGTHLIST 10,11 ## ## DIALPLANEXTENSIONLENGTHLIST defines the internal extension length list. List of comma separated integers (Basically a collection of PHNDPLENGTH values). ## If PHNDPLENGTH is also present DIALPLANEXTENSIONLENGTHLIST takes precedence (PHNDPLENGTH is ignored). The default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET DIALPLANEXTENSIONLENGTHLIST 7,8 ## ## DIALPLANPBXPREFIX defines the PBX Main Prefix. ## Note: PHNPBXMAINPREFIX is obsoleted by DIALPLANPBXPREFIX and the recommendation is to use DIALPLANPBXPREFIX (though, PHNPBXMAINPREFIX is supported ## for backward compatibility). The default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET DIALPLANPBXPREFIX 538 ## ## DIALPLANAREACODE defines the area code. ## Note: SP_AC is obsoleted by DIALPLANAREACODE and the recommendation is to use DIALPLANAREACODE (though, SP_AC is supported ## for backward compatibility). The default value is "". ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.1 and later; ## SET DIALPLANAREACODE 303 ## #################### CALL TYPE ANALYSIS ############### ## ## CTASTAT - Call Type Analysis Status ## Controls whether call type analysis algorithm in the Avaya Communication Manager is used ## during certain dialing behaviors. ## Value Operation ## 0 Do not use smart enbloc even if smart enbloc is enabled/supported by Avaya Communication Manager by History, Redial, WML browser and Contacts applications. ## This option shall be used to support call forward for history/redial/contacts until Avaya Communication Manager will officially support CTA with call forwarding. ## 1 use smart enbloc if smart enbloc is enabled/supported by Avaya Communication Manager by History, Redial and WML browser, but not for Contacts. ## 2 use smart enbloc if smart enbloc is enabled/supported by Avaya Communication Manager by History, Redial, WML browser and Contacts (Default). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.6.2 and later; Value 0 is supported in R6.6.5 and later. ## B189 H.323 R6.6.2 and later; Value 0 is supported in R6.6.5 and later. ## SET CTASTAT 1 ## #################### AUDIO SETTINGS ###################### ## ## Automatic Gain Control (AGC). ## These settings enable or disable AGC. ## ## A value of 1 (default) enables AGC. A value of 0 disables AGC. ## AGCHAND controls handset AGC. ## AGCHEAD controls headset AGC ## AGCSPKR controls speaker AGC. ## Note: AGCHAND, AGCHEAD and AGCSPKR are supported by J169/J179 H.323 R6.7 and later, H1xx SIP R1.0 and later and Avaya Vantage Devices SIP R1.0.0.0 and later. ## Note: AGCHAND and AGCSPKR are supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. AGCHEAD is supported by J100 SIP R2.0.0.0 and later (J169/J179 only) ## and J139 SIP R3.0.0.0 and later. ## Note: For 96x1 H.323 and J169/J179 H.323- User can also change the "Handset/Headset/Speaker Auto Gain Control" fields ## in HOME-> Options & Settings-> Advanced Options -> Automatic Gain Control... menu. ## AGCHAND/AGCHEAD/AGCSPKR will be enforced only in case user did not change at all the relevant "Handset/Headset/Speaker Auto Gain Control" field value. ## Please note that user changes are stored in backup/restore file as "Handset AGC", "Headset AGC" and "Speaker AGC" (if BRURI has a valid value) which means that if the ## restored file include "Handset AGC", "Headset AGC" and/or "Speaker AGC" parameters then they will take precedence over AGCHAND, AGCHEAD and AGCSPKR respectively. ## If BRURI is not valid, but user still change the content of "Handset/Headset/Speaker Auto Gain Control" fields, then user value will take precedence over ## AGCHAND, AGCHEAD and AGCSPKR respectively. The only way to clear user configuration in this case is by doing: ## a. "CLEAR" operation in CRAFT menu, ## b. New user login. ## SET AGCHAND 0 ## SET AGCHEAD 0 ## SET AGCSPKR 0 ## ## Audio Environment Index ## Enables you to customize the telephone's audio ## performance. (0-299) This parameter affects settings ## for AGC dynamic range, handset and headset noise ## reduction thresholds, and headset transmit gain. It is ## highly recommended you consult Avaya before changing ## this parameter. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0 and J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 "0" through "80" (R1.0, R1.1), "0" through "191" (R1.2 - R1.5), "0" through "299" (R2.0+) ## 96x0 SIP R2.6 and later ## 16xx H.323 R1.3 and later ## SET AUDIOENV 0 ## ##################### CUSTOM RING TONES #################### ## ## RINGTONESTYLE specifies the style of ring tones that are offered to the user ## for Personalized Ringing when "Classic" (as opposed to "Rich") is selected. ## Value Operation ## 0 North American ring tones are offered (default) ## 1 European ring tones are offered ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later ## 9670 H.323 R2.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.3 and later ## H1xx SIP R1.0 and later ## SET RINGTONESTYLE 1 ## ## RINGTONES specifies a list of display names and file names or URLs ## for a custom ring tone files to be downloaded and offered to users. ## The list can contain 0 to 1023 UTF-8 characters; the default value is null (""). ## Values are separated by commas without any intervening spaces. ## Each value consists of a display name followed by an equals sign followed by a file name or URL. ## Display names may contain spaces, but if any do, the entire list must be quoted. ## Ring tone files must be single-channel WAV files ## coded in ITU-T G.711 u-law or A-law PCM with 8-bit samples at 8kHz. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.1.0.0 and later; Avaya Vantage supports list of ringtones files (without the tuple format) if the files are stored in the same directory as FILE_SERVER_URL points to. ## If a different directory is needed, then the tuple format shall be used and the format shall be =”path/filename”. ## For example: “name.wav/mp3/ogg=URI”. ## If the ringtones files are stored on the same directory then list of filenames shall be in the following format: “ring1.wav,ring2.wav,ring3.mp3,rin4.ogg” ## If there is use of mp3/ogg files which includes the ID3 metadata container with non-empty title field then this title field will be ## presented by Android in the list of ringtones. ## If the mp3/ogg file include ID3 metadata container with empty title field then the filename shall be displayed. ## If wav file is used then the filename is always presented. ## When using tuple format then the filename MUST include the audio file suffix (.mp3/.wav.ogg). ## Changing file suffix only with the same filename will not trigger new download (I.e. this is not considered as new file). ## The filename must be changed in order to trigger new download of the ringtones files. ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.3 and later ## H1xx SIP R1.0 and later ## SET RINGTONES "Steam Whistle=tones/swhistle.wav,Car Horn=tones/chorn.wav,Siren=tones/siren.wav" ## Example for Avaya Vantage: ## SET RINGTONES "whistle.wav,chorn.wav,siren.wav" ## SET RINGTONES "you.wav=wavefiles/you_talkin.wav,ring4.mp3=mp3files/ring.mp3" ## Note: In order to set RINGTONES for Avaya Vantage and other phones then shall be use of the IF conditional statement (e.g. IF $MODEL4 SEQ K175 GOTO SETTINGSK1XX) ## with $MODEL4 to separate between K155/K175/K165 to other phones. ## ## RINGTONES_UPDATE specifies whether the phone will query the file server to determine whether ## there is an updated version of each custom ring tone file each time the phone starts up or resets. ## Value Operation ## 0 Phone will only try to download ring tones with new display names (default) ## 1 Phone will check for updated version of each ring tone file at startup ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only) ## 96x1 SIP R6.3 and later. ## SET RINGTONES_UPDATE 1 ## ## PROVIDE_CF_RINGTONE specifies whether the call forward ringtone option is provided to the user. ## Value Operation ## 0 the call forward ringtone option is not provided (default) ## 1 the call forward ringtone option is provided ## This parameter is supported by: ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later. ## The user option is displayed in "Settings > Audio > Personalize ringing > Call Fwd Ring". ## 96x1 SIP R6.3 and later. ## SET PROVIDE_CF_RINGTONE 1 ## ## ADMIN_CHOICE_RINGTONE specifies the administrator choice of ringtone to be used for incoming calls. ## The default value is "Default" which represents Avaya built in ringtone. ## Otherwise, one of the ringtones name shall be configured. ## User may choose to use its own ringtone. ## The parameter can also be used to choose ringtone out of downloaded ringtones (using RINGTONES parameter supported in Avaya Vantage Devices SIP R1.1.0.0 and later). ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ADMIN_CHOICE_RINGTONE "Titania" ## ################## FILE SERVER SETTINGS ################## ## ## HTTP Server Addresses ## [If you set your HTTP Server Addresses via DHCP, do not ## set them here as they will override your DHCP settings. ## Server used to download configuration script files. ## Zero or more HTTP server IP addresses in dotted-decimal, ## colon-hex (96x1 H.323 R6.0 onwards), or DNS name format, ## separated by commas without any intervening spaces. ## (0 to 255 ASCII characters, including commas). ## This parameter may also be changed via LLDP. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## B189 H.323 R1.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET HTTPSRVR 192.168.0.5 ## ## HTTP Server Directory Path ## Specifies the path name to prepend to all file names ## used in HTTP and HTTPS GET operations during startup. ## (0 to 127 ASCII characters, no spaces.) ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J169/J179 H.323 R6.7 and later. ## SET HTTPDIR myhttpdir ## ## HTTP port ## Sets the TCP port used for HTTP file downloads from ## non-Avaya servers. (0-65535) The default value is 80. ## Applies only to 96xx phones, 96x1 phones, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J169/J179 H.323 R6.7 and later. ## SET HTTPPORT 80 ## ## File Server URL ## Specifies list of file server URLs from which firmware and configuration files shall be downloaded. ## Each URL can include hostname or IP address, port and path. ## This parameter has higher precedence compare to HTTPSRVR, HTTPDIR and HTTPPORT which are kept supported ## for backward compatibility. If FILE_SERVER_URI is configured, then HTTPSRVR, HTTPPORT, HTTPDIR, TLSSRVR, ## TLSDIR and TLSPORT parameters are ignored. Default protocol is http://. Default ports are 80 when http ## is used and 443 when https is used. Default is "". ## By default Utility Server uses TCP port 411 with https://. In such case, port 411 shall be configured. ## This parameter may also be changed via LLDP. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET FILE_SERVER_URL http://example.com:8000/H1xx ## ## Server Authentication ## Sets whether script files are downloaded from an ## authenticated server over an HTTPS link. ## 0 for optional, 1 for mandatory ## Note: This parameter is also supported by J169/J179 H.323 R6.7 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## H1xx SIP R1.0 and later and Avaya Vantage Devices SIP R1.0.0.0 and later. ## SET AUTH 0 ## ## HTTPS Server Addresses ## [If you set your HTTP/S Server Addresses via DHCP, do not ## set them here as they will override your DHCP settings. ## Server used to download configuration script files. ## Zero or more HTTPS server IP addresses in dotted-decimal, ## colon-hex, or DNS name format, ## separated by commas without any intervening spaces. ## (0 to 255 ASCII characters, including commas). ## This parameter may also be changed via LLDP. ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET TLSSRVR 192.168.0.5 ## ## HTTPS Server Directory Path ## Specifies the path name to prepend to all file names ## used in HTTPS GET operations during startup. ## (0 to 127 ASCII characters, no spaces.) ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET TLSDIR myhttpdir ## ## HTTPS port ## Sets the port used for HTTPS file downloads from ## non-Avaya servers. (0-65535) The default value is 443. ## This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET TLSPORT 411 ## ################## DEVICE ENROLLMENT SERVICE (DES) ################## ## ## DES_STAT Specifies if DES discovery is to be attempted during the boot process if there is no configuration file server provisioned on the phone. ## Value Operation ## 0 DES discovery is disabled and can only be restored with Reset to Defaults ## 1 DES discovery is disabled ## 2 DES discovery is enabled (default); user prompt is displayed for end user to enforce "DES" or not. ## 3 DES discovery is enforced without dependency on user to select "yes" on the prompt appears after reboot. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, value 3 is supported by J100 SIP R4.0.0.0 and later ## Avaya Vantage Devices SIP R1.1.0.0 and later; if FILE_SERVER_URL/HTTPSRVR/TLSSRVR are received from DHCP/LLDP/UI/configuration file/AADS then DES will not be activated. ## No support for user prompt which can enforce DES or not. Value 3 is not supported. ## SET DES_STAT 1 ## ################## Upgrade Policy ################## ## ## UPGRADE_POLLING_PERIOD ## Specifies the periodic polling interval in minutes of ## upgrade and settings files. The range is 0-10080, where 0 means no periodic polling. The default is 60 minutes. ## In each polling, the upgrade file and settings files are downloaded if modified (using If-Modified-Since) and the device ## check whether new firmware is installed on the file server. If there is a change identified to the settings file ## then the device applies the new settings file. If there is a new firmware detected, then it will be downloaded and installed ## according to UPGRADE_POLICY, UPGRADE_DLOAD_START, UPGRADE_DLOAD_END, UPGRADE_INSTALL_DATE_TIME, DLOAD_RND_AFTER_RESET and ## DLOAD_RND. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET UPGRADE_POLLING_PERIOD 1000 ## ## UPGRADE_POLICY ## Specifies whether the image update is done after reset only, reset and policy configuration parameters or policy only. ## Value Operation ## 0 update of an image after reset only (any reset). ## 1 update of an image according to upgrade policy rules (reset will not trigger new update of an image). ## 2 update of an image after both reset and according to upgrade policy rules (Default) ## Note: Settings files are always downloaded/updated after reset or downloaded/updated if modified each UPGRADE_POLLING_PERIOD ## (independent to UPGRADE_POLICY). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; for IPO environment, please set this value to 0. ## H1xx SIP R1.0 and later ## SET UPGRADE_POLICY 1 ## ## UPGRADE_DLOAD_START ## Specifies the start time at which phone tries to download the new image files. ## The format is [Ddd]hh where "[Ddd]" is an optional argument ## Ddd is a 3-character string for a day of the week (Sun, Mon, Tue, Wed, Thu, Fri, Sat), ## "hh" is one or two numeric digits representing the hour of the day, from 0 through 23. ## If "Ddd" is omitted, the time period would occur every day, and if days are included, the time period would occur once every week. ## If UPGRADE_DLOAD_START and UPGRADE_DLOAD_END are the same then, download of files will be done at any time. ## The default is "00". Up to one start time can be configured. UPGRADE_DLOAD_START is applicable when UPGRADE_POLICY is 1 or 2. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET UPGRADE_DLOAD_START Sun00 ## ## UPGRADE_DLOAD_END ## Specifies the end time at which phone tries to download the new image files. ## The format is [Ddd]hh where "[Ddd]" is an optional argument. ## Ddd is a 3-character string for a day of the week (Sun, Mon, Tue, Wed, Thu, Fri, Sat), ## "hh" is one or two numeric digits representing the hour of the day, from 0 through 23. ## If "Ddd" is omitted, the time period would occur every day, and if days are included, the time period would occur once every week. ## If UPGRADE_DLOAD_START and UPGRADE_DLOAD_END are the same then, download of files will be done at any time. ## The default is "00". Up to one end time can be configured. UPGRADE_DLOAD_END is applicable when UPGRADE_POLICY is 1 or 2. ## Note: Existings image download will keep until finished even when UPGRADE_DLOAD_END reached, however, new image downloads will ## be scheduled to next download timeframe. ## Examples: ## If UPGRADE_DLOAD_START is 04 and UPGRADE_DLOAD_END is 08 then the device will download each day the new firmware files (if new ## firmware is detected) from 04:00 to 08:00. ## If UPGRADE_DLOAD_START is 08 and UPGRADE_DLOAD_END is 04 then the device will download each day the new firmware files (if new ## firmware is detected) from 08:00 to 04:00 the next day. ## If UPGRADE_DLOAD_START is Sat20 and UPGRADE_DLOAD_END is Mon04 then the device will download the new firmware (if new ## firmware is detected) from 20:00 on Saturday to 04:00 on Monday each week. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET UPGRADE_DLOAD_END Sun05 ## ## UPGRADE_INSTALL_DATE_TIME ## Specifies The date and time after which new image will be installed. ## If the image was not downloaded yet and the install date/time is reached, then the device download the image immediately (no matter what is the value of ## UPGRADE_DLOAD_START and UPGRADE_DLOAD_END). If the image was not downloaded yet and install date/time in future, then ## the device will download the image according to UPGRADE_DLOAD_START and UPGRADE_DLOAD_END and once UPGRADE_INSTALL_DATE_TIME is reached the image will be installed. ## The format is YYYY-MM-DDThh:mm, where YYYY is a 4 numeric digits representing the year, MM is 2 numeric digits for month 00-12, ## dd is two numeric digits representing the day of the month, from 01 through 31, "T" stand for Time separator, ## hh is two numeric digits representing the hour of the day, from 00 through 23 and mm is two numeric digits representing minutes of the hour, ## from 00 through 59. The default is 1970-01-01T00:00. ## Note: This parameter is related to the phone local time and not absolute time (UTC/GMT). This means that phones in different ## GMTOFFFSET will be installed in the same local time, but in different universe time. The phone calculate its local time ## based on GMTOFFSET, DAYLIGHT_SAVING_SETTING_MODE, DSTOFFSET, DSTSTART and DSTSTOP. ## UPGRADE_INSTALL_DATE_TIME is applicable when UPGRADE_POLICY is 1 or 2. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET UPGRADE_INSTALL_DATE_TIME 2015-04-12T23:20 ## ## DLOAD_RND_AFTER_RESET ## Specifies the interval in seconds for which downloading attempts will be randomized after reboot. ## The range is 0-32767 where 0 is for no randomization (I.e. download image file immediately after reboot). ## 0 is the default value. ## The parameter can be used to avoid congestion of file server in case where multiple devices are reset at the same time ## using SMGR. ## The parameter is applicable only if UPGRADE_POLICY<>1. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET DLOAD_RND_AFTER_RESET 600 ## ## DLOAD_RND ## Specifies the interval in seconds for which downloading attempts will be randomized during the download slot ## (UPGRADE_DLOAD_START<>UPGRADE_DLOAD_END). ## The range is 0-32767 where 0 is for no randomization (I.e. download image file immediately when UPGRADE_DLOAD_START is reached ## (if UPGRADE_DLOAD_START<>UPGRADE_DLOAD_END)). ## 3600 is the default value. ## The parameter can be used to avoid congestion of file server in case where multiple devices reach the download period slot. ## The parameter is applicable only if UPGRADE_POLICY==1 or 2 and UPGRADE_DLOAD_START<>UPGRADE_DLOAD_END. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## H1xx SIP R1.0 and later ## SET DLOAD_RND 0 ## ## IMAGE_DOWNLOAD_RATE_LIMIT ## Specifies the image download rate from HTTP/S server. ## The range is 0-1,000,000 kbps where 0 is for no rate limit (default). ## The parameter shall be used in cases where the download rate is limited (as in case of remote worker working via broadband connection at home). ## In those cases, administrator can use this parameter to configure the maximum bandwidth used for download of image files. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET IMAGE_DOWNLOAD_RATE_LIMIT 1000 ## ################### RTCP MONITORING ##################### ## ## The RTCP monitor ## One RTCP monitor (VMM server) IP address in ## dotted-decimal format or DNS name format (0 to 15 ## characters). Note that for H.323 telephones only this ## parameter may be changed via signaling from Avaya ## Communication Manager. For 96xx/J100 SIP models in Avaya Aura ## environments, this parameter is set via the PPM server. ## Note : This setting is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## H1xx SIP R1.0 and later for non-Aura environment (For example: IP Office, etc). ## SET RTCPMON 135.169.56.99 ## ## RTCPMONPORT sets the port used to send RTCP information ## to the IP address specified in the RTCPMON parameter. ## RTCPMONPORT is only supported on 96xx/J100 in non-Avaya environments. For 96xx/J100 SIP ## models in Avaya environments, this parameter is set via the PPM server. The default value is 5005. ## Note : This setting is supported by H1xx SIP R1.0 and later and J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, ## J139 SIP R3.0.0.0 and later for non-Aura environment (For example: IP Office, etc). ## SET RTCPMONPORT 5005 ## ## RTCP Monitor Report Period ## Specifies the interval for sending out RTCP monitoring ## reports (5-30 seconds). Default is 5 seconds. This ## parameter applies only to 96xx/J100 SIP telephones. ## Note : This setting is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## for non-Aura environment (For example: IP Office, etc). ## SET RTCPMONPERIOD 5 ## ###################### ICMP SETTINGS ##################### ## ## Destination Unreachable Message Control ## Controls whether ICMP Destination Unreachable messages ## are generated. ## 0 for No ## 1 for limited Port Unreachable messages ## 2 for Protocol and Port Unreachable messages ## Note 1: This settings is also applicable for J169/J179 H.323 R6.7 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## H1xx SIP R1.0 and later and Avaya Vantage Devices SIP R1.0.0.0 and later. ## SET ICMPDU 1 ## ## Redirect Message control ## Controls whether received ICMP Redirect messages will ## be processed ## 0 for No ## 1 for Yes ## Note 1: This settings is also applicable for J169/J179 H.323 R6.7 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## H1xx SIP R1.0 and later and Avaya Vantage Devices SIP R1.0.0.0 and later. ## SET ICMPRED 0 ## ########### BACKUP/RESTORE SETTINGS (H.323 and SIP) ########## ## ## Backup and Restore URI ## URI used for HTTP backup and retrieval of user data. ## Specify HTTP server and directory path to backup file. ## Do not specify backup file name. ## BRURI is supported by 16xx H323 phones using http only (not https). ## Note: This parameter is supported by J169/J179 H.323 R6.7 and later, 96x1 H323 R6.2.3 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## H1xx SIP R1.0 and later and 96x1 SIP R7.1.0.0 and later releases for sending a phone report to a HTTP/S file server (with the URI of the server defined by the BRURI parameter). ## In order to send the report the phone must be registered and the administrator must access the ## phone’s Admin menu and select the Send Report feature. This parameter is supported in both IPO and Aura environment with file server that support HTTP PUT messages. ## SET BRURI http://192.168.0.28 ## Note: 96x0 H.323 R3.2/96x1 H.323 R6.0 phones support in addition a format of "http://username:password@..." or "https://username:password@..." for HTTP Basic authentication. ## The username and password are removed from the configured URI and used in the Authorization Header. The HTTP request will be sent to the URI without the username and password fields. ## For example: ## SET BRURI http://Administrator:Catt*123@10.10.10.6/Backup/ ## SET BRURI http://ipphone:Avaya1234@10.10.10.1 ## ## Backup/Restore Authentication ## Specifies whether authentication is used for backup/restore file download. ## Call server IP address and telephone's registration can be used as credentials. ## 0: Call server IP address and telephone's registration password ## are not included as credentials (Default). ## 1: The call server IP address and the telephone's registration ## password are included as the credentials in an Authorization request-header ## Note: This parameter is also supported by J169/J179 H.323 R6.7 and later. ## SET BRAUTH 0 ## #################### AUDIBLE ALERTING ####################### ## ## Specifies the audible alerting setting for the telephone ## and whether users may change this setting. ## ## A value of 0 turns off audible alerting; user cannot ## adjust ringer volume at all. ## A value of 1 turns on audible alerting; user can adjust ## ringer volume but cannot turn off audible alerting. ## A value of 2 turns off audible alerting; user can adjust ## ringer volume and can turn off audible alerting. ## A value of 3 turns on audible alerting; user can adjust ## ringer volume and can turn off audible alerting. ## ## The default value is 3. ## SET AUDASYS 3 ## ## NOTE: This AUDASYS value is applicable for 16xx phones starting with R1.3. ## Note: AUDASYS is not supported by J129 SIP R1.0.0.0/R1.1.0.0. Supported by J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later. ## Note: This parameter is also supported by J169/J179 H.323 R6.7 and later. ## ###################### KEY LAYOUT FILES ##################### ## ## KEY_LAYOUT_FILES specifies the URL of the key layout files to be downloaded. ## 0 to 255 ASCII characters, zero or one URL. ## The URL may be specified relative path format ("../" for next higher directory level in relative path format; ## origin is the directory specified by FILE_SERVER_URL or HTTPDIR and TLSDIR depending on download via http or https). ## URL can be also absolute path – in this case it shall begin with http:// or https://. ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET KEY_LAYOUT_FILES https://149.49.77.1/DEVICE_NAME.kl ## ############################################################ ## ## VOICE MAIL SETTINGS ## ############################################################ ## ## Voice Mail Telephone Number ## Specifies the telephone number to be dialed ## automatically when the telephone user presses the ## Messaging button. The specified number is used to ## connect to the user's Voice Mail system. ## Note: PSTN_VM_NUM shall be used instead of MSGNUM in cases of IP Office environment, 3PCC SIP environment or when there is failover from Aura environment to a non-Aura server. ## Note: This parameter is ignored by 16xx phones when registered to CM version above 5.2. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## H1xx SIP R1.0 and later ## 96x1 H.323 R6.0 and later ## 96x1 SIP R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## Example: ## SET MSGNUM 1234 ## ############################################################ ## ## ENGLISH BUILT-IN LANGUAGE SETTINGS ## ############################################################ ## ## English Language Selection Status ## Specifies whether built-in English language text strings ## are selectable by the user. 0 for off, 1 for on. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R1.0 and later ## 96x0 SIP R1.0 and later ## 16xx H.323 R1.0 and later ## SET LANG0STAT 1 ## ################ A(Avaya) Menu Settings ################# ## ## WML-Application URI ## URI used for WML-applications under A (AVAYA) Menu. ## Specify HTTP server and directory path to administration ## file (AvayaMenuAdmin.txt). Do not specify the ## administration file name. This parameter applies to 96xx H323 ## model phones and also supported in 96xx SIP releases from R2.5 onwards, ## 96x1 SIP releases from R6.2 onwards, J169/J179 SIP R1.5.0 and J169/J179 H.323 R6.7 and later. ## ## SET AMADMIN http://192.168.0.28 ## ## ################################################################# ## ## H.323 SETTINGS for 96xx & 96x1 & J169/J179 H.323 ## Settings specific to 96xx & 96x1 & J169/J179 telephones ## with H.323 software ## ########################## Features on Softkeys ####################### ## ## Idle Feature Settings ## A list of feature identifiers for softkey features ## available in the Idle call state ## 0 to 255 ASCII characters: zero to six whole numbers ## separated by commas without any intervening spaces ## SET IDLEFEATURES "" ## ## Dial Feature Settings ## A list of feature identifiers for softkey features ## available in the Dialing call state ## 0 to 255 ASCII characters :zero to five whole numbers separated ## by commas without any intervening spaces ## SET DIALFEATURES "" ## ## Ring Back Feature Settings ## A list of feature identifiers for softkey features ## available in the Active with far end ringback call state ## 0 to 255 ASCII characters :zero to three whole numbers ## separated by commas without any intervening spaces ## SET RINGBKFEATURES "" ## ## Talk Feature Settings ## A list of feature identifiers for softkey features ## available in the Active with talk path call state ## 0 to 255 ASCII characters :zero to three whole numbers ## separated by commas without any intervening spaces ## SET TALKFEATURES "" ## ## Team Button Display ## When TEAMBTNDISPLAY is set to 1, use LED to mark the Busy state of their team member's phone ## When TEAMBTNDISPLAY is set to 0, use the LED to mark the Forwarding state of the team member's phone. ## Default = 0. ## Note: This feature is available on H.323 release 3.0 for 96xx & release 6.0 for 96x1 phones & J169/J179 H.323 R6.7 phones. ## SET TEAMBTNDISPLAY 0 ## ## WORLDCLOCKAPP specifies the application to display World Clock information. ## Note: This feature is available on H.323 release 2.0 for 9670 & release 6.0 for 9641 & 9621. ## "" : World Clock application is disabled ## "default" : World Clock application is enabled (default) ## SET WORLDCLOCKAPP default ## ## WEATHERAPP specifies the application to display the weather information. ## Value Operation ## "" (null) Weather application is disabled ## "default" Weather application is enabled (default) ## This parameter is supported by: ## 9621 and 9641 H.323 R6.0 and later ## 9670 H.323 R2.0 and later ## SET WEATHERAPP default ## ## CALCSTAT specifies whether the Calculator application should be displayed. ## Valid Values ## 0 Don't display Calculator ## 1 Display Calculator ## SET CALCSTAT 1 ## Note: This feature is available on release 6.0 for 9641 & 9621. ## ## RINGPRIORITY specifies which distinctive ring rate is really for a Priority Call. ## Valid Values ## 1 Inside Call rate ## 2 Outside Call rate ## 3 Priority Ring rate ## SET RINGPRIORITY 3 ## ## LEDMODE specifies whether the buttons red LEDs are controlled by CM (this will align the 96x1 phone behavior to the 16xx behavior) ## or locally by the phone (as 96x1 previous loads behavior ). ## Value Operation ## 0 Locally controlled by the phone, backward compatible to previous releases (default) ## 1 Controlled by CM, new behavior ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.3 and later ## SET LEDMODE 0 ## ################ VPN SETTINGS (H.323 ONLY) ################# ## ## B189 H.323 R6.6.5 supports all VPN parameters specified in "VPN SETTINGS (H.323 ONLY)" section below with exception of VPNCODE, VPNSTAT and QTESTRESPONDER. ## ## NVVPNMODE specifies whether or not VPN operation will be enabled. ## Valid Values ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET NVVPNMODE 1 ## ## NVSGIP specifies a list of IP addresses for VPN security gateways. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET NVSGIP primarysg.mycompany.com ## ## VPNALLOWTAGS specifies whether 802.1Q tags (controlled by L2Q parameter) can be used in VPN mode ## Value Operation ## 0 Tags will not be allowed if VPN mode is active (default) ## 1 Tags will be allowed if VPN mode is active ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.6 and later ## 96x0 H.323 R3.1.5 and later ## SET VPNALLOWTAGS 1 ## ## DHCPSRVR specifies a list of enterprise DHCP server IP addresses from which configuration ## parameters may be requested through a VPN tunnel via a DHCPINFORM message. ## Addresses can be in dotted-decimal or DNS name format, ## separated by commas without any intervening spaces. ## The list can contain up to 255 characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET DHCPSRVR 192.168.16.2 ## ## NVVPNCFGPROF ## Valid Values ## 0 No profile (default) ## 2 Checkpoint ## 3 Cisco Xauth with Preshared Key ## 5 Juniper/Netscreen Xauth with Preshared Key ## 6 Generic Preshared key ## 8 Cisco xauth with certificates ## 9 Juniper Xauth with certificates. ## 11 Nortel contivity ## Description ## Set this to 3 if Security Gateway Vendor is Cisco and Xauth is used for ## authenticating phone user. ## Set this to 5 if Security Gateway Vendor is Juniper, Xauth is used for ## authenticating phone user. ## Set this to 6 if Security Gateway Vendor does not support Xauth. ## Following Variables are set to specified value when NVVPNCFGPROF = 3 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 11 ## NVIKEXCHGMODE 1 ## Following Variables are set to specified value when NVVPNCFGPROF = 5 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 3 ## NVIKEXCHGMODE 1 ## Following Variables are set to specified value when NVVPNCFGPROF = 6 ## NVIKECONFIGMODE 2 ## NVIKEIDTYPE 3 ## NVIKEXCHGMODE 1 ## Following variables are set to specified value when NVVPNCFGPROF = 2 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 11 ## NVIKEOVERTCP 1 ## NVIKEXCHGMODE 2 ## Following variables are set to specified value when NVVPNCFGPROF = 11 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 11 ## NVIKEXCHGMODE 1 ## Following variables are set to specified value when NVVPNCFGPROF = 8 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 11 ## NVIKEXCHGMODE 1 ## Following variables are set to specified value when NVVPNCFGPROF = 9 ## NVIKECONFIGMODE 1 ## NVIKEIDTYPE 3 ## NVIKEXCHGMODE 1 ## Note: SET commands for all the dependent variables mentioned above must ## appear after SET command for NVVPNCFGPROF. ## SET NVVPNCFGPROF 5 ## ## NVIKEXCHGMODE specifies the exchange method to be used for IKE Phase 1. ## Valid Values ## 1 Aggressive Mode (default) ## 2 Main Mode ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET NVIKEXCHGMODE 2 ## ## NVIKECONFIGMODE enables IKE configuration mode. ## Valid Values: ## 1 Enabled (default) ## 2 Disabled ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET NVIKECONFIGMODE 1 ## ## NVVPNAUTHTYPE ## Valid Values ## 3 PSK (default) ## 4 PSK with XAUTH ## 5 RSA Signature with XAUTH ## 6 HYBRID XAUTH ## 7 RSA Signature ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later ## 96x0 H.323 R3.1 and later ## SET NVVPNAUTHTYPE 4 ## ## NVVPNUSER specifies the user name to use for VPN authentication. ## Normally, this value is entered by the user and not set in this file. ## However, if the user names are set in a staging environment using IF statements ## based on the telephones' MAC addresses, or if the user name is a fixed string ## based on the serial number and/or the MAC address of the telephone, ## it may be preconfigured in a file to eliminate the need for the user to manually enter the value. ## If the value contains the string "$SERIALNO" (without the double quotes), ## that string will be replaced by the telephone's serial number, and ## if the value contains the string "$MACADDR" (without the double quotes), ## that string will be replaced by the telephone's MAC address. ## Valid values are strings of ASCII characters; the default value is null (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## B189 H.323 R6.6.5 and later ## 96x1 H.323 R6.0 and later (0 to 30 ASCII characters) ## 96x0 H.323 R3.2 and later (0 to 50 ASCII characters) ## 96x0 H.323 R3.1.1 through R3.1.5 (0 to 30 ASCII characters) ## 96x0 H.323 R3.1 (0 to 16 ASCII characters) ## SET NVVPNUSER AvayaPhone$SERIALNO ## ## NVVPNPSWDTYPE ## Valid Values ## 1 Save in Flash. ## 2 Erase on reset. ## 3 Numeric One Time Password. ## 4 Alpha-Numeric One Time Password. ## 5 Erase on VPN Termination ## Description ## This variables determines how password should be treated. By default ## password type is set to 1. You must set this variable to 3 or 4 if ## using One Time Password such as SecureID from RSA. ## Note ## Setting password type to 3 will not let the user select "Alpahbets" ## while entering password. This might look like an obvious choice when ## using RSA secure ID tokens. However under some conditions user may ## need to respond back by entering 'y' or 'n' in the password field. ## This could happen if RSA ACE server is configured to generate PIN ## instead of letting the user select a PIN. ## Example : Setting password type to 2 (Erase on reset) ## SET NVVPNPSWDTYPE 1 ## ## NVVPNCOPYTOS ## Valid Values ## 1 YES ## 2 NO ## Description ## Value of this variable decides whether TOS bits should be copied from ## inner header to outer header or not. If it's value is 1, TOS bits are ## copied otherwise not. By default TOS bits are not copied from inner ## header to outer header. Some Internet Service Provider don't route the ## IP packets properly if TOS bits are set to anything other than 0. ## Example ## SET NVVPNCOPYTOS 1 ## Note ## It is highly recommended that this value should not be changed if phone ## is downloading the script over the VPN tunnel in order to avoid ## overriding end user setting due to ISP specific issues. For example you ## can set this value to 1 while provisioning phone with VPN firmware so ## that phone can take advantage of QOS service provided by home router but ## if the phone's ISP (Few percent cases) does not handle properly the ## packets with non-zero TOS bits in IP header, phone user will have to ## revert back this value to 2. Under such circumstances it is desirable ## the user's choice don't get overriden every time script is downloaded. ## Example: Setting NVVPNCOPYTOS to 1 if script is not downloaded over VPN tunnel. ## IF $VPNACTIVE SEQ 1 GOTO skipcopytos ## SET NVVPNCOPYTOS 1 ## # skipcopytos ## SET NVVPNCOPYTOS 2 ## ## NVVPNENCAPS ## Valid Values ## 0 4500-4500 ## 1 Disable ## 2 2070-500 ## 4 RFC (As per RFC 3947 and 3948) ## Description ## Type of UDP encapsulation method to use if there is a NAT device between ## phone and the security gateway. By default UDP Encapsulation 4500-4500 ## is used. ## If NVVPNENCAPS is 0, ike negotiation starts with source port of 2070 ## and destination port 500. Negotiation switches to port source port ## 4500 and destination port 4500 if peer supports port floating (Ref ## RFC 3947,3948). Finally IPsec traffic is send inside UDP packets ## from/to port 4500 if supported by peer or port 2070<->500 if port ## floating is not supported but UDP encapsulation is supported as ## published in the initial draft versions of RFC 3947 and 3948. ## If NVVPNENCAPS is 1, ike nat traversal is completly disabled. ## If NVVPNENCAPS is 2, Port floating is disabled during IKE nat traversal. ## If NVVPNENCAPS is 4, ike negotiation starts with source port of 500 and ## destination port 500. Negotiation switches to port source port 4500 ## and destination port 4500 if peer supports port floating (Ref RFC 3947 ## and 3948). Finally IPsec traffic is send inside UDP packets from/to ## port 4500 if supported by peer or port 500<->500 if port floating is ## not supported but UDP encapsulation is supported as published in the ## initial draft versions of RFC 3947 and 3948. ## Note ## UDP Encapsulation causes overhead hence it might be desirable to disable ## udp encapsulation if NAT device supports IPsec pass through and there is ## only one IPsec client behind the NAT connecting to the same security ## gateway. However not all devices support IPsec pass through hence this ## value must not be pushed if phone is downloading the script over the VPN tunnel. ## Example : Setting NVVPNENCAPS to 1 if script is not downloaded over VPN tunnel. ## ## IF $VPNACTIVE SEQ 1 goto skipencaps ## SET NVVPNENCAPS 1 ## # skipencaps ## ## The example above will set NVVPNENCAPS to 1 if script is not downloaded over the tunnel. ## SET NVVPNENCAPS 0 ## ## NVIKEPSK ## Valid Values ## String. Length of the string cannot exceed 30 characters. ## Description ## Preshared Key to use during phase 1 negotiation. ## Note ## It is recommened that user enter his/her Preshared Key using phone's ## dialpad. However if you don't want to share PSK with the end user ## because it's common for multiple users you can use this variable to ## push PSK (Group password) to each phone and the end user will never ## know what the PSK is. But if you are doing this, make sure that the file ## server is on an isolated network and is used only for provisioning ## VPN parameters to the phones. ## Example : Setting abc1234 as Preshared Key ## SET NVIKEPSK abc1234 ## ## NVIKEID ## Valid Values ## String. Length of the string cannot exceed 30 characters. ## Description ## Phone uses this string as IKE Identifier during phase 1 negotiation. ## Some XAuth documentation refer to this variable as group name because ## same IKE Id is shared among a group of user and indvidual user ## authentication is done using XAuth after establishing IKE phase 1 ## security association. ## The default value is "VPNPHONE". ## SET NVIKEID phones@sales.com ## ## NVIKEIDTYPE ## Valid Values ## 1 IP Address ## 2 FQDN ## 3 User-FQDN (E-Mail) ## 9 Directory-Name ## 11 KEY-ID (Opaque) ## Description ## Phone uses this variable as the IKE Identifier type for the ## IKE-ID specified via NVIKEID variable. ## Note ## This variable default value depends on the value of variable NVVPNCFGPROF. ## SET NVIKEIDTYPE 2 ## ## NVIPSECSUBNET ## Valid Values ## Comma separated list of strings containing subnet and masks. Number of ## strings cannot exceed 5. ## Description ## This variable contains IP subnets protected by the security gateway. ## By default phone assumes that all the network resources are behind ## the security gateway hence it negotiates for a security association ## between it's IP address (or Virtual IP if delevired via IKE Config ## mode) and 0.0.0.0 with the security gateway. If your security gateway ## is configured to allow building security association for only selected ## subnets, you can specify them here. ## Example : ## Configuring 10.1.12.0/24 and 172.16.0.0/16 as the subnets protected by ## the Security Gateway ## SET NVIPSECSUBNET 10.1.12.0/24,172.16.0.0/16 ## SET NVIPSECSUBNET 0.0.0.0/0 ## ## NVIKEDHGRP specifies the number of the DH group to use during IKE phase 1 negotiation. ## Valid Values ## 1 Diffie-Hellman Group 1 ## 2 Diffie-Hellman Group 2 (default) ## 5 Diffie-Hellman Group 5 ## 14 Diffie-Hellman Group 14 ## 15 Diffie-Hellman Group 15 ## SET NVIKEDHGRP 1 ## ## NVPFSDHGRP specifies the number of the DH group to use during IKE phase 2 negotiation ## for establishing IPsec security associations also known as perfect forward secrecy (PFS). ## Valid Values ## 0 No PFS (default) ## 1 Diffie-Hellman Group 1 ## 2 Diffie-Hellman Group 2 ## 5 Diffie-Hellman Group 5 ## 14 Diffie-Hellman Group 14 ## 15 Diffie-Hellman Group 15 ## SET NVPFSDHGRP 14 ## ## NVIKEP1ENCALG specifies the Encryption Algorithm(s) proposed for the IKE Phase 1 Security Association. ## Valid Values ## 0 ANY (default) ## 1 AES-128 ## 2 3DES ## 3 DES ## 4 AES-192 ## 5 AES-256 ## Note that the priority order of algorithms proposed is AES-128,3DES,DES,AES-192,AES-256. ## In very rare circumstances a security gateway may not handle multiple proposals. ## In such cases only you should try overriding the default behaviour. ## SET NVIKEP1ENCALG 1 ## ## NVIKEP2ENCALG specifies the encryption algorithm(s) proposed for the IKE Phase 2 Security Association. ## Valid Values ## 0 ANY (default) ## 1 AES-128 ## 2 3DES ## 3 DES ## 4 AES-192 ## 5 AES-256 ## 6 NULL ## Note that the priority order of algorithms proposed is AES-128,3DES,DES,AES-192,AES-256. ## In very rare circumstances a security gateway may not handle multiple proposals. ## In such cases only you should try overriding the default behaviour. ## SET NVIKEP2ENCALG 1 ## ## NVIKEP1AUTHALG specifies the authentication algorithm(s) to propose for the IKE phase 1 Security Association. ## Valid Values ## 0 ANY (default) ## 1 MD5 ## 2 SHA1 ## Note that the prioirity order of algorithims proposed is MD5,SHA1. ## In very rare circumstances a security gateway may not handle multiple proposals. ## In such cases only you should try overriding the default behaviour. ## SET NVIKEP1AUTHALG 1 ## ## NVIKEP2AUTHALG ## Valid Values ## 0 ANY ## 1 MD5 ## 2 SHA1 ## Description ## Authentication Algorithim(s) to propose for IKE phase 2 Security ## Association ## Note ## Phone by default proposes all Authentication algorithms. Security ## Gateway picks the algorithm mandated by administrator. Priority order ## of algorithms proposed by phone is MD5,SHA1. In very rare circumstances ## security gateway may not handle multiple proposals. In such cases ## only you should try overriding the default behaviour. ## Example : Setting Authentication Alg to SHA1 ## SET NVIKEP2AUTHALG 1 ## SET NVIKEP2AUTHALG 0 ## ## NORTELAUTH specifies the Authentication method for Nortel Contivity security gateways. ## Valid Values ## 1 Local username and password ## 2 RADIUS username and password ## 3 Radius SecureId ## 4 RADIUS Axent ## SET NORTELAUTH 2 ## ## NVXAUTH specifies whether or not XAUTH user authentication is enabled. ## Valid Values ## 1 Enabled (default) ## 2 Disabled ## This parameter can be used to disable XAUTH user authentication ## for profiles which enable XAUTH by default. ## SET NVXAUTH 1 ## ## QTESTRESPONDER ## Valid Values: ## IP Address or domain name of the host acting as QTESTRESPONDER ## Description ## If this information is supplied, phone performs QTEST using ## UDP Echo port 7 with the host indicated by this variable. ## SET QTESTRESPONDER 10.1.1.1 ## ## VPNCODE ## Valid Values: 0 to 7 ASCII numeric digits, null ("") and "0" through "9999999" ## Description: Specifies the VPN procedure access code ## SET VPNCODE 876 ## ## VPNPROC specifies whether VPNCODE can be used to access the VPN procedure. ## Value Operation ## 0 Disabled ## 1 View only ## 2 View and edit ## SET VPNPROC 1 ## ## ALWCLRNOTIFY specifies whether unencrypted ISAKMP Notification Payloads will be accepted. ## Valid Values: 1 ASCII numeric digit, "0" or "1" ## SET ALWCLRNOTIFY 0 ## ## DROPCLEAR specifies the treatment of received unencrypted (clear) IPsec packets. ## Valid Values: 1 ASCII numeric digit, "0" or "1" ## SET DROPCLEAR 1 ## ## NVMCIPADD ## Valid Values: 0 to 255 ASCII characters zero or more IP addresses in dotted decimal, ## colon-hex or DNS name format, separated by commas without any intervening spaces. ## Description: Call server IP addresses ## SET NVMCIPADD 0.0.0.0 ## ## NVHTTPSRVR ## Valid Values: 0 to 255 ASCII characters zero or more IP addresses in dotted decimal, ## colon-hex or DNS name format, separated by commas without any intervening spaces. ## Description: HTTP file server IP addresses used to initialize HTTPSRVR the next time the phone starts up. ## SET NVHTTPSRVR 0.0.0.0 ## ## NVTLSSRVR ## Valid Values: 0 to 255 ASCII characters zero or more IP addresses in dotted decimal, ## colon-hex or DNS name format, separated by commas without any intervening spaces. ## Description: HTTPS file server IP addresses used to initialize TLSSRVR the next time the phone starts up. ## SET NVTLSSRVR 0.0.0.0 ## ## NVIKEOVERTCP specifies whether and when to use TCP as a transport protocol for IKE. ## Value Operation ## 0 Never ## 1 Auto ## 2 Always ## SET NVIKEOVERTCP 0 ## ## NVIKEP1LIFESEC specifies the proposed IKE SA lifetime in seconds. ## Valid Values: 3 to 8 ASCII numeric digits, "600" through "15552000" ## SET NVIKEP1LIFESEC 432000 ## ## NVIKEP2LIFESEC specifies the proposed IPsec SA lifetime in seconds. ## Valid Values: 3 to 8 ASCII numeric digits, "600" through "15552000" ## SET NVIKEP2LIFESEC 432000 ## ## NVVPNSVENDOR specifies the security gateway Vendor to be used. ## Value Vendor ## 1 Juniper/Netscreen ## 2 Cisco ## 3 Checkpoint/ Nokia ## 4 Other ## 5 Nortel ## SET NVVPNSVENDOR 4 ## ## NVVPNUSERTYPE specifies whether the user can change the VPN username. ## Value Operation ## 1 The username can be changed during manual credential entry ## 2 The username cannot be changed during manual credential entry ## SET NVVPNUSERTYPE 1 ## ## VPNTTS specifies whether TTS is enabled or disabled when VPN is enabled. ## Value Operation ## 0 Disabled ## 1 Enabled ## SET VPNTTS 0 ## ############################################################# ## IPv6 related settings are applicable for 96x1 H.323 R6.0 and later, J169/J179 H.323 R6.7 and later, 96x1 SIP R7.1.0.0 and later, J169/J179 SIP R1.5.0 and J100 SIP R2.0.0.0 and later. ## Avaya Vantage Devices SIP R1.0.0.0 and later support only IPV6STAT. ## ## NDREDV6 ## Valid Values ## 0 disable ## 1 enable ## Description ## Controls whether IPv6 Neighbor Discovery Redirect messages will be processed ## Note ## Received Redirect messages will be processed if and only if the value of ## the parameter NDREDV6 is "1" otherwise they will be ignored. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and R6.0 ## SET NDREDV6 0 ## ## DHCPPREF ## Valid Values ## 4 DHCPv4 ## 6 DHCPv6 ## Description ## Specifies whether new values received via DHCPv4 orDHCPv6 will be preferred ## when both are used, ## Example : Setting prefernace to recived DHCPPv4 values ## SET DHCPPREF 4 ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and R6.0 ## SET DHCPPREF 6 ## ## DHCPSTAT ## Valid Values ## 1 run DHCPv4 only (IPv4only-mode, if no own IPv6 address is programmed statically), Default. ## 2 run DHCPv6 only (IPv6only-mode, if no own IPv4 address is programmed statically) ## 3 run both DHCPv4 & DHCPv6 (dual-stack mode) ## Description ## Specifies whether DHCPv4, DHCPv6, or both will be used in case IPV6STAT has enabled IPv6 support generally ## Example : Setting dual stack mode ## SET DHCPSTAT 3 ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## For J100 SIP R4.0.0.0 and later - Specifies whether DHCPv6 is enabled or disabled. DHCPSTAT is used in dual (IPV6STAT=1) and IPv6-only (IPV6STAT=2) modes. ## NOTE: DHCPv4 is always enabled in IPv4 only and dual mode. DHCPv4 is disabled in IPv6 only mode. ## Value 1: disable DHCPv6 client, (For IPV6STAT=1 (IPv6 enabled) & "Use SLAAC=No" (SLAAC disabled): IPv4only-mode, if no Phone(v6) IPv6 address is programmed statically ##                   For IPV6STAT=2 (IPv6 only) & "Use SLAAC=No" (SLAAC disabled): Phone(v6) address has to be set manually). ## Value 2,3: enable DHCPv6 client (dual-stack mode). ## The default value is 3. ## J169/J179 H.323 R6.7 and later ## 96x1 SIP R7.1.0.0 and later; Value 1 as described above, Value 2/3 - run both DHCPv4 & DHCPv6  ## 96x1 H.323 R6.0 and R6.0 ## SET DHCPSTAT 1 ## ## PRIVACY_SLAAC_MODE ## Valid Values: ##   0 - Privacy extension disabled, one stable address is generated using modified EUI-64 format interface identifier (Based on MAC address). ## The phone address selection preference is based on default RFC6724 SASA rules. ##   1 - Privacy extension enabled, one stable address is generated using modified EUI-64 format interface identifier (Based on MAC address) ## and one temporary private address is generated. The phone address selection preference is based on RFC6724 SASA rules. ## PRIVACY_SLAAC_MODE changes default SASA rules (i.e. Rule 7) to prefer a manual, DHCPv6 or Stable SLAAC over SLAAC temporary address. (default) ##   2 - Privacy extension enabled, one stable address is generated using modified EUI-64 format interface identifier (Based on MAC address) and ## one temporary private IPv6 address is generated. The phone address selection preference is based on default RFC6724 SASA rules. Default ## SASA Rule 7 is used to prefer SLAAC temporary address over a manual, DHCPv6 or Stable SLAAC addresses.  ## Definition: Specifies the preference for Privacy Extensions(RFC3041) ## This parameter is supported by: ##   J100 SIP R4.0.0.0 and later ## Example: ## SET PRIVACY_SLAAC_MODE 2  ## ## IPPREF ## Valid Values ## 4 IPv4 ## 6 IPv6 ## Description ## Control whether an IPv4 or an IPv6 address returned by DNS would be ## tried first during dual-mode operation. ## In general, if dual-stack operation is enabled, whether IPv4 or IPv6 ## is to be used to contact a server is determined by the value of the ## parameter that contains the server address(es). However, if the value ## is a DNS name and if DNS returns both an IPv4 and an IPv6 address, ## the order in which they will be tried will be based on the order in ## which they are returned to the application by the DNS resolver, which ## is controlled by the parameter ## Example : Setting preference to IPv4 ## SET IPPREF 4 ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and R6.0 ## SET IPPREF 6 ## ## DUAL_IPPREF ## Valid Values ##    4    IPv4 preference (default) ##    6    IPv6 preference ## Description: ##  DUAL_IPPREF controls: ##       1. The selection of SSON either from DHCPv4 or DHCPv6 server, when phone is in dual mode, and ##      2. Whether an IPv4 or IPv6 addresses returned by DNS would be tried first during dual-mode operation. ## DHCP clients use DUAL_IPPREF to decide which SSON configuration attributes to apply for DHCPv4/DHCPv6 interworking in dual mode. Based on DUAL_IPPREF the phone selects SSON attributes either from DHCPv4 or DHCPv6 server. ##     If DNS server name is provided, and if DNS resolver returns both IPv4 and IPv6 addresses, the order in which they will be tried will be based on DUAL_IPPREF parameter. ## NOTE: SIP server FQDNs are resolved into addresses that are ordered based on SIGNALING_ADDR_MODE, not DUAL_IPPREF ## Example: Setting preference to IPv4 ## SET DUAL_IPPREF 6 ##  This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## ## IPV6STAT ## Valid Values ##    0    IPv6 will not be supported (IPv4 only mode). ##    1    Dual mode (IPv4 and IPv6) will be supported. ##    2    IPv6 only mode (only supported by J100 SIP 4.0.0.0 and greater) ## Description ## Specifies whether IPv6 will be supported ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later (Default is 0), J100 SIP R4.0.0.0.0 and later (default is 1). ## Avaya Vantage Devices SIP R1.0.0.0 and later; (Default is 1); IPV6STAT shall be set to 0 as IPv6 is not supported by Avaya Vantage Device. ## 96x1 H.323 R6.0 and R6.0 (Default is 0). ## 96x1 SIP R7.1.0.0 and later (Default is 0). ## SET IPV6STAT 1 ## ## SIGNALING_ADDR_MODE ## Valid Values ## 4 IPv4 (default) ## 6 IPv6 ## Description ## This parameter is used by SIP signaling on a dual mode phone (phone with both IPv4 and IPv6 addresses configured) to select the preferred SIP controller IP addresses ## from SIP_CONTROLLER_LIST_2. The phone registers to SIP controllers using IPv4 address if SIGNALING_ADDR_MODE=4, ## otherwise registration is over IPv6. ## The single IPv4 mode phone ignores SIGNALING_ADDR_MODE and SIP_CONTROLLER_LIST_2 and selects the SIP controller's IP addresses from SIP_CONTROLLER_LIST. ## The single IPv6 mode phone ignores SIGNALING_ADDR_MODE and SIP_CONTROLLER_LIST and selects the SIP controller's IPv6 addresses from SIP_CONTROLLER_LIST_2. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Example: ## SET SIGNALING_ADDR_MODE 4 ## ## MEDIA_NEG_PREFERENCE ## Valid Values ##    0    Remote or offerer's precedence (default) ##    1    Local ## NOTE: MEDIA_NEG_PREFERENCE is NOT used in Avaya environment. Default is remote preference. ## It is used by a dual mode answerer in non-Avaya environment to allow a local preference ## It is used in non-Avaya environment to allow a local preference. ## NOTE: Not applicable on single mode phones. ## Description ##     MEDIA_NEG_PREFERENCE option is used by the answerer only to change the default address ## family preference. ##     In dual IPv4/IPv6 mode, during SIP ANAT negotiation, ##     MEDIA_NEG_PREFERENCE is used to prioritize media lines in SDP. ##     By default offerer's preference is used. ## ##     MEDIA_NEG_PREFERENCE of zero means when there is a choice between IPv4 and IPv6 address, ##     the answerer honors the offerer's preference. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Example ## SET MEDIA_NEG_PREFERENCE 0 ## ## MEDIA_ADDR_MODE ## Valid Values ## 4 IPv4 (default) ## 6 IPv6 ## 46 Prefer IPv4 over IPv6 ## 64 Prefer IPv6 over IPv4 ## Description ## MEDIA_ADDR_MODE specifies the preference of SDP media group lines [per RFC 4091, 4092 and 5888] and the SDP answer / offer format. ## By default v4 media line is preferred. ## MEDIA_ADDR_MODE is only used by dual stack phones which are configured with both IPv4 and IPv6 addresses. ## IPv4 only or IPv6 only phones ignores MEDIA_ADDR_MODE. ## Environment SDP Offer SDP Answer (Note2) ## Avaya 4 – Non ANAT IPv4 only is advertised (Note 3) 4 – IPv4 is chosen (see Note1,3) ## 6 - Non ANAT IPv6 only is advertised (Note 3) 6 – IPv6 is chosen (see Note1,3) ## 46 – ANAT offer where IPv4 is preferred over IPv6 46 – Follow the remote preference. ## 64- ANAT offer where IPv6 is preferred over IPv4 64 – Follow the remote preference. ## Non-Avaya Same as for Avaya environment 4 – IPv4 is chosen (see Note1,3) ## 6 – IPv6 is chosen (see Note1,3) ## 46 – Prefer IPv4 (if available in SDP offer) only if MEDIA_NEG_PREFERENCE ## is set to local, otherwise grants the remote preference. ## 64 – Prefer IPv6 (if available in SDP offer) only if MEDIA_NEG_PREFERENCE ## is set to local, otherwise grants the remote preference. ## Note1: MEDIA_ADDR_MODE=4 and 6 answerers select the MEDIA_ADDR_MODE address family in ANAT offer. ## For non-ANAT offers or ANAT offers with selected "m" line (e.g. re-INVITE), answerers reject the call ## with 488, if MEDIA_ADDR_MODE does not match any of offered audio lines (with non-zero port). ## NOTE2: Answerers are always ANAT capable. ## NOTE3: MEDIA_ADDR_MODE 4 or 6 enforces a dual stack phones which are configured with both IPv4 and IPv6 address to behave as IPv4 only or IPv6 only phone. ## MEDIA_NEG_PREFERENCE is ignored when MEDIA_ADDR_MODE is 4 or 6. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Example : Setting to use IPv6 only ## SET MEDIA_ADDR_MODE 6 ## Example : Setting to preference of IPv6 over IPv4 ## SET MEDIA_ADDR_MODE 64 ## ## IPV6DADXMITS specifies whether Duplicate Address Detection is performed ##    on tentative addresses, as specified in RFC 4862. ##    Non zero value specifies the maximum number of transmitted Neighbor Solicitation messages ##    to determine whether an IPv6 address is already in use. ##  Value  Operation ##    0    DAD is disabled ##    1-5  maximum number of transmitted NS messages ## Default value is 1 ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## Example: ## SET IPV6DADXMITS 1 ## ## PINGREPLYV6 ## Valid Values ## 0 ICMPv6 Echo Reply messages will not be sent ## 1 ICMPv6 Echo Reply messages will be sent only in reply to received Echo ## Request messages with a Destination Address equal to one of the telephone's ## unicast IPv6 addresses. ## 2 ICMPv6 Echo Reply messages will be sent in reply to received Echo Request ## messages with a Destination Address equal to one of the telephone's unicast, ## multicast or anycast IPv6 addresses. ## Description ## Specifies whether ICMPv6 Echo Reply messages will be sent. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and R6.0 ## SET PINGREPLYV6 1 ## ## GRATNAV6 specifies whether gratuitous (unsolicited) IPv6 Neighbor Advertisement messages ## will be processed if they are received. ## Value Operation ## 0 Gratuitous IPv6 Neighbor Advertisement messages will not be processed (default) ## 1 Gratuitous IPv6 Neighbor Advertisement messages will be processed ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and R6.0 ## SET GRATNAV6 0 ## ############################################################ ## ## SIP SETTINGS ## Settings applicable only to 96xx/J100 telephone models ## in non-Avaya environments ## ############################################################ ## ## CALLFWDSTAT sets the call forwarding mode. ## 0 No call forwarding is permitted ## 1 Permits unconditional call forwarding ## 2 Permits call forward on busy ## 3 Permits call forward on busy and unconditional call forwarding ## 4 Permits call forward/no answer ## 5 Permits call forward/no answer and unconditional call forwarding ## 6 Permits call forward/no answer and call forward on busy ## 7 Permits call forward/no answer, call forward on busy and unconditional call forwarding ## The default is 0. ## Note: This parameter is supported by 96x1 SIP, J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later when failed ## over from Aura SM to a non-Aura survivable server (excluding BSM) ## Note: This parameter is supported by J129 SIP R1.0.0.0 and later when configured for IP Office SIP or 3PCC SIP ## SET CALLFWDSTAT 3 ## ## CALLFWDDELAY sets the number of ring cycles before the ## call is forwarded to the forward or coverage address. ## The default delay is one ring cycle. ## Note: This parameter is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later for IP Office Environment. The range is 0 to 20 with Default is 1. ## SET CALLFWDDELAY 5 ## ## CALLFWDADDR sets the address to which calls are forwarded for the call forwarding feature. ## The default is null (""). ## Note the user can change or replace this administered value if CALLFWDSTAT is not 0. ## Note: This parameter is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later for IP Office Environment. ## SET CALLFWDADDR cover@avaya.com ## ## COVERAGEADDR sets the address to which calls will be forwarded for the call coverage feature. ## The default is null (""). ## Note the user can change or replace this administered value if CALLFWDSTAT is not 0. ## This parameter is not supported for 3PCC environment. ## SET COVERAGEADDR cover@avaya.com ## ## SIPCONFERENCECONTINUE specifies whether a conference call continues after the host hangs up. ## 0 for drop all parties (default) ## 1 for continue conference ## SET SIPCONFERENCECONTINUE 0 ## ## ENABLE_AUTO_ANSWER_SUPPORT specifies whether the Auto Answer feature is available to users. ## Value Operation ## 0 Auto Answer feature is not available to users default) ## 1 Auto Answer feature is available to users ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_AUTO_ANSWER_SUPPORT 1 ## ## Auto Answer Mute controls the speakerphone muting when call is auto answered by phone. ## Value Operation ## 0 Speakerphone is Unmuted when Auto Answered ## 1 Speakerphone is Muted when Auto Answered (default) ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET AUTO_ANSWER_MUTE_ENABLE 0 ## ## ENABLE_DND specifies whether the Do Not Disturb feature is available to users. ## Value Operation ## 0 Do Not Disturb feature is not available to users ## 1 Do Not Disturb feature is available to users (default) ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_DND 0 ## ## DND_PRIORITY_OVER_CFU_CFB defines the priority between features Do Not Disturb and Call Forward Unconditional/Busy when both are activated by user. ## Value Operation ## 0 Call Forward Unconditional/Busy feature has priority over Do Not Disturb feature (default) ## 1 Do Not Disturb feature has priority over Call Forward Unconditional/Busy feature ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET ENABLE_DND_PRIORITY_OVER_CFU_CFB 1 ## ## HOLD_REMINDER_TIMER specifies the number of seconds after which the phone will alert (visual and audible) user when any call is kept on hold. ## Valid values are 0 through 999 seconds; the default value is 0. ## Value 0 means phone will not alert user when any call is kept on hold. ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later only for 3PCC environment. ## SET HOLD_REMINDER_TIMER 60 ## ## PROVIDE_TRANSFER_TYPE provides the call transfer type in 3rd party environments. ## No meaning for Avaya environment ## Value 0 or 1 (default 0) ## SET PROVIDE_TRANSFER_TYPE 0 ## ## CALL_TRANSFER_MODE determines the call transfer mode in 3rd party environments. ## Value 0 or 1 (default is 0) ## SET CALL_TRANSFER_MODE 0 ## ############################################################ ## ## 96xx, J100, H1xx SIP SETTINGS ## Settings applicable only to 96xx, J100, and H1xx Video collaboration Station ## running the SIP protocol ## ############################################################ ## ## Power over Ethernet conservation mode ## If POE_CONS_SUPPORT is set to 1 then Power conservation mode is supported. ## If this parameter is set to 0 then Power conservation mode is not supported. ## Note: Not supported by H1xx SIP and J100 SIP. ## SET POE_CONS_SUPPORT 1 ## ## Personalize button labels ability ## CNGLABEL determines ability to personalize button labels to be displayed to ## the user. If it is set to 0 then ability will not be displayed to user. ## If it is set to 1 then personalize button labels ability will be exposed to user. ## Default value is 1. ## Note: Not supported by H1xx SIP, J100 SIP and 96x1 SIP. ## SET CNGLABEL 1. ## ## Selection of Conference Method ## If CONFERENCE_TYPE is set to 0 then local conferencing is supported based on ## sipping services. If set to 1 then server based conferencing is supported. ## If it is set to 2 then click-to conference server based conferencing is supported. ## If it is set to outside range then default value is selected. ## Default value is 1. ## Note: Not supported by H1xx SIP. Supported by J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET CONFERENCE_TYPE 1 ## ## Call Coverage Tone ## Specifies the tone to play when a call goes to ## coverage. The default is 1 and valid values are 1-4. ## SET REDIRECT_TONE 1 ## ## ENABLE_EARLY_MEDIA specifies whether the phone sets up a voice channel ## to the called party before the call is answered. ## Setting this parameter to 1 can speed up call setup. ## 0 for No ## 1 for Yes ## SET ENABLE_EARLY_MEDIA 1 ## ## USE_QUAD_ZEROES_FOR_HOLD specifies the method to use to indicate that a call is on hold. ## A setting of 1 is useful for compatibility with 3rd party SIP endpoints. ## 0 for "a= directional attributes" ## 1 for 0.0.0.0 IP address ## SET USE_QUAD_ZEROES_FOR_HOLD 0 ## ## RTCPCONT specifies whether the sending of RTCP is enabled. ## 0 for No ## 1 for Yes ## SET RTCPCONT 1 ## ## RTCP_XR specifies whether VoIP Metrics Report Block as defined in RTP Control Protocol Extended Reports (RTCP XR) ## (RFC 3611) is sent as part of RTCP packets to remote peer or to RTCP monitoring server. ## 0 for No (Default) ## 1 for Yes ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.0.0 and later ## SET RTCP_XR 1 ## ## MTU_SIZE specifies the maximum transmission unit (MTU) size transmitted by the phone. ## Valid values are 1496 or 1500. ## Use 1496 for older Ethernet switches. ## Note: This parameter is also applicable for H1xx SIP R1.0 and later and for Avaya Vantage Devices SIP R1.0.0.0 and later ## for Ethernet interface only (not Wi-Fi interface where the MTU is fixed 1500 bytes). ## SET MTU_SIZE 1500 ## ## MEDIAENCRYPTION specifies which media encryption (SRTP) options will be supported. ## Up to 2 or 3 options may be specified in a comma-separated list. ## 2 options are supported by: ## 1. Prior releases to 96x1 SIP 7.0.0 ## 2. H1xx SIP R1.0 and later ## 3. 96x0 SIP R1.0 to R2.6.14.1 ## 3 options are supported by 96x1 SIP R7.0.0 and later, J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later and H1xx SIP R1.0.1 and later. ## For 96x0 SIP R2.6.14.5 and later, up to 3 options may be specified, but only the first two supported options are used. ## Options should match those specified in CM IP-codec-set form. ## 1 = aescm128-hmac80 ## 2 = aescm128-hmac32 ## 3 = aescm128-hmac80-unauth ## 4 = aescm128-hmac32-unauth ## 5 = aescm128-hmac80-unenc ## 6 = aescm128-hmac32-unenc ## 7 = aescm128-hmac80-unenc-unauth ## 8 = aescm128-hmac32-unenc-unauth ## 9 = none (default) ## 10 = aescm256-hmac80 ## 11 = aescm256-hmac32 ## Options 10 and 11 are supported by 96x1 SIP R7.0.0 and later, H1xx SIP R1.0.1 and later and J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## Note: The list of media encryption (SRTP) options is ordered from high (left) to the low (right) options. The phone will publish this list in the SDP-OFFER ## or choose from SDP-OFFER list according to the list order defined in MEDIAENCRYPTION. Please note that Avaya Communication Manager has the capability ## to change the list order in the SDP-OFFER (for audio only) when the SDP-OFFER pass through CM. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later ## Avaya Equinox 3.1.2 and later; supported values: 1,2,9,10 and 11. The default value is 1,2,9. ## Avaya Vantage Basic Application SIP R1.0.0.0 and later; supported values: 1,2,9,10 and 11. The default value is 1,2,9. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R1.0 and later ## SET MEDIAENCRYPTION 1,9 ## SET MEDIAENCRYPTION 10,1,9 ## ## ENCRYPT_SRTCP specifies whether RTCP packets are encrypted or not. SRTCP is only used if SRTP is enabled using ## MEDIAENCRYTION (values other than 9 (none) are configured). ## This parameter controls RTCP encryption for RTCP packets exchanged between peers. ## RTCP packets sent to Voice Monitoring Tools are always sent unencrypted. ## Value Operation ## 0 SRTCP is disabled (default). ## 1 SRTCP is enabled. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Equinox 3.1.2 and later ## 96x1 SIP R7.1.0.0 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET ENCRYPT_SRTCP 1 ## ## SUBSCRIBE_SECURITY specifies the use of SIP or SIPS for subscriptions. ## If SUBSCRIBE_SECURITY is 0, the phone uses SIP for both the Request URI and the ## Contact Header regardless of whether SRTP is enabled. If SUBSCRIBE_SECURITY is 1, ## the phone uses SIPS for both the Request URI and the Contact Header if SRTP is enabled ## (TLS is on and MEDIAENCRYPTION has at least one valid crypto suite). ## If SUBSCRIBE_SECURITY is 2, and the SES/PPM does not show a FS-DeviceData FeatureName ## with a FeatureVersion of 2 in the response to the getHomeCapabilities request ## For IP office environment, the applicable values are 0 and 1. ## SET SUBSCRIBE_SECURITY 2 ## #################### IP OFFICE SETTINGS #################### ## ## ENABLE_IPOFFICE specifies whether the deployment environment is IP Office ## Value Operation ## 0 Not IP Office environment (except failover mode to IP Office in Avaya Aura environment) (Default) ## 1 IP Office environment; Native support of IP Office with a limited feature set. ## 2 IP Office environment; Additional features driven by the IP Office SIP proxy. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J129 supports values 0-1 and J169/J179 support values 0 and 2). J139 SIP R3.0.0.0 and later supports values 0 and 2. ## When ENABLE_IPOFFICE is set to 2, some of the 46xxsettings.txt file parameters have no effect. ## Avaya Vantage Basic Application SIP R1.1.0.1 and later (values 0-1) ## Avaya Vantage Devices SIP R1.1.0.1 and later (values 0-1) ## J169/J179 SIP R1.5.0 (values 0-1) ## J129 SIP R1.0.0.0 (or R1.1.0.0) (values 0-1) ## H1xx SIP R1.0.2 and later (values 0-1) ## SET ENABLE_IPOFFICE 1 ## ## MEDIA_PRESERVATION specifies whether a call will be preserved when there is no SIP connectivity to IP Office. ## This parameter is only applicable when ENABLE_IPOFFICE is set to 2. ## Value Operation ## 0 Phone will not preserve a call. As soon as the phone detects SIP connectivity failure to IP Office, phone will drop a call and make re-registration attempt. ## 1 Phone will try to preserve a call for a duration specified by PRESERVED_CALL_DURATION settings parameter (Default). ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## SET MEDIA_PRESERVATION 0 ## ## PRESERVED_CALL_DURATION specifies how long the call will be preserved if ENABLE_IPOFFICE is set to 2 and if MEDIA_PRESERVATION is set to 1. ## In such case, the call will be preserved for a duration of PRESERVED_CALL_DURATION minutes. Valid values are 10-120. Default value is 120 minutes. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## SET PRESERVED_CALL_DURATION 10 ## ## SUBSCRIBE_LIST_NON_AVAYA specifies comma separated list of event packages to subscribe to after registration. ## Possible values are: "reg", "dialog", "mwi", "ccs", "message-summary" which is identical to "mwi", "avaya-ccs-profile" which is identical to "ccs" ## The values are case insensitive. ## For IPO the recommended value shall be "reg, message-summary, avaya-ccs-profile". ## For 3PCC environment the value "message-summary" may be required. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## Avaya Vantage Basic Application SIP R1.1.0.1 and later ## H1xx SIP R1.0.2 and later ## SET SUBSCRIBE_LIST_NON_AVAYA "reg, message-summary, avaya-ccs-profile" ## ## USER_STORE_URI for User Data ## URI used for HTTP/S backup and retrieval of user data. ## Specify HTTP/S server and directory path to backup file. ## Do not specify backup file name. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.1.0.1 and later ## J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later ## SET USER_STORE_URI https://192.168.0.28 ## Note: This parameter is supported by Avaya Vantage Devices SIP R1.1.0.1 and later to define user store URI for personal/enterprise contacts. Used in IP Office environment only. ## The default ports are 80 for HTTP and 443 for HTTPS. When "Use Preferred Port" is enabled in IP Office 11.0 and later then the ports are changed to 8411 for HTTP and 411 for HTTPS. ## SET USER_STORE_URI https://192.168.0.28:411 ## SET USER_STORE_URI http://192.168.0.28:8411 ## Note: This parameter is supported by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later for 3PCC Environment and IP Office R10.1 and later. ## ## ENABLE_OOD_RESET_NOTIFY specifies whether the phone supports out of dialog (OOD) SIP NOTIFY message with ## Event:resync or Event:check-sync only. The events are used to remotely restart the phone (once all calls end). ## The parameter is used with 3PCC environment only. ## Value Operation ## 0 OOD is not supported (Default) ## 1 OOD is supported ## This parameter is supported by: ## J129 J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## SET ENABLE_OOD_RESET_NOTIFY 1 ## #################### 3PCC ENVIRONMENT SETTINGS #################### ## ## ENABLE_3PCC_ENVIRONMENT specifies whether the deployment environment is third party SIP Server ## Value Operation ## 0 Not 3PCC environment ## 1 3PCC environment (Default) ## Note: This parameter should be set to '0' for Aura environment and IP Office ## This parameter is supported by: ## J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## SET ENABLE_3PCC_ENVIRONMENT 0 ## ## 3PCC_SERVER_MODE specifies if the phone expects a generic 3PCC server or a BroadSoft server (applicable when ENABLE_3PCC_ENVIRONMENT is set to 1). ## Value Operation ## 0 Generic (Default) ## 1 BroadSoft ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET 3PCC_SERVER_MODE 1 ## ######################## BROADSOFT CALL FEATURES SETTINGS ################## ## ## The parameters below are applicable when 3PCC_SERVER_MODE=1 and ENABLE_3PCC_ENVIRONMENT=1. ## ## BLF_LIST_URI specifies the BroadSoft Busy Lamp Field (BLF) Resource URI. It defines the unique name for the list of users to be monitored.   ## BLF_LIST_URI must be configured to enable BLF feature unless Xtended Services Interface (XSI) is enabled. When XSI_URL parameter is defined, the phone retrieves BLF configuration from XSP server and ## ignores BLF_LIST_URI value. The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET BLF_LIST_URI sip:mylist1@as.iop1.broadworks.net ## SET BLF_LIST_URI mylist1@as.iop1.broadworks.net ## SET BLF_LIST_URI sip:mylist1 ## SET BLF_LIST_URI mylist1 ## ## CALL_PICKUP_FAC specifies the Directed Call Pickup feature access code that phone will use to pick up the call for a BroadSoft Busy Lamp Field (BLF) monitored station. ## The parameter should be provided in BroadSoft environment unless Xtended Services Interface (XSI) is enabled. When XSI_URL parameter is defined, the phone retrieves BLF configuration from ## BroadWorks Xtended Service Platform (XSP) server and ignores CALL_PICKUP_FAC value. The default is "*97". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET CALL_PICKUP_FAC *98 ## ## CALL_PICKUP_BARGEIN_FAC specifies the Directed Call Pickup with Barge-in feature access code that phone will use to barge into a call between a remote user ## and BroadSoft Busy Lamp Field (BLF) monitored station. This parameter should be provided in BroadSoft environment unless Xtended Services Interface (XSI) is enabled. When XSI_URL parameter is defined, ## the phone retrieves BLF configuration from BroadWorks Xtended Service Platform (XSP) server and ignores CALL_PICKUP_BARGEIN_FAC value. The default value is "*33". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET CALL_PICKUP_BARGEIN_FAC *32 ## ## CALL_UNPARK_FAC specifies the Call Retrieve feature access code that phone will use to unpark the call that has been parked for a BroadSoft Busy Lamp Field (BLF) monitored station. ## This parameter should be provided in BroadSoft environment unless Xtended Services Interface (XSI) is enabled. When XSI_URL parameter is defined, the phone retrieves ## BLF configuration from BroadWorks Xtended Service Platform (XSP) server and ignores CALL_UNPARK_FAC value. The default is "*88". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET CALL_UNPARK_FAC *89 ## ## ALLOW_BLF_LIST_CHANGE specifies the user permissions for adding/removing BroadSoft Busy Lamp Field (BLF) monitored users from the phone. ## Value Operation ## 0 User is not allowed to add/delete BLF monitored users ## 1 User is allowed only to delete BLF monitored users ## 2 User is allowed only to add BLF monitored users ## 3 User is allowed to add and delete BLF monitored users (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET ALLOW_BLF_LIST_CHANGE 2 ## ###################### BROADSOFT XTENDED SERVICES INTERFACE (XSI) SETTINGS ################## ## ## The parameters below are applicable when 3PCC_SERVER_MODE=1 and ENABLE_3PCC_ENVIRONMENT=1. ## ## XSI_URL specifies BroadWorks Xtended Service Platform (XSP) server FQDN/IP address, HTTP or HTTPS mode and port. If port is not defined, 80 is used for HTTP and 443 for HTTPS by default. ## The default is "". ## This is the main parameter to make features work in BroadSoft environment. ## If value of this parameter is non-empty, phone will initiate Xtended Services Interface (XSI) connections establishment to retrieve feature list. ## Note: If XSI_URL is defined the following local call features will not be available. They must be enabled for the user on the BroadSoft server: ## - Do Not Disturb (ENABLE_DND, ENABLE_DND_PRIORITY_OVER_CFU_CFB) ## - Call Forward CFA, CFB, CFNA ## - Auto Answer (ENABLE_AUTO_ANSWER_SUPPORT, AUTO_ANSWER_MUTE_ENABLE) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET XSI_URL http://xsp1.iop2.broadworks.net ## SET XSI_URL https://xsp1.iop2.broadworks.net:443 ## SET XSI_URL http://192.168.111.111:8080 ## SET XSI_URL https://192.168.111.111 ## ## XSI_CHANNEL_DURATION defines the time duration in minutes for Xtended Services Interface (XSI) event channel, i.e. phone will ask BroadWorks Xtended Service Platform (XSP) server ## to maintain the established Comet HTTP connection for the specified period of time. After 50% of this time phone will reestablish Comet HTTP connection. ## The values range is 60-1440. The default is 60. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET XSI_CHANNEL_DURATION 100 ## ## XSI_HEARTBEAT defines the interval in seconds to send heartbeat messages over Comet HTTP connection to BroadWorks Xtended Service Platform (XSP) server. ## Ideally XSI_HEARTBEAT should be configured to BroadSoft’s eventTimeout/2. The eventTimeout value is configurable on ## BroadSoft, by default it is 30 seconds. ## The values range is 1-999. The default value is 15. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET XSI_HEARTBEAT 20 ## ## FORCE_XSI_USER_ID specifies the BroadSoft's User Id which phone should use for Xtended Services Interface (XSI) authentication (SIP or Web methods). ## Note: It shall be User Id without @ and domain. The default is "" ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET FORCE_XSI_USER_ID userhandle ## ## FORCE_XSI_WEB_PASSWORD specifies the BroadSoft's Web portal password which phone should use for Xtended Services Interface (XSI) authentication (Web method). ## If empty, it means that SIP authentication method should be used. The default is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET FORCE_XSI_WEB_PASSWORD userpassword ## ###################### BROADSOFT SHARED CALL APPEARANCE SETTINGS ################## ## ## The parameters below are applicable when 3PCC_SERVER_MODE=1 and ENABLE_3PCC_ENVIRONMENT=1. ## ## PRIMARY_LINE_TYPE specifies if the phones primary line is a private or shared line. The primary line is the one associated with the user's login credentials. ## Value Operation ## 0 Primary line is a private line (default) ## 1 Primary line is a shared line ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET PRIMARY_LINE_TYPE 1 ## ## PRIMARY_LINE_BARGE_IN_ENABLED specifies whether a primary line which is shared is configured in the BroadSoft server to either enable or disable the ability ## of a user to barge into a call at a different location on the shared line. ## This setting is ignored if PRIMARY_LINE_TYPE = 0 ## Value Operation ## 0 Barge in is disabled for the shared line ## 1 Barge in is enabled for the shared line (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET PRIMARY_LINE_BARGE_IN_ENABLED 0 ## ## SCA1_ENABLED, SCA2_ENABLED, SCA3_ENABLED specifies if first, second or third Shared Call Appearance (SCA) is/are enabled. ## For example, SCA1_ENABLED defines if the first shared line is enabled. ## Value Operation ## 0 The shared line is disabled (default) ## 1 The shared line is enabled ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_ENABLED 1 ## SET SCA2_ENABLED 1 ## SET SCA3_ENABLED 1 ## ## SCA1_MAX_CALL_APPEARANCES, SCA2_MAX_CALL_APPEARANCES, SCA3_MAX_CALL_APPEARANCES specify the maximum number of simultaneous calls on the first, second and third shared line ## respectively. Value range 1-5. Default value is 3. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_MAX_CALL_APPEARANCES 4 ## SET SCA2_MAX_CALL_APPEARANCES 5 ## SET SCA3_MAX_CALL_APPEARANCES 2 ## ## SCA1_SIPUSERID, SCA2_SIPUSERID, SCA3_SIPUSERID specify the AOR(Address of Record) for first, second and third shared line respectively. ## It should only specify the handle (e.g. 123456_2) since the domain is specified independently. ## Note that shared lines are only supported for the same SIP domain as the primary line. ## The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_SIPUSERID 6459137 ## SET SCA2_SIPUSERID 6459138 ## SET SCA3_SIPUSERID 6459139 ## ## SCA1_USERNAME, SCA2_USERNAME, SCA3_USERNAME specify the username to be used for authentication when challenged for credentials on SIP requests associated with ## the first, second and third shared line respectively. ## This value is optional and if not specified the SCA1_SIPUSERID, SCA2_SIPUSERID, SCA3_SIPUSERID would be used for authentication. ## The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_USERNAME 6459137 ## SET SCA2_USERNAME 6459138 ## SET SCA3_USERNAME 6459139 ## ## SCA1_PASSWORD, SCA2_PASSWORD, SCA3_PASSWORD specify the password to be used for authentication when challenged for credentials on SIP requests associated ## with the first, second and third shared line respectively. The default is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_PASSWORD sca1pass ## SET SCA2_PASSWORD sca1pass ## SET SCA3_PASSWORD sca1pass ## ## SCA1_EXTENSION, SCA2_EXTENSION, SCA3_EXTENSION specify the display name for the first, second and third shared line respectively. ## If not specified the SCA1_SIPUSERID, SCA2_SIPUSERID, SCA3_SIPUSERID would be used. The default is "". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_EXTENSION UsernameA ## SET SCA2_EXTENSION UsernameB ## SET SCA3_EXTENSION UsernameC ## ## SCA1_BARGE_IN_ENABLED, SCA2_BARGE_IN_ENABLED, SCA3_BARGE_IN_ENABLED specify for each shared line in the BroadSoft server to either enable or disable the ## ability of a user to barge into a call at a different location on the first, second, third shared line respectively. ## Value Operation ## 0 Barge in is disabled for the shared line ## 1 Barge in is enabled for the shared line (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA1_BARGE_IN_ENABLED 0 ## SET SCA2_BARGE_IN_ENABLED 0 ## SET SCA3_BARGE_IN_ENABLED 0 ## ## SCA_LINE_SEIZE_DURATION specifies the length of time in seconds to be used for a line-seize subscription on any (first, second, third) shared line or on a primary shared line. ## Value range is 5-40. The default is 15. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET SCA_LINE_SEIZE_DURATION 0 ## ## PROVIDE_SHARED_LINE_CONFIG specifies if the user has the ability to change Shared Line configuration using the Settings menu on the phone. ## Value Operation ## 0 Shared lines is not displayed in settings menu ## 1 Shared lines is displayed in settings menu but all information is read-only ## 2 Shared lines is displayed in settings menu and is fully configurable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (J129 and J139 do not support this parameter) ## SET PROVIDE_SHARED_LINE_CONFIG 0 ## ###################### BROADWORKS DIRECTORY SETTINGS ################## ## ## The parameters below are applicable when 3PCC_SERVER_MODE=1 and ENABLE_3PCC_ENVIRONMENT=1. ## ## BW_ENABLE_DIR specifies BroadWorks Directory feature availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR 0 ## ## BW_ENABLE_DIR_ENTERPRISE specifies BroadWorks Enterprise directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_ENTERPRISE 0 ## ## BW_ENABLE_DIR_ENTERPRISE_COMMON specifies BroadWorks Enterprise Common directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_ENTERPRISE_COMMON 0 ## ## BW_ENABLE_DIR_GROUP specifies BroadWorks Group directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_GROUP 0 ## ## BW_ENABLE_DIR_GROUP_COMMON specifies BroadWorks Group Common directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_GROUP_COMMON 0 ## ## BW_ENABLE_DIR_PERSONAL specifies BroadWorks Personal directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_PERSONAL 0 ## ## BW_ENABLE_DIR_CUSTOM specifies BroadWorks Custom directory availability state. ## Value Operation ## 0 disable ## 1 enable (default) ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_ENABLE_DIR_CUSTOM 0 ## ## BW_DIR_ENTERPRISE_DESCRIPTION specifies the display name for BroadWorks Enterprise directory. ## The default is "Enterprise". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_ENTERPRISE_DESCRIPTION "LargeEnterprise" ## ## BW_DIR_ENTERPRISE_COMMON_DESCRIPTION specifies the display name for BroadWorks Enterprise Common directory. ## The default is "Enterprise Common". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_ENTERPRISE_COMMON_DESCRIPTION "DirectoryName" ## ## BW_DIR_GROUP_DESCRIPTION specifies the display name for BroadWorks Group directory. ## The default is "Group". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_GROUP_DESCRIPTION "Sales" ## ## BW_DIR_GROUP_COMMON_DESCRIPTION specifies the display name for BroadWorks Group Common directory. ## The default is "Group Common". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_GROUP_COMMON_DESCRIPTION "SalesTeam" ## ## BW_DIR_PERSONAL_DESCRIPTION specifies the display name for BroadWorks Personal directory. ## The default is "Personal". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_PERSONAL_DESCRIPTION "PersonalList" ## ## BW_DIR_CUSTOM_DESCRIPTION specifies the display name for BroadWorks Custom directory. ## The default is "Custom". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_CUSTOM_DESCRIPTION "CustomDescription" ## ## BW_DIR_ENTERPRISE_EXTENSION specifies the display name for BroadWorks Enterprise directory extension. ## The default is "BWEntr". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_ENTERPRISE_EXTENSION "BWEntrA" ## ## BW_DIR_ENTERPRISE_COMMON_EXTENSION specifies the display name for BroadWorks Enterprise Common directory extension. ## The default is "BW EnCom". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_ENTERPRISE_COMMON_EXTENSION "BW EnCom1" ## ## BW_DIR_GROUP_EXTENSION specifies the display name for BroadWorks Group directory extension. ## The default is "BW Group". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_GROUP_EXTENSION "BW GroupA" ## ## BW_DIR_GROUP_COMMON_EXTENSION specifies the display name for BroadWorks Group Common directory extension. ## The default is "BW GrCom". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_GROUP_COMMON_EXTENSION "BW GrComA" ## ## BW_DIR_PERSONAL_EXTENSION specifies the display name for BroadWorks Personal directory extension. ## The default is "BW Pers". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_PERSONAL_EXTENSION "BW PersA" ## ## BW_DIR_CUSTOM_EXTENSION specifies the display name for BroadWorks Custom directory extension. ## The default is "BW Cust". ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later (Not supported by J129) ## SET BW_DIR_CUSTOM_EXTENSION "BW CustA" ## #################### DISPLAY SETTINGS #################### ## ## Display Colors and Layout ## Specifies a list of tuples describing color scheme and ## layout used in phone display. See Administrator's guide ## for additional detail. (0 to 1023 ASCII characters) ## This parameter is supported by: ## 96x0 SIP. ## SET SKINS Yankees=http://mycompany.com/skins/yankees_color/pinstripes.xml ## ## Selected skin for display layout ## If CURRENT_SKIN is selected(not empty string), then that particular skin is selected ## for display. This parameter should be one of the label as defined in 'SKINS' ## configuration parameter. If it is empty or not set then default skin is used. ## This parameter is supported by: ## 96x0 SIP. ## SET CURRENT_SKIN "" ## ## Display Logo (96x1) / Wallpaper (H1xx/Avaya Vantage) ## Specifies a list of tuples describing logo/wallpaper used as phone ## display background. See Administrator's guide for ## additional detail. ## This parameter is supported by: ## J169/J179 SIP R1.5.0 - Only Full path URLs are supported (relative paths are not supported). ## The Maximum size (pixels) is 320 x 240 (color depth 16 bit for J179) and JPG file type. ## 96x1 SIP R6.0 and later. Only Full path URLs are supported (relative paths are not supported). ## The models supported are: 9611G, 9621G and 9641G. The Maximum size (pixels) are: 217 x 130, ## 232 x 140 and 232 x 140 respectively with color depth 16 bit and JPG and PNG file types. ## For JPG files, in order to invoke transparent backgrounds with logos, use a background color of 0,255,0 (brightest possible green). ## For PNG files, the transparency setting supported in the PNG file format are used. ## H1xx SIP R1.0.1 and later. LOGOS defines list of administrator wallpapers. ## For best results, H175 Wallpapers resolution shall be 1280x800 with 24 bits color depth. ## The following file types are supported by H175: PNG, JPG (JPEG), GIF and BMP (GIF is presented without animation). ## Avaya Vantage Devices SIP R1.0.0.2 and later. LOGOS defines list of administrator wallpapers. ## For best results, Avaya Vantage Wallpapers resolution shall be 1280x800 with 24 bits color depth. ## The following file types are supported by Avaya Vantage: PNG, JPG (JPEG), GIF and BMP (GIF is presented without animation). ## Note: LOGOS is not supported by J100 SIP R2.0.0.0 and later. Please refer to BACKGROUND_IMAGE. ## SET LOGOS FIFAWorldCup=../fifa_logo.jpg ## SET LOGOS FIFAWorldCup=http://10.11.12.13/logo.jpg ## SET LOGOS FIFAWorldCup=http://logos.com/logo.jpg ## ## Selected background logo (96x1)/ Wallpaper (H1xx/Avaya Vantage) on display ## CURRENT_LOGO defines if custom logo/wallpaper is selected for display. ## This is used to display custom logo/wallpaper or built in default logo/wallpaper is to be used. ## If CURRENT_LOGO is selected (not empty string), then the resource should be ## available using "LOGOS" configuration parameter. ## The default value is "" where Avaya Logo is displayed. ## The CURRENT_LOGO configured in the settings file is used in the following cases: ## 1. The phone is not registered to Avaya Aura Session Manager ## 2. If the phone is registered to Avaya Aura Session Manager AND ## A. there is no information stored for the current logo file for this specific user (first time login of this user) ## AND ## B. there is no support of "Profile Settings" in the "Endpoint Template" (which is supported by SMGR 6.3.8 and up). ## This parameter is supported by: ## J169/J179 SIP R1.5.0 - "none" is used for no logo/wallpaper display (Only Time/date is displayed). ## 96x1 SIP R6.0 and later; 96x1 SIP R7.1.0.0 and later - "none" is used for no logo/wallpaper display (Only Time/date is displayed). ## H1xx SIP R1.0.1 and later. CURRENT_LOGO defines the administrator choice of wallpaper. ## Avaya Vantage Devices SIP R1.0.0.2 and later. CURRENT_LOGO defines the administrator choice of wallpaper. The wallpaper will appear on Android home screen and on the ## Avaya Vantage Android Launcher application (when the latter is used in Kiosk mode). ## SET CURRENT_LOGO "FIFAWorldCup" ## ## Access privileges for Wallpaper configuration ## LOGOSTAT defines wallpaper configuration is allowed for administrators only or both users and administrators. ## Value Operation ## 0 Wallpaper configuration is defined according to CURRENT_LOGO only (CURRENT_LOGO can be defined by administrator only). Users are not allowed to change ## wallpaper configuration in the settings application. ## 1 The user is given an option to choose wallpaper in the settings application. By default, CURRENT_LOGO will be used but users can override this configuration. ## Once users override this configuration, CURRENT_LOGO will not be used unless the device return to factory defaults or LOGOSTAT is changed to 0 (default). ## This parameter is supported by: ## H1xx SIP R1.0.1 and later. ## SET LOGOSTAT 0 ## ## BACKGROUND_IMAGE specifies a list of background images. The default value is "". ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## Up to 5 background images are supported. Only jpeg/jpg files are supported. ## The maximum size of any jpeg file is 256 KB. The filenames are case insensitive. ## J169/J179 screen resolution is 320 pixels x 240 pixels. J179 color depth is 16 bits. ## J129 screen resolution is 128 pixels x 32 pixels. ## The files shall be stored in the same directory defined by HTTPDIR / TLSDIR. ## SET BACKGROUND_IMAGE "background_example1.jpg,background_example2.jpeg" ## ## BACKGROUND_IMAGE_DISPLAY specifies the administrator choice of background image. ## The filename shall be one of the filenames listed in BACKGROUND_IMAGE. ## If BACKGROUND_IMAGE_SELECTABLE is set to 1 then the end user may override this setting. ## The default value is "". ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## SET BACKGROUND_IMAGE_DISPLAY background_example1.jpg ## ## BACKGROUND_IMAGE_SELECTABLE specifies whether end users are allowed to choose background images ## (and overrides administrator choice as configured using BACKGROUND_IMAGE_DISPLAY parameter). ## Value Operation ## 0 End user is not allowed to choose background image and will not see the background image selection in the Settings -> Display menu. ## 1 End user is allowed to choose the background image from the Settings -> Display menu (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J169 and J179 only) ## SET BACKGROUND_IMAGE_SELECTABLE 0 ## ## BRANDING_FILE specifies the branding file to be downloaded by the Avaya Vantage Basic Application ## and to be presented on the top left corner of the application. Up to one URL shall be specified. The default value ## is "" (in such case Avaya logo will be displayed). URL shall be absolute path (start with http:// or https://). ## The resolution of the file shall be 142x56. The file types supported are PNG, JPG (JPEG), GIF and BMP. ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET BRANDING_FILE "http://www.spacegrant.org/sg_graphics/animated_nasa_meatball.gif" ## ## HOME_SCREEN_GRID_SIZE specifies the icon grid size in the home screen ## Value Operation ## 1 6x3 icons (width x height), Default. ## 2 4x2 icons (width x height) ## This parameter is supported by: ## H1xx SIP R1.0.2 and later. ## SET HOME_SCREEN_GRID_SIZE 2 ## ## ADMIN_INITIAL_SCREEN specifies whether home screen or phone screen is presented when all calls end or after login. This configuration is only ## enforced if "Screen presented when all calls end or after login" field (in the settings application --> Call settings menu) is configured as "Admin Default" (default). ## Value Operation ## HOMESCREEN Home screen is presented when all calls end or after login. ## PHONE Phone screen is presented when all calls end or after login (Default). ## Please note that if the screen before all call ends was NOT phone screen, then this parameter (including the field "Screen presented when all calls end or after login") ## will not have any effect since the user choose to work with other application (for example, browser) during the call and the preference is to avoid changing the screen in such case. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; the parameter is not supported when Avaya Vantage Open is used. The parameter is applicable when Avaya Breeze Client SDK is installed ## (including Avaya Vantage Basic application and Avaya Equinox). ## H1xx SIP R1.0.2 and later. ## SET ADMIN_INITIAL_SCREEN HOMESCREEN ## ## EXTENSION_ON_TOP_LINE Specifies whether extension shall be displayed on top line or not. ## Value Operation ## 1 – Extension is NOT presented on the top line (default). ## 2 – Extension is presented on the top line. ## This parameter is supported by: ## H1xx SIP R1.0.2 and later. ## SET EXTENSION_ON_TOP_LINE 2 ## ## EXTENSION_NAME_DISPLAY_OPTIONS specifies whether extension or name are presented in top right corner of Avaya Vantage Basic application screen. ## Value Operation ## 0 Presents both extension and name (as in pre R1.1.0.0), default. ## 1 Presents name only ## 2 Presents extension only ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.1.0.0 and later. ## SET EXTENSION_NAME_DISPLAY_OPTIONS 1 ## ## Options Menu Display ## Determines whether Options & Settings menu is displayed ## on phone. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET PROVIDE_OPTIONS_SCREEN 1 ## ## Network Info Menu Display ## Determines whether Network Information menu is displayed ## on phone. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET PROVIDE_NETWORKINFO_SCREEN 1 ## ## Logout Enabled ## Determines whether user can log out from phone. ## 0 for No ## 1 for Yes ## SET PROVIDE_LOGOUT 1 ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## Determines whether log out option is available or not in Avaya Menu options. ## ## DISPLAY_SSL_VERSION - display version of OpenSSH/OpenSSL ## Value Operation ## 0 No display of OpenSSH/OpenSSL version (default) ## 1 Display of OpenSSH/OpenSSL version ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96x1 SIP R7.1.0.0 and later ## SET DISPLAY_SSL_VERSION 1 ## ## HOMESCREENLAYOUT specifies home screen layout. ## Value Operation ## 0 Top of mind (default) ## 1 Top of mind (as value 0) ## 2 Top Of Mind Lite ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## SET HOMESCREENLAYOUT 2 ## ## SHOW_EQUINOX_MEETING_PANEL_IN_TOM specifies whether to show the "My Meeting Room" panel in the "Top Of Mind" page as default ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## Avaya Equinox 3.2 and later ## SET SHOW_EQUINOX_MEETING_PANEL_IN_TOM 0 ## #################### CALL LOG SETTINGS ################### ## ## Call Log Enabled ## Determines whether call logging and associated menus ## are available on the phone. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later and Avaya Vantage Basic Application SIP R1.0.0.1 and later. ## SET ENABLE_CALL_LOG 1 ## ## Redial Enabled ## Determines whether redial softkey is available. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later and ## Avaya Vantage Basic Application SIP R1.0.0.0 and later. ## SET ENABLE_REDIAL 1 ## ## Redial List Enabled ## Determines whether phone redials last number or ## displays list of recently dialed numbers. ## 0 for last number redial ## 1 user can select between last number redial and ## redial list (default). By default, user choice is "last number redial". ## Note: This parameter is also supported by J100 SIP R2.0.0.0 and later (J169/J179 only) ## SET ENABLE_REDIAL_LIST 1 ## ## ENABLE_PPM_CALL_JOURNALING specifies whether enable or disable call journaling. ## Value Operation ## 0 Disabled ## 1 Enabled (Default) ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.1.0.0; ENABLE_PPM is not supported by Avaya Vantage Basic application. ## Avaya Equinox 3.2 and later ## SET ENABLE_PPM_CALL_JOURNALING 0 ## #################### CONTACTS SETTINGS ################### ## ## Contacts Enabled ## Determines whether the contacts application and ## associated menus are available on the phone. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later and Avaya Vantage Basic Application SIP R1.0.0.1 and later. ## Note: This parameter is also supported by H1xx R1.0.1 SIP and later, but it controls only ## the "Contacts" virtual button LED whether it is dimmed and pressing on it has no effect (ENABLE_CONTACTS==0) or ## whether "Contacts" virtual button LED is ON and pressing on it has effect (ENABLE_CONTACTS==1, default). ## SET ENABLE_CONTACTS 1 ## ## Contacts Modification Enabled ## Determines whether the list of contacts and ## the function of the contacts application can ## be modified on the phone. ## 0 for No ## 1 for Yes ## Note: This parameter is also supported Avaya Vantage Basic Application SIP R1.0.0.0 and later. ## SET ENABLE_MODIFY_CONTACTS 1 ## ## Multiple Contacts Warning Display ## Determines whether a warning message is displayed if ## there are multiple devices registered on a user's ## behalf. Multiple registered devices may lead to ## service disruption. ## 0 for No ## 1 for Yes ## SET ENABLE_MULTIPLE_CONTACT_WARNING 1 ## ## CONTACT_NAME_FORMAT specifies how contact names are displayed. ## Value Operation ## 0 "Last Name, First Name" (Default) ## 1 "First Name Last Name" ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## SET CONTACT_NAME_FORMAT 0 ## ## NAME_SORT_ORDER specifies how contact names are sorted. ## Value Operation ## "last,first" Sorting according to "Last Name" and then "First Name" (Default) ## "first,last" Sorting according to "First Name" and then "Last Name" ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET NAME_SORT_ORDER "first,last" ## ## NAME_DISPLAY_ORDER specifies how contact names are displayed. ## Value Operation ## "last,first" Display "Last Name First Name" ## "first,last" Display "First Name Last Name" (Default) ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## Avaya Vantage Basic Application SIP R1.0.0.0 and later ## SET NAME_DISPLAY_ORDER "first,last" ## ## PHONE_NUMBER_PRIORITY specifies the default phone number priority. This parameter is a comma separated list of the following strings: ## "work", "mobile" and "home". The parameter defines the priority order (left to right) in which these numbers will be used. ## The default value is: "work,mobile,home" ## This parameter is supported by: ## Avaya Equinox 3.1.2 and later; ## SET PHONE_NUMBER_PRIORITY "mobile,work,home" ## ## ENABLE_FAVORITES specifies whether favorites tab is displayed. ## Value Operation ## 0 No display of favorites tab. ## 1 Display of favorites tab (default) ## This parameter is supported by: ## Avaya Vantage Basic Application SIP R1.0.0.1 and later ## SET ENABLE_FAVORITES 0 ## #################### LANGUAGE SETTINGS #################### ## ## System-Wide Language ## Contains the name of the default system language file ## used in the phone. The filename should be one of the ## files listed in the LANGUAGES parameter. If no ## filename is specified, or if the filename does not ## match one of the LANGUAGES values, the phone shall use ## its built-in English text strings. 0 to 32 ASCII ## characters. Filename must end in .xml ## ## NOTE: ## For 96xx SIP Release 1.0 phones only, all language ## filenames begin with Mls_Spark_. For example, ## Mls_Spark_English.xml ## ## For 96xx SIP Release 2.0 and later, ## all language filenames begin with Mlf_ ## ## SET SYSTEM_LANGUAGE Mlf_English.xml ## ## The language files of 96x0 SIP 2.6.13 and later in the 96x0 SIP firmware distributions are different than 96x1 ## and therefore their filenames were changed to Mlf_S96x0_.xml. ## Mlf_.xml filename convention is used by: ## 1. 96x1 SIP Release 6.0 and later and ## 2. 96xx SIP Release 2.0 up to 2.6.13 (excluded). ## In mutual environment of 96x0 SIP and 96x1 SIP phones there shall be use of IF conditional statement ## base on MODEL/GROUP to assign different language files for each phone family. ## Note: SYSTEM_LANGUAGE is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later as 96x1 SIP phones. ## SET SYSTEM_LANGUAGE Mlf_English.xml ## SET SYSTEM_LANGUAGE Mlf_S96x0_English.xml ## ## Installed Languages ## Specifies the language files to be installed/downloaded ## to the phone. Filenames may be full URL, relative ## pathname, or filename. (0 to 1096 ASCII characters, ## including commas). Filenames must end in .xml. ## ## NOTE: ## For 96xx SIP Release 1.0 phones only, all language ## filenames begin with Mls_Spark_ For example, ## Mls_Spark_English.xml ## ## For 96xx SIP Release 2.0 and later, ## all language filenames begin with Mlf_ ## ## The language files of 96x0 SIP 2.6.13 and later in the 96x0 SIP firmware distributions are different than 96x1 ## and therefore their filenames were changed to Mlf_S96x0_.xml. ## Mlf_.xml filename convention is used by: ## 1. 96x1 SIP Release 6.0 and later and ## 2. 96xx SIP Release 2.0 up to 2.6.13 (excluded). ## In mutual environment of 96x0 SIP and 96x1 SIP phones there shall be use of IF conditional statement ## base on MODEL/GROUP to assign different language files for each phone family. ## Note: LANGUAGES is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later as 96x1 SIP phones. ## ## SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml ## SET LANGUAGES Mlf_S96x0_German.xml,Mlf_S96x0_ParisianFrench.xml,Mlf_S96x0_LatinAmericanSpanish.xml ## #################### ISO LANGUAGE SETTINGS #################### ## ## ISO System-Wide Language ## Contains the language and country codes for the administrator language choice ## until user changes the language using settings application. ## The language codes are two-letter lowercase ISO language codes (such as "en") ## as defined by ISO 639-1 (LL). The optional country codes (CC) are two-letter uppercase ## ISO country codes (such as "US") as defined by ISO 3166-1. ## The default value is en_US. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; Supported languages: Brazilian Portuguese, English, French, German, Italian, Japanese, Korean, ## Latin Spanish, Russian and Simplified Chinese. ## Examples are: en_AU, en_CA, en_IN, en_NZ, en_SG, en_GB, en_US,es_US, fr_BE, fr_CA, fr_FR, fr_CH, pt_BR, ru_RU, etc. ## H1xx SIP R1.0 and later; supported values: ar_SA, en_AU, en_CA, en_IN, en_NZ, en_SG, en_GB, ## en_US,es_US, fr_BE, fr_CA, fr_FR, fr_CH, pt_BR, ru_RU ## SET ISO_SYSTEM_LANGUAGE en_AU ## #################### COUNTRY AND DATE SETTINGS #################### ## ## Call Progress Tone Country ## Country used for network call progress tones. ## For Argentina use keyword "Argentina" ## For Australia use keyword "Australia" ## For Brazil use keyword "Brazil" ## For Canada use keyword "USA" ## For France use keyword "France" ## For Germany use keyword "Germany" ## For Italy use keyword "Italy" ## For Ireland use keyword "Ireland" ## For Mexico use keyword "Mexico" ## For Spain use keyword "Spain" ## For United Kingdom use keyword "UK" ## For United States use keyword "USA" ## ## NOTE 1: For a complete list of supported countries, see your telephone's Administrators Guide. ## Note 2: Country names with spaces shall be enclosed in double quotes, as in: ## SET COUNTRY "Saudi Arabia" ## NOTE 3: This setting is also applicable for J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. For J100 SIP R2.0.0.0 (J129 and J179 only) please use ## WLAN_COUNTRY for Wi-Fi Regulatory Domain configuration. ## Note 4: This parameter is supported by H1xx SIP R1.0 and later. For H1xx this parameter is used for country configuration for the following: ## a. Call Progress Tones, b. cordless handset, c. Wi-Fi and d. default anti-flickering ("50" or "60" Hz). ## This parameter MUST be configured for cordless handset operation (only certain countries ## are supported with cordless handset. Refer to Administrator guide for the full list). ## The default of this parameter is "Undefined" which means: ## a. Call progress Tones for "USA", b. Cordless handset is disabled, c. Wi-Fi is configured as WorldWide and d. "60 Hz" anti-flickering is used. ## Note 5: This parameter is supported by Avaya Vantage Devices SIP R1.0.0.0 and later. For Avaya Vantage Devices this parameter is used for ## country configuration for Wi-Fi. The default of this parameter is "USA". ## ## SET COUNTRY USA ## ## Date Format ## Specifies the format for dates displayed in the phone. ## Use %d for day of month ## Use %m for month in decimal format ## Use %y for year without century (e.g., 07) ## Use %Y for year with century (e.g., 2007) ## Any character not preceded by % is reproduced exactly. ## SET DATEFORMAT %m/%d/%y ## ## Time Format ## Specifies the format of the time displayed in the phone. ## 0 for am/pm format (Default) ## 1 for 24h format ## The TIMEFORMAT configured in the settings file is used in the following cases: ## 1. The phone is not registered to Avaya Aura Session Manager ## 2. If the phone is registered to Avaya Aura Session Manager AND ## A. there is no information stored for the timeformat for this specific user (first time login of this user) ## AND ## B. there is no support of "Profile Settings" in the "Endpoint Template" (which is supported by SMGR 6.3.8 and up). ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - The TIMEFORMAT can also be set from the phone's local menu ## independent of the environment (3PCC, Aura, IP Office). The TIMEFORMAT will be used in the above cases until user manually changes the value. ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later (up to R1.0.0.0 build 2304). ## SET TIMEFORMAT 1 ## ## Administrator Time Format ## Specifies the format of the time displayed in the phone. By default, time format will be according to ADMINTIMEFORMAT. However, users can change their preference. ## 0 for am/pm format (default) ## 1 for 24h format ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later ## SET ADMINTIMEFORMAT 1 ## ## Daylight Savings Time Mode ## Specifies daylight savings time setting for phone. ## 0 for no daylight saving time ## 1 for daylight savings activated (time set to DSTOFFSET) ## 2 for automatic daylight savings adjustment (as ## specified by DSTSTART and DSTSTOP) ## Note: This parameter is supported by H1xx SIP R1.0 only (TIMEZONE shall be used in R1.0.0.1 and later). ## Note: This parameter is also supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET DAYLIGHT_SAVING_SETTING_MODE 2 ## ##################### PORT SETTINGS (SIP ONLY) ##################### ## ## UDP Minimum Port Value ## Specifies the lower limit of the UDP port range ## to be used by RTP/RTCP or SRTP/SRTCP connections. ## (1024 -65503). ## Note : This setting is also applicable for J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## Avaya Vantage Basic Application SIP R1.0.0.0 and later and Avaya Equinox 3.1.2 and later. ## Note: For H1xx SIP R1.0 and later the first half of the range is used for audio ## and the second half for video. ## SET RTP_PORT_LOW 5004 ## ## UDP Port Range ## Specifies the range or number of UDP ports ## available for RTP/RTCP or SRTP/SRTCP connections. ## This value is added to RTP_PORT_LOW to determine ## the upper limit of the UDP port range (32-64511). ## Note : This setting is also applicable for J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, ## Avaya Vantage Basic Application SIP R1.0.0.0 and later and Avaya Equinox 3.1.2 and later. ## Note: For H1xx SIP R1.0 and later the first half of the range is used for audio ## and the second half for video. ## SET RTP_PORT_RANGE 40 ## ## Signaling Port Minimum Value ## Specifies the minimum port value for SIP ## signaling. ## (1024 -65503). ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - the default and minimum values are 5062. ## J169/J179 SIP R1.5.0 - the default and minimum values are 5062 ## J129 SIP R1.0.0.0 (or R1.1.0.0) - the default is 1024 ## 96xx SIP R2.0 and later ## 96x1 SIP R6.0 and later; Pre R7.1.1.0 the default is 1024. R7.1.1.0.0+ the default and minimum values are 5062. ## H1xx SIP R1.0 and later ## SET SIG_PORT_LOW 1024 ## ## Signaling Port Range ## Specifies the range or number of SIP signaling ## ports. This value is added to SIG_PORT_LOW to ## determine the upper limit of the SIP signaling ## port range (32-64511). ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later - the maximum value is 60473 ## J169/J179 SIP R1.5.0 - the maximum value is 60473 ## J129 SIP R1.0.0.0 (or R1.1.0.0) - the maximum value is 64511 ## 96xx SIP R2.0 and later ## 96x1 SIP R6.0 and later; Pre R7.1.0.0 the maximum value is 64511. R7.1.1.0.0+ the maximum value is 60473. ## H1xx SIP R1.0 and later ## SET SIG_PORT_RANGE 64511 ## ############################################################ ## ## 96xx/96x1/H1xx/J129/J139/J169/J179 SIP TELEPHONE SETTINGS ## ############################################################ ## ## INGRESS_DTMF_VOL_LEVEL specifies the power level of tone, expressed in dBm0. ## The possible values are in the range of -20dBm to -7dBm. ## The default value is -12dBm. ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later ## 96xx SIP R2.0 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## SET INGRESS_DTMF_VOL_LEVEL -12 ## ## CURRENT_CONTENT specifies the URL of an XML file that is used to customize the home screen. ## The default value of the parameter is null. ## This parameter is supported by 96x0 SIP R2.2 and later. ## SET CURRENT_CONTENT http://135.27.67.137/screen.xml ## ############################################# ## ## Conference transfer on primary appearance ## When CONF_TRANS_ON_PRIMARY_APPR is set to 1, ## conference and transfer setup will first attempt ## to use an idle primary call appearance even if ## initiated from a bridged call appearance. ## If an idle primary call appearance is not available, ## then an idle bridged call appearance will be used. ## Conference and transfer setup initiated from a bridged call ## appearance when no idle primary call appearance is available ## will next attempt to use an idle bridged call appearance of ## the same extension and if not available, an idle bridged call ## appearance of a different extension. ## Note: When CONF_TRANS_ON_PRIMARY_APPR is set to 1, AUTO_SELECT_ANY_IDLE_APPR is ignored. ## ## When CONF_TRANS_ON_PRIMARY_APPR is set to 0, ## conference and transfer setup initiated from a primary call ## appearance will first attempt to use an idle primary call appearance. ## If an idle primary call appearance is not available, it will use an idle ## bridged call appearance regardless of the setting of AUTO_SELECT_ANY_IDLE_APPR. ## Conference and transfer setup initiated from a bridged call appearance will attempt ## to use an idle bridged call appearance of the same extension. ## If an idle bridged call appearance of the same extension is not available ## and AUTO_SELECT_ANY_IDLE_APPR is set to 1, then conference and transfer ## setup will use any idle call appearance (primary or bridged). ## It will first attempt to find an idle primary call appearance and if not ## available will then attempt to find an idle bridged call appearance of a different extension. ## However, if AUTO_SELECT_ANY_IDLE_APPR is set to 0, transfer and conference setup ## initiated on a bridged call appearance will be denied if an idle bridged call appearance ## of the same extension is not available. ## ## The Default value of CONF_TRANS_ON_PRIMARY_APPR is 0. ## Note: These parameters are supported on SIP release R2.4.1 and later release of 96xx SIP telephones. ## Note: CONF_TRANS_ON_PRIMARY_APPR is supported by J100 SIP R2.0.0.0 and later (J169/J179 only) and J139 SIP R3.0.0.0 and later. ## ## Auto Select any idle appearance ## When AUTO_SELECT_ANY_IDLE_APPR is active then any idle appearance is selected. ## When AUTO_SELECT_ANY_IDLE_APPR is set to 0 and CONF_TRANS_ON_PRIMARY_APPR is 0, ## then if no associated call appearance is selected, ## the conference or transfer operation will be denied. ## When AUTO_SELECT_ANY_IDLE_APPR is set to 1 and CONF_TRANS_ON_PRIMARY_APPR is 0, ## then if no associated call appearance is selected, the conference or transfer ## operation will be tried on any available call appearance (primary or bridged). ## This parameter is supported by: ## 96x0 SIP R2.4.1 and later releases ## J100 SIP R2.0.0.0 and later (J169/J179 only) and J139 SIP R3.0.0.0 and later. ## SET AUTO_SELECT_ANY_IDLE_APPR 0 ## ## EXTEND_RINGTONE provides a way to customize ring tone files. ## This is a comma separated list of file names in xml format. ## The default value of this parameter is null. ## This parameter is supported by: ## 96x0 SIP R2.4.1 and later releases ## 96x1 SIP R6.0 and later releases ## SET EXTEND_RINGTONE "" ## ## Selection of Active Controller ## When FAILBACK_POLICY parameter is set to "auto", the phone's active controller will ## always be the highest priority available controller. ## If FAILBACK_POLICY parameter is set to "admin", ## then a controller lower down the priority list may be active. ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones. ## The parameter is not supported on 96x1 R6.2 and later. ## SET FAILBACK_POLICY auto ## ## WAIT_FOR_CALL_OPERATION_RESPONSE specifies the time in seconds before providing a response for user initiated call operation.  ## This parameter is applicable to all server environments (Aura, IP Office and 3PCC). ## When user goes off-hook, then phone sends an invite. If there is no response from the SIP proxy for the number of seconds defined in WAIT_FOR_CALL_OPERATION_RESPONSE, ## it will result in a user notification that the operation is in progress but is delayed. ## Value range is 1-4. The default is 3. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later ## SET WAIT_FOR_CALL_OPERATION_RESPONSE 2 ## ## Dynamic Feature Set Discovery ## If the DISCOVER_AVAYA_ENVIRONMENT parameter value is 1, the phone discovers (determines) ## if that controller supports the AST feature set or not. The phone will send a SUBSCRIBE ## request to the active controller for the Feature Status Event Package (avaya-cm-feature-status). ## If the request succeeds, then the phone proceeds with PPM Synchronization. ## If the request is rejected, is proxied back to the phone or does not receive a response, ## the phone will assume that AST features are not available. ## If the parameter value is 0, the phone operates in a mode where AST features are not available. ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones, H1xx SIP R1.0 and later ## and J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later. ## For IP office and 3PCC environments this parameter shall be set to 0. ## SET DISCOVER_AVAYA_ENVIRONMENT 1 ## ## Telephone number to call into the messaging system ## PSTN_VM_NUM is the "dialable" string is used to call into the messaging system ## (e.g. when pressing the Message Waiting button). ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones, H1xx SIP R1.0 and later, ## Avaya Vantage Basic Application SIP R1.1.0.1 and later and J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## PSTN_VM_NUM shall be used instead of MSGNUM in cases of IP Office environment, 3PCC SIP environment or when there is failover from Aura environment to a non-Aura server. ## SET PSTN_VM_NUM "" ## ## PSTN Access Prefix ## ENABLE_REMOVE_PSTN_ACCESS_PREFIX parameter allows telephone to ## perform digit manipulation during failure scenarios. This parameter ## allows removal of PSTN access prefix from the outgoing number. ## 0 - PSTN access prefix is retained in the outgoing number ## 1 - PSTN access prefix is stripped from the outgoing number. ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones, H1xx SIP R1.0 and later ## and J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later and J139 SIP R3.0.0.0 and later when the phone is failed over. ## This parameter is not supported in IP Office and 3PCC environments as there is no support for failover. ## SET ENABLE_REMOVE_PSTN_ACCESS_PREFIX 0 ## ## Local Dial Area Code ## LOCAL_DIAL_AREA_CODE indicates whether user must dial area code for calls within same ## area code regions. when LOCAL_DIAL_AREA_CODE is enabled (1), the area code parameter (PHNLAC) ## should also be configured (ie. not the empty string). ## 0 - User don't need to dial area code. ## 1 - User need to dial area code. ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones ## when the phone is failed over. ## Note: This parameter is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET LOCAL_DIAL_AREA_CODE 0 ## ## Phone's Local Area Code ## When PHNLAC is set,it indicates the telephone's local area code, which along with ## the parameter LOCAL_DIAL_AREA_CODE, allows users to dial local numbers with more flexibility. ## PHNLAC is a string representing the local area code the telephone. ## Note: This parameter is supported on R2.4.1 and later release of 96xx SIP telephones ## when the phone is failed over. ## Note: This parameter is supported by J129 SIP R1.0.0.0 (or R1.1.0.0), J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## SET PHNLAC "" ## ##################### SIP USER CREDENTIALS SETTINGS ##################### ## ## SIP User Credentials settings ## Configure Username, Password and User ID to be used ## for SIP Registration. Usernames are often identical ## to User ID. ## FORCE_SIP_USERNAME replaces user field entered by user during Login ## FORCE_SIP_PASSWORD replaces password entered by user during Login ## FORCE_SIP_EXTENSION replaces User ID entered by user during Login ## If these are set, the user will not be prompted to Login on power cycle. ## Note: This parameter is supported by: ## J129 SIP R1.1.0.0, J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later ## 96x1 SIP R7.1.1.0 and later ## SET FORCE_SIP_USERNAME "7415" ## SET FORCE_SIP_PASSWORD "2222" ## SET FORCE_SIP_EXTENSION "741515" ## ## GET $MACADDR will request for the "MACADDR" file from the HTTP/HTTPS Server where "$MACADDR" which will be replaced by the telephone's MAC address. ## Note: This parameter is supported by J129 SIP R1.1.0.0, J169/J179 R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later and 96x1 SIP R7.1.1.0 and later ## GET $MACADDR.txt ## ############################################################ ## # PRODUCT_LINE_SETTINGS ## ############################################################ IF $MODEL4 SEQ 1603 GOTO SETTINGS16XX IF $MODEL4 SEQ 1608 GOTO SETTINGS16XX IF $MODEL4 SEQ 1616 GOTO SETTINGS16XX IF $MODEL4 SEQ 9610 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9620 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9630 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9640 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9650 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9670 GOTO SETTINGS96X0 IF $MODEL4 SEQ 9608 GOTO SETTINGS96X1 IF $MODEL4 SEQ 9611 GOTO SETTINGS96X1 IF $MODEL4 SEQ 9621 GOTO SETTINGS96X1 IF $MODEL4 SEQ 9641 GOTO SETTINGS96X1 IF $MODEL4 SEQ J129 GOTO SETTINGSJ100 IF $MODEL4 SEQ J139 GOTO SETTINGSJ100 IF $MODEL4 SEQ J169 GOTO SETTINGSJ100 IF $MODEL4 SEQ J179 GOTO SETTINGSJ100 ## Note that the 9601 is not grouped with the 96x1 telephones GOTO PER_MODEL_SETTINGS ############################################################## ## # SETTINGSJ100 ## ########## Add settings for J100 telephones below ########## IF $SIG_IN_USE SEQ H323 GOTO SETTINGS96X1H323 ######## Add settings for J100 SIP telephones below ######## ## ## LANGLARGEFONT specifies the name of the language file for the display of large text. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in the Text Size option not being offered. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J139 SIP R3.0.0.0 and later ## SET LANGLARGEFONT Mlf_Englarge.xml ## ###### End of J100 SIP product line-specific settings ###### GOTO PER_MODEL_SETTINGS ############################################################## ## # SETTINGS96X1 ## ########## Add settings for 96x1 telephones below ########## IF $SIG_IN_USE SEQ H323 GOTO SETTINGS96X1H323 ######## Add settings for 96x1 SIP telephones below ######## ## ## LANGLARGEFONT specifies the name of the language file for the display of large text. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in the Text Size option not being offered. ## This parameter is supported by: ## 96x1 SIP R6.2 and later ## SET LANGLARGEFONT Mlf_Englarge.xml ## ###### End of 96x1 SIP product line-specific settings ###### GOTO PER_MODEL_SETTINGS # SETTINGS96X1H323 ####### Add settings for 96x1 H.323 telephones below ####### ## ## Note: Starting R6.6 release language file name convention was changed from "mlf_s96x1_..." to "mlf_96x1_..." ## In addition, the template English filename was changed from "..._template_english.txt to "..._template_en.txt". ## ## LANGSYS specifies the name of a language file to use for the default language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in built-in English language strings being used. ## Note: User can also change the language using the field "Language..." in HOME-> Options & Settings-> Screen & Sound Options menu. ## LANGSYS will be enforced only on login screen or in case user did not change at all the field "Language..." value. ## Please note that user changes are stored in backup/restore file as LANGUSER (if BRURI has a valid value) which means that if the ## restored file include LANGUSER parameter then it will take precedence over LANGSYS. If BRURI is not valid, but user still change ## the content of "Language..." field, then user value will take precedence over LANGSYS (The only way to clear user configuration ## in this case is by doing: ## a. "CLEAR" operation in CRAFT menu, ## b. Logout (In this case the user configuration is not erased, but LANGSYS will be enforced). ## c. New user login. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET LANGSYS "" ## ## LANG1FILE specifies the name of the language file for the first user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET LANG1FILE mlf_96x1_v131_german.txt ## ## LANG2FILE specifies the name of the language file for the second user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET LANG2FILE mlf_96x1_v131_russian.txt ## ## LANG3FILE specifies the name of the language file for the third user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET LANG3FILE mlf_96x1_v131_spanish_latin.txt ## ## LANG4FILE specifies the name of the language file for the fourth user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.0 and later ## SET LANG4FILE mlf_96x1_v131_korean.txt ## ## LANGLARGEFONT specifies the name of the language file for the display of large text. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in the Text Size option not being offered. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## 96x1 H.323 R6.1 and later ## SET LANGLARGEFONT mlf_96x1_v131_english_large.txt ## ## VOXFILES specifies a list of voice language files that determine the ## list of Voice Dialing Languages that is presented to the user. ## The list can contain up to 255 characters; the default value is null (""). ## File names are separated by commas without any intervening spaces. ## The first file in the list will be downloaded by default. ## The first three characters of the filename indicate the language supported as follows: ## DUN Dutch ## ENG U.K. English ## ENU U.S. English ## FRF Parisian French ## GED German ## ITI Italian ## PTB Brazilian Portuguese ## SPE European Spanish ## This parameter is supported by: ## 96x1 H.323 R6.2 and subsequent dot releases, but not by R6.3 and later ## SET VOXFILES ENU_S20_FL_v1.tar,SPE_S20_FL_v1.tar,GED_S20_FL_v1.tar ## ##### End of 96x1 H.323 product line-specific settings ##### ## ################# END OF 96X1 SETTINGS ####################### GOTO PER_MODEL_SETTINGS ############################################################## ## # SETTINGS96X0 ## ########## Add settings for 96x0 telephones below ########## ## ## LANGSYS specifies the name of a language file to use for the default language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in built-in English language strings being used. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANGSYS "" ## ## LANG1FILE specifies the name of the language file for the first user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANG1FILE mlf_S31_v49_german.txt ## ## LANG2FILE specifies the name of the language file for the second user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANG2FILE mlf_S31_v49_russian.txt ## ## LANG3FILE specifies the name of the language file for the third user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANG3FILE mlf_S31_v49_spanish_latin.txt ## ## LANG4FILE specifies the name of the language file for the fourth user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANG4FILE mlf_S31_v49_korean.txt ## ## LANGLARGEFONT specifies the name of the language file for the display of large text. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in the Text Size option not being offered. ## This parameter is supported by: ## 96x0 H.323 R1.1 and later ## SET LANGLARGEFONT mlf_S31_v49_english_large.txt ## ## VOXFILES specifies a list of voice language files that determine the ## list of Voice Dialing Languages that is presented to the user. ## The list can contain up to 255 characters; the default value is null (""). ## File names are separated by commas without any intervening spaces. ## The first file in the list will be downloaded by default. ## The first three characters of the filename indicate the language supported as follows: ## DUN Dutch ## ENG U.K. English ## ENU U.S. English ## FRF Parisian French ## GED German ## ITI Italian ## PTB Brazilian Portuguese ## SPE European Spanish ## This parameter is supported by: ## 96x0 H.323 R2.0 and later ## SET VOXFILES ENU_S20_v3.tar,SPE_S20_v3.tar,GED_S20_v3.tar ## ######## End of 96x0 product line-specific settings ######## GOTO PER_MODEL_SETTINGS # SETTINGS16XX ## ########## Add settings for 16xx telephones below ########## ## ## The following 10 predefined language files are supported by all 16xx H.323 software releases ## and don't require a font file. ## mlf_Sage_v54_dutch.txt ## mlf_Sage_v54_french_can.txt ## mlf_Sage_v54_french_paris.txt ## mlf_Sage_v54_german.txt ## mlf_Sage_v54_italian.txt ## mlf_Sage_v54_japanese_kat.txt ## mlf_Sage_v54_portuguese.txt ## mlf_Sage_v54_russian.txt ## mlf_Sage_v54_spanish.txt ## mlf_Sage_v54_spanish_latin.txt ## ## The following 5 predefined language files supported by 16xx H.323 R1.1 and later software releases ## also require a font file, but only one font file may be downloaded, ## so at most one of these language files should be specified. ## mlf_Sage_v54_arabic.txt ## mlf_Sage_v54_chinese.txt ## mlf_Sage_v54_trad_chinese.txt ## mlf_Sage_v54_hebrew.txt ## mlf_Sage_v54_korean.txt ## ## The font files for the 5 languages above are as follows, in the same order: ## Arabic_S11_V34.rbm.lzma ## GB_S11_V34.rbm.lzma ## Big5_S11_V34.rbm.lzma ## Hebrew_S11_V34.rbm.lzma ## KSC_S11_V34.rbm.lzma ## ## LANGSYS specifies the name of a language file to use for the default language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results in built-in English language strings being used. ## This parameter is supported by: ## 16xx H.323 R1.0 and later ## SET LANGSYS "" ## ## LANG1FILE specifies the name of the language file for the first user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 16xx H.323 R1.0 and later ## SET LANG1FILE mlf_Sage_v54_german.txt ## ## LANG2FILE specifies the name of the language file for the second user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 16xx H.323 R1.0 and later ## SET LANG2FILE mlf_Sage_v54_russian.txt ## ## LANG3FILE specifies the name of the language file for the third user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 16xx H.323 R1.0 and later ## SET LANG3FILE mlf_Sage_v54_spanish_latin.txt ## ## LANG4FILE specifies the name of the language file for the fourth user-selectable language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no user-selectable language for this parameter. ## This parameter is supported by: ## 16xx H.323 R1.0 and later ## SET LANG4FILE mlf_Sage_v54_korean.txt ## ## FONTFILE specifies the name of a font file to use for a non-Latin based language. ## The file name can contain 0-32 ASCII characters. ## The default value is null, which results no font file being downloaded. ## This parameter is supported by: ## 16xx H.323 R1.1 and later ## SET FONTFILE KSC_S11_V34.rbm.lzma ## ######## End of 16xx product line-specific settings ######## GOTO PER_MODEL_SETTINGS ############################################################## ## # PER_MODEL_SETTINGS ## ############################################################## IF $MODEL4 SEQ 1603 GOTO SETTINGS1603 IF $MODEL4 SEQ 1608 GOTO SETTINGS1608 IF $MODEL4 SEQ 1616 GOTO SETTINGS1616 IF $MODEL4 SEQ 9610 GOTO SETTINGS9610 IF $MODEL4 SEQ 9620 GOTO SETTINGS9620 IF $MODEL4 SEQ 9630 GOTO SETTINGS9630 IF $MODEL4 SEQ 9640 GOTO SETTINGS9640 IF $MODEL4 SEQ 9650 GOTO SETTINGS9650 IF $MODEL4 SEQ 9670 GOTO SETTINGS9670 IF $MODEL4 SEQ 9601 GOTO SETTINGS9601 IF $MODEL4 SEQ 9608 GOTO SETTINGS9608 IF $MODEL4 SEQ 9611 GOTO SETTINGS9611 IF $MODEL4 SEQ 9621 GOTO SETTINGS9621 IF $MODEL4 SEQ 9641 GOTO SETTINGS9641 IF $MODEL4 SEQ H175 GOTO SETTINGSH175 IF $MODEL4 SEQ J129 GOTO SETTINGSJ129 IF $MODEL4 SEQ J139 GOTO SETTINGSJ139 IF $MODEL4 SEQ J169 GOTO SETTINGSJ169 IF $MODEL4 SEQ J179 GOTO SETTINGSJ179 IF $MODEL4 SEQ K155 GOTO SETTINGSK155 IF $MODEL4 SEQ K165 GOTO SETTINGSK165 IF $MODEL4 SEQ K175 GOTO SETTINGSK175 GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS1603 ## ########## Add settings for 1603 telephones below ########## ## ## ############# End of 1603 model-specific settings ############ GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS1608 ## ########## Add settings for 1608 telephones below ########## ## ## ############# End of 1608 model-specific settings ############ GOTO GROUP_SETTINGS ############################################################ ## # SETTINGS1616 ## ########## Add settings for 1616 telephones below ########## ## ## ############# End of 1616 model-specific settings ############ GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9610 ## ########## Add settings for 9610 telephones below ########## ## ## Handset Sidetone ## Controls the level of sidetone in the handset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## ## SET AUDIOSTHS 0 ## ############## 9610 WML BROWSER SETTINGS ################ ## ## WMLSMALL specifies the URL of a WML page that will be rendered by the 9610 WML browser ## after the number of minutes of idle time specified by WMLIDLETIME ## if the value of IDLEAPP (specified in the 9610 backup/restore file) is null. ## If the values of WMLSMALL and IDLEAPP are both null, the Avaya one-X(TM) screen saver ## will be displayed after the number of minutes of idle time specified by WMLIDLETIME. ## ## SET WMLSMALL http://www.mycompany.com/my_screen.wml ## ############## End of 9610 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9620 ## ########## Add settings for 9620 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ############### 9620 PROGRESS TONE LEVELS ################ ## ## These parameters are supported by 96x0 SIP R2.6.5 and later releases. ## ## Headset Progress tone adjust ## Controls the level of progress tones in the headset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HEADSET_LEVEL 0 ## ## Handset Progress tone adjust ## Controls the level of progress tones in the handset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HANDSET_LEVEL 0 ## ## Speaker Progress tone adjust ## Controls the level of progress tones in the speaker. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_SPEAKER_LEVEL 0 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9620 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9630 ## ########## Add settings for 9630 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ############### 9630 PROGRESS TONE LEVELS ################ ## ## These parameters are supported by 96x0 SIP R2.6.5 and later releases. ## ## Headset Progress tone adjust ## Controls the level of progress tones in the headset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HEADSET_LEVEL 0 ## ## Handset Progress tone adjust ## Controls the level of progress tones in the handset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HANDSET_LEVEL 0 ## ## Speaker Progress tone adjust ## Controls the level of progress tones in the speaker. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_SPEAKER_LEVEL 0 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9630 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9640 ## ########## Add settings for 9640 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ############### 9640 PROGRESS TONE LEVELS ################ ## ## These parameters are supported by 96x0 SIP R2.6.5 and later releases. ## ## Headset Progress tone adjust ## Controls the level of progress tones in the headset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HEADSET_LEVEL 0 ## ## Handset Progress tone adjust ## Controls the level of progress tones in the handset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HANDSET_LEVEL 0 ## ## Speaker Progress tone adjust ## Controls the level of progress tones in the speaker. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_SPEAKER_LEVEL 0 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9640 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9650 ## ########## Add settings for 9650 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ############### 9650 PROGRESS TONE LEVELS ################ ## ## These parameters are supported by 96x0 SIP R2.6.5 and later releases. ## ## Headset Progress tone adjust ## Controls the level of progress tones in the headset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HEADSET_LEVEL 0 ## ## Handset Progress tone adjust ## Controls the level of progress tones in the handset. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## 6 nominal value 18 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_HANDSET_LEVEL 0 ## ## Speaker Progress tone adjust ## Controls the level of progress tones in the speaker. ## ## setting level ## 0 NORMAL level for most users (default) ## 1 nominal value 3 dB louder than default value ## 2 nominal value 6 dB louder than default value ## 3 nominal value 9 dB louder than default value ## 4 nominal value 12 dB louder than default value ## 5 nominal value 15 dB louder than default value ## ## SET NETWORK_PROGRESS_TONE_SPEAKER_LEVEL 0 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9650 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9670 ## ########## Add settings for 9670 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9670 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9601 ## ########## Add settings for 9601 telephones below ########## ## ## ############## End of 9601 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9608 ## ########## Add settings for 9608 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9608 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9611 ## ########## Add settings for 9611 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9611 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9621 ## ########## Add settings for 9621 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9621 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGS9641 ## ########## Add settings for 9641 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of 9641 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGSH175 ## ########## Add settings for H175 Video Collaboration Station below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. The sidetone level adjustment ## provided by the AUDIOSTHS parameter is applicable to both wired and DECT handsets. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## H1xx SIP R1.0.1 and later ## SET AUDIOSTHD 1 ## ############## End of H175 model-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # SETTINGSJ129 ## ########## Add settings for J129 SIP phones ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. The sidetone level adjustment ## provided by the AUDIOSTHS parameter is applicable to both wired and DECT handsets. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J129 SIP R1.0.0.0 and later ## SET AUDIOSTHS 1 ## ############## End of J129 SIP model-specific settings ########### GOTO GROUP_SETTINGS # SETTINGSJ139 ## ########## Add settings for J139 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169 SIP R1.5.0; J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169 SIP R1.5.0; J100 SIP R2.0.0.0 and later; J139 SIP R3.0.0.0 and later ## J169/J179 H.323 R6.7 and later ## SET AUDIOSTHD 1 ## ############## End of J139 model-specific settings ########### GOTO GROUP_SETTINGS # SETTINGSJ169 ## ########## Add settings for J169 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169 SIP R1.5.0; J100 SIP R2.0.0.0 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J169 SIP R1.5.0; J100 SIP R2.0.0.0 and later ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## Note: Supported by J169/J179 SIP R1.5.0 and J169/J179 H.323 R6.7 and later ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## Note: Supported by J169/J179 SIP R1.5.0 and by J169/J179 H.323 R6.7 and later ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of J169 model-specific settings ########### GOTO GROUP_SETTINGS # SETTINGSJ179 ## ########## Add settings for J179 telephones below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later ## J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later ## SET AUDIOSTHD 1 ## ## WMLHOME specifies the URL of a WML page to be displayed by default in the WML browser, ## and whenever the Home softkey is selected in the browser. ## The value can contain zero or one URL of up to 255 characters; the default value is null (""). ## If the value is null, the WML browser will be disabled. ## Note: Supported by J169/J179 SIP R1.5.0 and J169/J179 H.323 R6.7 and later ## SET WMLHOME http://www.myco.com/ipphoneapps/home.wml ## ## WMLIDLEURI specifies zero or one URL for a WML page to be displayed when the telephone ## has been idle for the number of minutes specified by the value of WMLIDLETIME. ## The value can contain up to 255 characters; the default value is null (""). ## Note: Supported by J169/J179 SIP R1.5.0 and J169/J179 H.323 R6.7 and later ## SET WMLIDLEURI http://www.myco.com/ipphoneapps/idlepage.wml ## ############## End of J179 model-specific settings ########### GOTO GROUP_SETTINGS # SETTINGSK155 ## ########## Add settings for Avaya Vantage Entry with Camera below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. The sidetone level adjustment ## provided by the AUDIOSTHS parameter is applicable to wired handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.0.0 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.0.0 and later ## SET AUDIOSTHD 1 ## ############## End of Avaya Vantage Entry with Camera ########### GOTO GROUP_SETTINGS # SETTINGSK165 ## ########## Add settings for Avaya Vantage without Camera below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. The sidetone level adjustment ## provided by the AUDIOSTHS parameter is applicable to wired handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET AUDIOSTHD 1 ## ############## End of Avaya Vantage without Camera-specific settings ########### GOTO GROUP_SETTINGS # SETTINGSK175 ## ########## Add settings for Avaya Vantage with Camera below ########## ## ## AUDIOSTHS specifies the level of sidetone in the handset. The sidetone level adjustment ## provided by the AUDIOSTHS parameter is applicable to wired handset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET AUDIOSTHS 1 ## ## AUDIOSTHD specifies the level of sidetone in the headset. ## Value Operation ## 0 NORMAL level for most users (default) ## 1 three levels softer than NORMAL ## 2 OFF (inaudible) ## 3 one level softer than NORMAL ## 4 two levels softer than NORMAL ## 5 four levels softer than NORMAL ## 6 five levels softer than NORMAL ## 7 six levels softer than NORMAL ## 8 one level louder than NORMAL ## 9 two levels louder than NORMAL ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later ## SET AUDIOSTHD 1 ## ############## End of Avaya Vantage with Camera-specific settings ########### GOTO GROUP_SETTINGS ############################################################## ## # GROUP_SETTINGS ## ############################################################## ## ## Parameter values can be set for specifically-designated groups of ## telephones by using IF statements based on the GROUP parameter. ## ## The value of GROUP can be set manually in a telephone by using the ## GROUP local craft procedure or, for H.323 telephones, it can be set ## remotely by CM based on the telephone's extension number. ## The default value of GROUP in each telephone is 0, ## and the maximum value is 999. ## ## To create a group of settings, use one of the templates below, ## or create others just like them. ## ############################################################## IF $GROUP SEQ 1 GOTO GROUP_1 IF $GROUP SEQ 2 GOTO GROUP_2 IF $GROUP SEQ 3 GOTO GROUP_3 IF $GROUP SEQ 4 GOTO GROUP_4 IF $GROUP SEQ 5 GOTO GROUP_5 GOTO END ############################################################## # GROUP_1 ########## Add SET Statements for GROUP 1 below ############ ################ END OF GROUP 1 SETTINGS ##################### GOTO END ############################################################## # GROUP_2 ########## Add SET Statements for GROUP 2 below ############ ################ END OF GROUP 2 SETTINGS ##################### GOTO END ############################################################## # GROUP_3 ########## Add SET Statements for GROUP 3 below ############ ################ END OF GROUP 3 SETTINGS ##################### GOTO END ############################################################## # GROUP_4 ########## Add SET Statements for GROUP 4 below ############ ################ END OF GROUP 4 SETTINGS ##################### GOTO END ############################################################## # GROUP_5 ########## Add SET Statements for GROUP 5 below ############ ################ END OF GROUP 5 SETTINGS ##################### GOTO END ############################################################## # END ############## END OF CONFIGURATION FILE ##################### ###################################################################################### ## ## HISTORY TABLE ## ## 04-May-2017: ## 1. List 96x1 SIP R7.1.0.0 changes: ## 1. Mark as supported the following parameters: ADMIN_PASSWORD, OCSP_CACHE_EXPIRY, FIPS_ENABLED, IPV6STAT, FQDN_IP_MAP, ENCRYPT_SRTCP, DELETE_MY_CERT, PKCS12URL, ## PKCS12_PASSWD_RETRY, SERVER_CERT_RECHECK_HOURS, CERT_WARNING_DAYS, USBPOWER, OCSP_HASH_ALGORITHM, OCSP_USE_CACHE, OCSP_CACHE_EXPIRY, ## OCSP_ACCEPT_UNK, OCSP_NONCE, OCSP_URI, OCSP_URI_PREF, OCSP_TRUSTCERTS, ADMIN_PASSWORD, ADMIN_LOGIN_ATTEMPT_ALLOWED, ADMIN_LOGIN_LOCKED_TIME, DISPLAY_SSL_VERSION, ## DHCPSTAT, SNTP_SYNC_INTERVAL and VLANSEPMODE. ## 2. Add EXCHANGE_AUTH_USERNAME_FORMAT to provide support for Office 365. ## 3. Add MAX_TRUSTCERTS that specifies the maximum number of trusted certificate files that can be downloaded to the phone. ## 4. Add EASG_SITE_CERTS that specifies the list of EASG site certificates. ## 5. Add EASG_SITE_AUTH_FACTOR that specifies Site Authentication Factor code associated with the EASG site certificate being installed. ## 6. Add CERT_WARNING_DAYS_EASG that specifies how many days before the expiration of EASG product certificate a warning should first appear on the phone screen and ## Syslog message will be generated as well. ## 7. Add MEDIA_ADDR_MODE that specifies the IPv4/IPv6 SDP preference. ## 8. Add AUTO_UNMUTE that specifies whether the call will be unmuted on a transducer changing. ## 9. Add FORBIDDEN_SESSION_REMOVAL_TIMER which specifies the duration of an off-hook session before call is automatically ended in case no more call appearances ## is available on the called/remote party. ## 10. Add IPV6DADXMITS which specifies whether Duplicate Address Detection is performed on tentative addresses, as specified in RFC 4862. ## 11. Add SIGNALING_ADDR_MODE which specifies whether to use IPv4 or IPv6 for SIP registration. ## 12. Add MEDIA_NEG_PREFERENCE which specifies whether the answerer honor its own media preference or remote/offerer's media precedence. ## 13. Add SIP_CONTROLLER_LIST_2 which specifies the list of IPv4 or IPv6 SIP controllers. ## 14. Add ENABLE_MLPP which specifies whether MLPP feature is enabled or not. ## 15. Add MLPP_NET_DOMAIN which specifies MLPP Network Domain. ## 16. Add MLPP_MAX_PREC_LEVEL which specifies maximum allowed precedence level for the user. ## 17. Add ENABLE_PRECEDENCE_SOFTKEY which specifies whether precedence soft key should be enabled on idle line appearances on Phone Screen. ## 18. Add FORBIDDEN_SESSION_REMOVAL_TIMER which specifies the duration of an off-hook session before call is automatically ended in case no more call appearances ## is available on the called/remote party. ## 19. Add QTP_BUTTON_COMPRESS which specifies the range of features which can be assigned to Quick Touch Panel on Phone Screen. ## 20. Add note that MYCERTURL supports http or https. ## 21. Add note to SSH_ALLOWED that value 2 is supported and it is the default. ## 22. Add note to CURRENT_LOGO that "none" represents no wallpaper/logo is presented. Only time/data is presented. ## 23. Add note to MYCERTKEYLEN that only "2048" is supported. ## 24. Add DSCPAUD_FO which specifies the DSCP value for Flash Override precedence/priority level voice call. ## 25. Add DSCPAUD_FL which specifies the DSCP value for Flash precedence/priority level voice call. ## 26. Add DSCPAUD_IM which specifies the DSCP value for Immediate precedence/priority level voice call. ## 27. Add DSCPAUD_PR which specifies the DSCP value for Priority precedence/priority level voice call. ## 28. Add DSCPMGMT which specifies the DSCP value for OA&M management packet. ## 29. Add ENABLE_BLIND_TRANSFER which indicates whether to enable blind transfer or not ## 30. Add note to CONNECTION_REUSE that it supports value 1 only. ## 2. Add note that describes special characters supported with ADMIN_PASSWORD. ## 3. Add note to ENABLE_PHONE_LOCK that on J129 the Lock option appears in the main menu. There is no Lock softkey or feature button. ## 4. Update the list of products that support HTTPSRVR. ## 5. Add information to EXCHANGE_USER_DOMAIN and EXCHANGE_EMAIL_DOMAIN for their use and the way they are configured. ## 6. Add note that SCEP shall not be used when FIPS_ENABLED is set to 1. ## 7. Add note to OCSP_CACHE_EXPIRY that it will be used when nextUpdate field in the OCSP response is not available. ## 8. Add note to the SSH server section about the new authentication algorithm used with the SSH server - Enhanced Access Security Gateway (EASG). ## 9. Add note that FAST_RESPONSE_TIMEOUT is used in non-Avaya environment. In Avaya environment, this parameter will be overwritten by PPM configuration (96x1 SIP R6.2 and later). ## 10. Add note that MYCERTURL supports http or https for J129 R1.1.0.0 and later. ## ## 15-May-2017: ## 1. List Avaya Vantage Basic Application SIP R1.0.0.1 changes: ## 1. Add LOG_VERBOSITY defines whether or not the verbose logging is enabled or disabled. ## 2. Mark ENABLE_OPUS and OPUS_PAYLOAD_TYPE as supported. ## 3. Mark ENHDIALSTAT as supported (values 0,1). ## 4. Mark PHNOL, PHNCC, PHNLD, PHNIC, PHNDPLENGTH and PHNLDLENGTH as supported (with default ""). ## 5. Add APPLY_DIALINGRULES_TO_PLUS_NUMBERS, AUTOAPPLY_ARS_TO_SHORTNUMBERS, DIALPLANLOCALCALLPREFIX, DIALPLANNATIONALPHONENUMLENGTHLIST, DIALPLANEXTENSIONLENGTHLIST, ## DIALPLANPBXPREFIX and DIALPLANAREACODE. ## 2. Add note to MSGNUM that PSTN_VM_NUM shall be used with IP Office and 3PCC SIP environments instead of MSGNUM. ## ## 05-June-2017: ## 1. Update SIPPORT notes: SIPPORT is not supported by J129. Regarding 96x1 SIP, SIPPORT is supported up to R6.4.0 (excluded), from R6.4.0 and up to R7.1.0.0 (excluded) ## SIPPORT is only applied if CONNECTION_REUSE was set to 0 and from 7.1.0.0 and later SIPPORT is obsolete. ## 2. List Avaya Vantage SIP R1.0.0.0 (build 2304) changes: ## 1. BLUETOOTHSTAT is supported with default (1). ## 2. Add BLUETOOTH_FEATURES_SHARED_VIA_STAT which specifies whether "Shared via Bluetooth" option will be offered to the users or not. ## 3. Mark ADMIN_PASSWORD as supported with default "". ## 4. Add note to HEADSETBIDIR that only value 2 is supported. ## 5. Mark TIMEFORMAT as obsolete and add ADMINTIMEFORMAT which specifies the format of the time displayed in the phone (am/pm or 24h format). ## 6. Add PKCS12PASSWORD which specifies the PKCS12 file password. Mark PKCS12URL and PKCS12_PASSWD_RETRY as supported. ## 7. Mark EASG_SITE_CERTS and EASG_SITE_AUTH_FACTOR as supported. ## 3. List Avaya Equinox SIP R3.1 (running on Avaya Vantage Devices) changes: ## 1. Mark SIP_CONTROLLER_LIST, SIPDOMAIN and CONFERENCE_FACTORY_URI and supported. ## 2. Add CONFERENCE_ACCESS_NUMBER which specifies the default Conference Access Number. ## 3. Add CONFERENCE_PORTAL_URI which specifies the URI of the Conference Portal. ## 4. Add CONFERENCE_MODERATOR_CODE which specifies the conference moderator code. ## 5. Add CONFERENCE_PARTICIPANT_CODE which specifies the conference participant code. ## 6. Add CONFERENCE_VIRTUAL_ROOM which specifies the Scopia Virtual Room ID for the virtual room owner. ## 7. Add CONFERENCE_FQDN_SIP_DIAL_LIST which specifies a list of Scopia conferences bridges that can support SIP Enhanced Conference Experience. ## 8. Add UCCPENABLED which specifies whether to to enable or disable UCCP Conferencing protocol. ## 9. Add ESMENABLED which specifies whether Avaya Multimedia Messaging Service is enabled or not. ## 10. Add ESMHIDEONDISCONNECT which specifies whether to hide Avaya Multimedia Messaging conversations and message details in the Messages screen and ## Messaging area of the Top Of Mind screen when not connected to Avaya Multimedia Messaging. ## 11. Add ESMSRVR which specifies IP address or FQDN of Avaya Multimedia Messaging server. ## 12. Add ESMPORT which specifies the port number of Avaya Multimedia Messaging server. ## 13. Add ESMSECURE which specifies whether to use TLS or TCP. ## 14. Add ESMREFRESH which specifies Messaging refresh interval in minutes. ## 15. Mark ACSENABLED, ACSSRVR, ACSPORT and ACSSECURE as supported. ## 16. Add CONTACT_MATCHING_SEARCH_LOCATION which specifies whether to resolve the contact in local contact cache or search the AADS or both. ## 17. Mark ENHDIALSTAT (values 0 and 1), PHNCC, PHNDPLENGTH, PHNDPLENGTH, PHNIC, PHNLD, PHNLDLENGTH, PHNOL, APPLY_DIALINGRULES_TO_PLUS_NUMBERS, ## AUTOAPPLY_ARS_TO_SHORTNUMBERS, DIALPLANLOCALCALLPREFIX, DIALPLANNATIONALPHONENUMLENGTHLIST, DIALPLANEXTENSIONLENGTHLIST, DIALPLANPBXPREFIX ## and DIALPLANAREACODE as supported. ## 18. Add DND_SAC_LINK specifies whether to activate the SendAllCall when user enables DoNotDisturb. ## 19. Add EWSENABLED which specifies whether EXCHANGE WEB SERVICES (EWS) is enabled or not. ## 20. Add EWSSERVERADDRESS which specifies the Server Address that can be used to connect to EWS directly. ## 21. Add EWSDOMAIN which specifies the Exchange Server domain. ## 22. Mark DTMF_PAYLOAD_TYPE as supported. ## 23. Mark RTP_PORT_LOW and RTP_PORT_RANGE as supported. ## 24. Mark ENABLE_OPUS and OPUS_PAYLOAD_TYPE as supported. ## 25. Mark MEDIAENCRYPTION and ENCRYPT_SRTCP as supported. ## 26. Mark ENABLE_VIDEO and VIDEO_MAX_BANDWIDTH_ANY_NETWORK as supported. ## 27. Add SUPPORTEMAIL which defines the default E-mail address to send diagnostic logs. ## 28. Add SUPPORTURL which defines the default URL to get support. ## 29. Mark LOG_VERBOSITY as supported. ## 30. Add ANALYTICSENABLED which defines whether to allow data collection by Avaya using Google Analytics on behalf of the administrator's user community or not. ## 31. Mark TLSSRVRID as supported. ## 32. Add LOCKED_PREFERENCES which specifies list of parameters configured in the Avaya Equinox Application under "User preferences" menus ## which shall be blocked for user configuration. ## 33. Add OBSCURE_PREFERENCES which specifies list of parameters configured in the Avaya Equinox Application under "User preferences" menus ## which shall be hidden for users. I.e. users cannot see them. ## 34. Add AUTO_AWAY_TIME which specifies the idle time (in minutes) until presence automatically changes to 'away'. ## 35. Add ADDRESS_VALIDATION which specifies whether messaging address validation is enabled or not. ## 36. Add PHONE_NUMBER_PRIORITY which specifies the default phone number priority. ## 37. Mark NAME_SORT_ORDER and NAME_DISPLAY_ORDER as supported. ## 38. Add HOMESCREENLAYOUT which specifies home screen layout. ## ## 20-June-2017: ## 1. Update Avaya Equinox version to 3.1.2. ## 2. Remove SUPPORTURL. ## 3. Mark ENABLE_CONTACTS, ENABLE_CALL_LOG and ENABLE_VIDEO as supported by Avaya Vantage Basic application R1.0.0.1 and later. ## 4. Add ENABLE_FAVORITES which specifies whether favorites tab is displayed. Mark it as supported by Avaya Vantage Basic application R1.0.0.1 and later. ## 5. Mention that SYMMETRIC_RTP is supported by 96x1 SIP phones with hardware version below 3. Remove J129 support. ## ## 04-July-2017: ## 1. Add information about Avaya Vantage Open application. ## a. Avaya Vantage Open application retrieves all its configuration from the Avaya MPS server. ## All Avaya Vantage configuration parameters are applicable when Avaya Vantage Open application is used unless explicitly stated otherwise below: ## USER_AUTH_FILE_SERVER_URL, ALLOW_LOGOUT_WHEN_LOCKED, ACTIVE_CSDK_BASED_PHONE_APP, PIN_APP, SIPDOMAIN, SIP_CONTROLLER_LIST, WIFI_CON_STATUS_ON_LOGOUT, ## ENABLE_PHONE_LOCK, PHONE_LOCK_IDLETIME, PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS, ADMIN_INITIAL_SCREEN. ## b. All the below parameters are NOT used by Avaya Vantage Open application. However, the Avaya Vantage device can use them while Avaya Vantage Open application ## is used: ## TRUSTCERTS, TLSSRVRID, TLS_VERSION, MYCERTURL, MYCERTCN, MYCERTDN, MYCERTCAID, MYCERTREPLACE, SCEPPASSWORD, PKCS12URL, PKCS12_PASSWD_RETRY, PKCS12PASSWORD ## 2. Remove no-longer supported products. The support site shall include a copy of latest 46xxsettings.txt file that still include parameters for the below phones. ## 46xx H.323 R2.9.2 ## 46xx SIP R2.2.2 ## 364x SIP R1.1 ## 3631 H.323 R1.3.0 ## 16xx H.323 R1.3.3 ## 16CC SIP R1.0 ## 1603 SIP R1.0 ## 1692 H.323 R1.4 ## Softphone SIP R2.1 ## 3. Add comments to L2Q and VLANTEST parameters that for 96x1 SIP R7.1.0.0 and later, if L2QVLAN == 0, L2Q is treated as 2 (disabled). ## ## 10-Aug-2017: ## 1. 96x1 SIP R7.1.1.0 changes: ## a. Add note to SIG_PORT_LOW that the default and minimum values are 5062. ## b. Add note to SIG_PORT_RANGE that the maximum value is 60473. ## c. SYMMETRIC_RTP is supported by 9608 and SIP9611 HW version 3 and higher. ## 2. 96x1/B189 H.323 R6.6.5 changes: ## a. Support value 0 for CTATSTAT which means to not use smart enbloc even if smart enbloc is enabled or supported by Avaya Communication Manager by ## History, Redial, WML browser and Contacts applications. ## b. Mark SCREENSAVER as supported by B189. ## c. B189 H.323 R6.6.5 supports all VPN parameters specified in "VPN SETTINGS (H.323 ONLY)" section with exception of VPNCODE, VPNSTAT and QTESTRESPONDER. ## This includes (but not limited to): NVVPNMODE, NVSGIP, VPNALLOWTAGS, DHCPSRVR, NVVPNCFGPROF, NVIKEXCHGMODE, NVIKECONFIGMODE, NVVPNAUTHTYPE, NVVPNUSER, etc. ## 3. Mark HOMEIDLETIME as supported by J129 R1.0.0.0 and later. ## 4. PHONE_LOCK_IDLETIME, PHONE_LOCK_IDLETIME and PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS are supported when Avaya Vantage Open is used. Remove notes saying otherwise. ## 5. Add note to TRUSTCERTS that when Avaya Vantage Open is used, then this parameter is used to download trusted certificates to be used Avaya Vantage device ## (for example, 802.1x EAP-TLS) or by other applications (for example, Android Browser, etc.). ## 6. Remove a note saying MWISRVR is supported by H175/96x1. ## ## 28-Aug-2017: ## 1. 96x1 SIP R7.1.1.0 changes: ## a. Mark FORCE_SIP_USERNAME, FORCE_SIP_PASSWORD and FORCE_SIP_EXTENSION as supported. ## b. Mark "GET $MACADDR" as supported. ## ## 19-Oct-2017: ## 1. General - add note that double quotes (" ASCII 34) shall only be used. Left double quotation mark (“ ASCI 8220) and right double quotation mark (” ASCII 8221) ## shall NOT be used. Remove all left and right double quotation marks from this template file. ## 2. Avaya Vantage devices do not support PHY1STAT configured from the 46xxsettings.txt file. Remove the note which implies so. ## 3. The default of OCSP_USE_CACHE is 1 and not 0 (96x1 SIP 7.1.0.0 and later and J129 SIP R1.0.0.0 and later). ## ## 04-Jan-2018: ## 1. Update the default of ASTCONFIRMATION to 32 for 96x1, J129 and H175. ## 2. Avaya Vantage Devices SIP R1.0.0.2 changes: ## a. Support LOGOS and CURRENT_LOGO. ## b. PIN_APP supports list of applications which can be pinned when using a special Avaya Android launcher application for kiosk mode. ## ## 20-Mar-2018: ## 1. Add note to VLANSEP parameter that for H175 devices, when VLANSEP ==0, H1xx sends untagged packets even if L2Q==0 or 1 and VLANTEST==0 or timer < VLANTEST. ## 2. Add note that the default for BRAUTH is 0. ## 3. Add example to BRURI configuration with username and password for HTTP Basic authentication for H.323 endpoints. ## 4. J169/J179 SIP R1.5.0: ## a. All 96x1 SIP R7.1.1.0 parameters are supported by J169/J179 SIP R1.5.0. In particular: L2Q, L2QVLAN, L2QAUD, L2QSIG, VLANSEP, VLANSEPMODE, PHY2VLAN, ## PHY2PRIO, PHY2TAGS, DSCPAUD, DSCPVID, DSCPSIG, WBCSTAT, QLEVEL_MIN, DHCPSTD, VLANTEST, REUSETIME, DNSSRVR, DOMAIN, SIPDOMAIN, SIP_CONTROLLER_LIST, ## SIP_CONTROLLER_LIST_2, SIMULTANEOUS_REGISTRATIONS, ENABLE_PPM_SOURCED_SIPPROXYSRVR, CONFIG_SERVER_SECURE_MODE, VOLUME_UPDATE_DELAY, ENABLE_AVAYA_ENVIRONMENT, ## DIALPLAN, PHNNUMOFSA, SNTPSRVR, SNTP_SYNC_INTERVAL, GMTOFFSET, DSTOFFSET, DSTSTART, DSTSTOP, WAIT_FOR_REGISTRATION_TIMER, REGISTERWAIT, WAIT_FOR_UNREGISTRATION_TIMER, ## WAIT_FOR_INVITE_RESPONSE_TIMEOUT, OUTBOUND_SUBSCRIPTION_REQUEST_DURATION, NO_DIGITS_TIMEOUT, INTER_DIGIT_TIMEOUT, FAILED_SESSION_REMOVAL_TIMER, TCP_KEEP_ALIVE_STATUS, ## TCP_KEEP_ALIVE_TIME, TCP_KEEP_ALIVE_INTERVAL, CONTROLLER_SEARCH_INTERVAL, ASTCONFIRMATION, FAST_RESPONSE_TIMEOUT, RDS_INITIAL_RETRY_TIME, RDS_MAX_RETRY_TIME, ## RDS_INITIAL_RETRY_ATTEMPTS, FORBIDDEN_SESSION_REMOVAL_TIMER, CONFERENCE_FACTORY_URI, EVENT_NOTIFY_AVAYA_MAX_USERS, ENABLE_PRESENCE, PRESENCE_SERVER, PRESENCE_ACL_CONFIRM, ## INSTANT_MSG_ENABLED, ENABLE_MLPP, MLPP_NET_DOMAIN, MLPP_MAX_PREC_LEVEL, ENABLE_PRECEDENCE_SOFTKEY, DSCPAUD_FO, DSCPAUD_FL, DSCPAUD_IM, DSCPAUD_PR, DSCPMGMT, EXCHANGE_SERVER_LIST, ## EXCHANGE_SERVER_SECURE_MODE, EXCHANGE_SERVER_MODE, PROVIDE_EXCHANGE_CONTACTS, USE_EXCHANGE_CALENDAR, EXCHANGE_USER_DOMAIN, EXCHANGE_AUTH_USERNAME_FORMAT, EXCHANGE_EMAIL_DOMAIN, ## ENABLE_EXCHANGE_REMINDER, EXCHANGE_REMINDER_TIME, EXCHANGE_SNOOZE_TIME, EXCHANGE_NOTIFY_SUBSCRIPTION_PERIOD, SPEAKERSTAT, MUTE_ON_REMOTE_OFF_HOOK, AUTO_UNMUTE, SDPCAPNEG, ENFORCE_SIPS_URI, ## 100REL_SUPPORT, DISPLAY_NAME_NUMBER, HOTLINE, PLAY_TONE_UNTIL_RTP, PLUS_ONE, TEAM_BUTTON_RING_TYPE, QTP_BUTTON_COMPRESS, LOCALLY_ENFORCE_PRIVACY_HEADER, BRANDING_VOLUME, ENABLE_OOD_MSG_TLS_ONLY, ## TEAM_BUTTON_REDIRECT_INDICATION, ENABLE_BLIND_TRANSFER, PROVIDE_KEY_REPEAT_DELAY, HEADSET_PROFILE_DEFAULT, HEADSET_PROFILE_NAMES, PHNEMERGNUM, ENABLE_SHOW_EMERG_SK, ## ENABLE_SHOW_EMERG_SK_UNREG, CLDISPCONTENT, LOG_DIALED_DIGITS, HEADSYS, SKILLSCREENTIME, UUIDISPLAYTIME, ENTRYNAME, BUTTON_MAPPINGS, CC_INFO_TIMER, RECORDINGTONE, RECORDINGTONE_INTERVAL, ## RECORDINGTONE_VOLUME, TRUSTCERTS, MAX_TRUSTCERTS, TLSSRVRID, FQDN_IP_MAP, SERVER_CERT_RECHECK_HOURS, CERT_WARNING_DAYS, DELETE_MY_CERT, TLS_VERSION, HTTPPROXY, HTTPEXCEPTIONDOMAINS, ## MYCERTURL, MYCERTCN, MYCERTDN, MYCERTCAID, MYCERTKEYLEN, MYCERTRENEW, MYCERTWAIT, SCEPPASSWORD, PKCS12URL, PKCS12_PASSWD_RETRY, DOT1XSTAT, DOT1X, DOT1XEAPS, FIPS_ENABLED, OCSP_ENABLED, ## OCSP_ACCEPT_UNK, OCSP_NONCE, OCSP_URI, OCSP_URI_PREF, OCSP_TRUSTCERTS, OCSP_HASH_ALGORITHM, OCSP_USE_CACHE, OCSP_CACHE_EXPIRY, TPSLIST, SUBSCRIBELIST, PUSHCAP, PUSHPORT, WMLHOME, WMLPROXY, ## WMLPORT, WMLEXCEPT, WMLHELPSTAT, BAKLIGHTOFF, SCREENSAVERON, WMLIDLETIME, WMLIDLEURI, ENABLE_PHONE_LOCK, PHONE_LOCK_IDLETIME, ENABLE_G711A, ENABLE_G711U, ENABLE_G722, ENABLE_G726, G726_PAYLOAD_TYPE, ## ENABLE_G729, SEND_DTMF_TYPE, DTMF_PAYLOAD_TYPE, SYMMETRIC_RTP, PHNMUTEALERT_BLOCK, MATCHTYPE, USE_QUAD_ZEROES_FOR_HOLD, SIG, PHY1STAT, PHY2STAT, PHY2_AUTOMDIX_ENABLED, PROCSTAT, PROCPSWD, ## ADMIN_LOGIN_ATTEMPT_ALLOWED, ADMIN_LOGIN_LOCKED_TIME, SNMPSTRING, SNMPADD, LLDP_ENABLED, LOGSRVR, LOCAL_LOG_LEVEL, LOG_CATEGORY, ENABLE_RECORDING, WARNING_FILE, SSH_ALLOWED, SSH_BANNER_FILE, ## SSH_IDLE_TIMEOUT, EASG_SITE_CERTS, EASG_SITE_AUTH_FACTOR, CERT_WARNING_DAYS_EASG, AUTHCTRLSTAT, ENHDIALSTAT, PHNCC, PHNDPLENGTH, PHNIC, PHNLD, PHNLDLENGTH, PHNOL, ELD_SYSNUM, AUDIOENV, ## RINGTONESTYLE, RINGTONES, RINGTONES_UPDATE, PROVIDE_CF_RINGTONE, HTTPSRVR, HTTPDIR, HTTPPORT, AUTH, TLSSRVR, TLSDIR, TLSPORT, RTCPMON, RTCPMONPORT, RTCPMONPERIOD, ICMPDU, ICMPRED, AMADMIN, ## DHCPSTAT, IPV6STAT, SIGNALING_ADDR_MODE, MEDIA_NEG_PREFERENCE, MEDIA_ADDR_MODE, IPV6DADXMITS, RTCP_XR, MEDIAENCRYPTION, ENCRYPT_SRTCP, ENABLE_IPOFFICE, SUBSCRIBE_LIST_NON_AVAYA, LOGOS, ## CURRENT_LOGO, DISPLAY_SSL_VERSION, SYSTEM_LANGUAGE, LANGUAGES, COUNTRY, SIG_PORT_LOW, SIG_PORT_RANGE, INGRESS_DTMF_VOL_LEVEL, EXTEND_RINGTONE, FORCE_SIP_USERNAME, FORCE_SIP_PASSWORD, ## FORCE_SIP_EXTENSION, LANGLARGEFONT, AUDIOSTHS, AUDIOSTHD and ADMIN_PASSWORD. ## b. The following parameters are not supported by J169/J179 SIP R1.5.0: SLMSTAT, SLMCAP, SLMCTRL, SLMPERF, SLMPORT, SLMSRVR, USBPOWER (as there is no USB port), HOMEIDLETIME, BLUETOOTHSTAT. ## c. The following parameters are supported by J169/J179 SIP R1.5.0 (and not supported yet by 96x1 SIP R7.1.1.0): HANDSET_PROFILE_DEFAULT and HANDSET_PROFILE_NAMES. ## 5. Mark BRURI as supported by 96x1 SIP R7.1.0.0 for uploading phone report to HTTP/S file server. ## 6. Add EEESTAT which controls whether Energy-Efficient Ethernet (802.3az) is enabled on PHY1 and PHY2. Mark it as supported by J129 SIP R1.0.0.0. ## 7. Add note to PSTN_VM_NUM and MSGNUM that PSTN_VM_NUM is also in case of failover from Aura environment to a non-Aura server. ## 8. Mask the example of MUTE_ON_REMOTE_OFF_HOOK. ## 9. Remove the parameter ENABLE_MULTIPLE_CONTACT_WARNING as it is supported by 16cc which is no longer described in the new 46xxsettings.txt file template. ## ## 30-Mar-2018: ## 1. Avaya Equinox 3.2 running on Avaya Vantage: ## a. Add relevant parameters for Avaya Equinox 3.2 running on Avaya Vantage - UNIFIEDPORTALENABLED, SHOW_EQUINOX_MEETING_PANEL_IN_TOM and ENABLE_MEDIA_HTTP_TUNNEL. ## 2. Add note to EWS, Unified Portal, AADS contacts services and Avaya Multimedia Messaging Services that these services are supported on Avaya Vantage devices with unified login only ## (EWSSSO, UNIFIED_PORTAL_SSO, ACSSSO, ESMSSO are enforced internally to 1). SIPSSO is enforced to 0 as there is no unified login with SIP yet. ## 3. Avaya Equinox 3.3 running on Avaya Vantage: ## a. Add relevant parameters for Avaya Equinox 3.3 running on Avaya Vantage - ENABLE_PPM_CALL_JOURNALING. ## 4. Avaya Vantage Basic Application SIP R1.1.0.0 running on Avaya Vantage: ## a. Add relevant parameters for Avaya Vantage Basic Application SIP R1.1.0.0 running on Avaya Vantage: ## 1. Extension or name display - EXTENSION_NAME_DISPLAY_OPTIONS ## 2. Call log journaling: ENABLE_PPM_CALL_JOURNALING. Add note that ENABLE_PPM is not supported by Avaya Vantage Basic application. ## 3. Support Google Analytics: ANALYTICSENABLED ## 5. Avaya Vantage Devices SIP R1.1.0.0: ## a. Add relevant parameters for Avaya Vantage Devices SIP R1.1.0.0: ## 1. Ringtones related parameters: RINGTONESTYLE and RINGTONES. Add note to ADMIN_CHOICE_RINGTONE parameter that the parameter can also be used to choose ringtone out of downloaded ## ringtones (using RINGTONES parameter supported in Avaya Vantage SIP R1.1.0.0 and later). ## 2. Sending audio and report to email address according to SUPPORTEMAIL. ## 3. Ethernet switch (for K165/K175 with embedded Ethernet switch) - Support new configuration parameters: PHY2STAT ("Auto-Neg" and "Disabled" values only), PHY2VLAN, PHY2TAGS, PHY2_AUTOMDIX_ENABLED, ## DOT1X and VLANSEP. ## 4. Add note to PHY1STAT that the parameter is permanently configured to "Auto-Negotiate" on Avaya Vantage. ## 5. Add support for Device Enrollment Service (DES): DES_STAT and ENABLE_PUBLIC_CA_CERTS. ## 6. Mark TLSSRVR as supported. ## 7. Add HW_ENCODER_STAT. ## b. Add note to SELINUX_MODE that changing SELINUX_MODE triggers resets on the Avaya Vantage devices. There is a confirmation message to the end user that reset is about to happen and users can do ## the reset immediately or later. ## 6. Avaya Vantage Basic Application SIP R1.1.0.1 running on Avaya Vantage: ## a. Add relevant parameters for Avaya Vantage Basic Application SIP R1.1.0.1 running on Avaya Vantage: ## 1. IP Office support: ENABLE_IPOFFICE, DSCPAUD, DSCPSIG, DSCPVID, PSTN_VM_NUM, SIMULTANEOUS_REGISTRATIONS, SUBSCRIBE_LIST_NON_AVAYA, REGISTERWAIT, POUND_KEY_AS_CALL_TRIGGER. ## 2. Add note that DSCPAUD, DSCPSIG and DSCPVID are used in IP office environment only (for Aura environment ## DSCPAUD is taken from PPM and configured using SMGR). ## b. Add note to ENABLE_OPUS description that it is supported in IPO environment as well. ## 7. Avaya Vantage Devices SIP R1.1.0.1: ## a. Add relevant parameters for Avaya Vantage Devices SIP R1.1.0.1: ## 1. IP Office Support: USER_STORE_URI, ENABLE_IPOFFICE, PHNMOREEMERGNUMS, PHNEMERGNUM. ## 2. Add note to UPGRADE_POLICY that for IPO environment, it shall be set 0. ## 8. Avaya Equinox 3.3.1 running on Avaya Vantage: ## a. Add relevant parameters for Avaya Equinox 3.3.1 running on Avaya Vantage - ENABLE_EQUINOX_MEETING_ACCOUNT_DISCOVERY. ## 9. Add note to USER_AUTH_FILE_SERVER_URL and ACSPORT that fresh installations of AADS 7.1.2+ will default to port 443. If an older AADS is upgraded to 7.1.2+, it will retain the old 8443 port. ## Add note to USER_AUTH_FILE_SERVER_URL that AADS supports only https. ## ## 24-Apr-2018: ## 1. J100 SIP R2.0.0.0 and later (J129, J169, J179): ## a. Mark the following parameters as supported: EXCHANGE_AUTH_USERNAME_FORMAT, ENABLE_PUBLIC_CA_CERTS, DES_STAT, SYMMETRIC_RTP, SIG_PORT_LOW, SIG_PORT_RANGE, SCREENSAVERON, ## GUESTLOGINSTAT, GUESTDURATION, GUESTWARNING, EXCHANGE_SERVER_LIST, PROVIDE_EXCHANGE_CONTACTS, PROVIDE_EXCHANGE_CALENDAR, EXCHANGE_USER_DOMAIN, ## EXCHANGE_EMAIL_DOMAIN, ENABLE_EXCHANGE_REMINDER, EXCHANGE_REMINDER_TIME, EXCHANGE_SNOOZE_TIME, EXCHANGE_REMINDER_TONE, EXCHANGE_NOTIFY_SUBSCRIPTION_PERIOD, ## L2Q, L2QVLAN, L2QAUD, L2QSIG, VLANSEP, VLANSEPMODE, PHY2VLAN, PHY2PRIO, PHY2TAGS, DSCPAUD, DSCPSIG, DHCPSTD, VLANTEST, REUSETIME, DNSSRVR, DOMAIN, SIPDOMAIN, ## SIP_CONTROLLER_LIST, ENABLE_UDP_TRANSPORT, SIPREGPROXYPOLICY, SIMULTANEOUS_REGISTRATIONS, ENABLE_PPM_SOURCED_SIPPROXYSRVR, CONFIG_SERVER_SECURE_MODE, ## VOLUME_UPDATE_DELAY, ENABLE_AVAYA_ENVIRONMENT, DIALPLAN, PHNNUMOFSA, SNTPSRVR, SNTP_SYNC_INTERVAL, GMTOFFSET, DSTOFFSET, DSTSTART,DSTSTOP, WAIT_FOR_REGISTRATION_TIMER, ## REGISTERWAIT, WAIT_FOR_UNREGISTRATION_TIMER, WAIT_FOR_INVITE_RESPONSE_TIMEOUT, OUTBOUND_SUBSCRIPTION_REQUEST_DURATION, NO_DIGITS_TIMEOUT, INTER_DIGIT_TIMEOUT, ## FAILED_SESSION_REMOVAL_TIMER, TCP_KEEP_ALIVE_STATUS, TCP_KEEP_ALIVE_TIME, TCP_KEEP_ALIVE_INTERVAL, CONTROLLER_SEARCH_INTERVAL, ASTCONFIRMATION, RDS_INITIAL_RETRY_TIME, ## RDS_MAX_RETRY_TIME, RDS_INITIAL_RETRY_ATTEMPTS, SIP_TIMER_T1, SIP_TIMER_T2, SIP_TIMER_T4, CONFERENCE_FACTORY_URI, EVENT_NOTIFY_AVAYA_MAX_USERS, ENABLE_PRESENCE, ## MUTE_ON_REMOTE_OFF_HOOK, SDPCAPNEG, ENFORCE_SIPS_URI, 100REL_SUPPORT, PLAY_TONE_UNTIL_RTP, LOCALLY_ENFORCE_PRIVACY_HEADER, ENABLE_SIP_USER_ID, ENABLE_STRICT_USER_VALIDATION, ## HANDSET_PROFILE_DEFAULT, HANDSET_PROFILE_NAMES, PHNEMERGNUM, PHNMOREEMERGNUMS, ENABLE_SHOW_EMERG_SK, ENABLE_SHOW_EMERG_SK_UNREG, TRUSTCERTS, TLSSRVRID, FQDN_IP_MAP, ## SERVER_CERT_RECHECK_HOURS, CERT_WARNING_DAYS, DELETE_MY_CERT, TLS_VERSION, HTTPPROXY, HTTPEXCEPTIONDOMAINS, MYCERTURL, MYCERTCN, MYCERTDN, MYCERTCAID, MYCERTKEYLEN, ## MYCERTRENEW, MYCERTWAIT, SCEPPASSWORD, PKCS12URL, PKCS12_PASSWD_RETRY, DOT1XSTAT, DOT1X, DOT1XEAPS, OCSP_ENABLED, OCSP_ACCEPT_UNK, OCSP_NONCE, OCSP_URI, OCSP_URI_PREF, ## OCSP_TRUSTCERTS, OCSP_HASH_ALGORITHM, OCSP_USE_CACHE, OCSP_CACHE_EXPIRY, ENABLE_PHONE_LOCK, PHONE_LOCK_IDLETIME, ENABLE_G711A, ENABLE_G711U, ENABLE_G722, ENABLE_G726, ## G726_PAYLOAD_TYPE, ENABLE_G729, ENABLE_OPUS, OPUS_PAYLOAD_TYPE, SEND_DTMF_TYPE, DTMF_PAYLOAD_TYPE, PHNMUTEALERT_BLOCK, MATCHTYPE, SIG, PHY1STAT, PHY2STAT, PHY2_AUTOMDIX_ENABLED, ## EEESTAT, PROCSTAT, PROCPSWD, ADMIN_PASSWORD, ADMIN_LOGIN_ATTEMPT_ALLOWED, ADMIN_LOGIN_LOCKED_TIME, SNMPSTRING, SNMPADD, LLDP_ENABLED, LOGSRVR, LOG_CATEGORY, SSH_ALLOWED, ## SSH_BANNER_FILE, SSH_IDLE_TIMEOUT, EASG_SITE_CERTS, EASG_SITE_AUTH_FACTOR, EASG_SITE_AUTH_FACTOR, CERT_WARNING_DAYS_EASG, AUTHCTRLSTAT, SLMSTAT, SLMCAP, SLMCTRL, SLMPERF, ## SLMPORT, SLMSRVR, ENHDIALSTAT, PHNCC, PHNDPLENGTH, PHNIC, PHNLD, PHNLDLENGTH, PHNOL, AGCHAND, AGCSPKR, HTTPSRVR, HTTPDIR, HTTPPORT, AUTH, TLSSRVR, TLSDIR, TLSPORT, ## RTCPMON, RTCPMONPORT, RTCPMONPERIOD, ICMPDU, ICMPRED, BRURI, CALLFWDSTAT, CALLFWDDELAY, CALLFWDADDR, ENABLE_AUTO_ANSWER_SUPPORT, AUTO_ANSWER_MUTE_ENABLE, ENABLE_DND, ## ENABLE_DND_PRIORITY_OVER_CFU_CFB, HOLD_REMINDER_TIMER, RTCP_XR, MEDIAENCRYPTION, ENCRYPT_SRTCP, ENABLE_IPOFFICE, SUBSCRIBE_LIST_NON_AVAYA, ENABLE_3PCC_ENVIRONMENT, ## USER_STORE_URI, PROVIDE_OPTIONS_SCREEN, PROVIDE_NETWORKINFO_SCREEN, PROVIDE_LOGOUT, DISPLAY_SSL_VERSION, ENABLE_CALL_LOG, ENABLE_REDIAL, ENABLE_CONTACTS, CONTACT_NAME_FORMAT, ## SYSTEM_LANGUAGE, LANGUAGES, COUNTRY, DAYLIGHT_SAVING_SETTING_MODE, RTP_PORT_LOW, RTP_PORT_RANGE, EXTEND_RINGTONE, DISCOVER_AVAYA_ENVIRONMENT, PSTN_VM_NUM, ## ENABLE_REMOVE_PSTN_ACCESS_PREFIX, LOCAL_DIAL_AREA_CODE, PHNLAC, FORCE_SIP_EXTENSION, SIP_CONTROLLER_LIST_2, ENABLE_MLPP, MLPP_NET_DOMAIN, MLPP_MAX_PREC_LEVEL, ## ENABLE_PRECEDENCE_SOFTKEY, DSCPAUD_FO, DSCPAUD_FL, DSCPAUD_IM, DSCPAUD_PR, DSCPMGMT, TEAM_BUTTON_RING_TYPE, TEAM_BUTTON_REDIRECT_INDICATION, ## HEADSET_PROFILE_DEFAULT, HEADSET_PROFILE_NAMES, HEADSYS, SKILLSCREENTIME, UUIDISPLAYTIME, DHCPSTAT, IPV6STAT, SIGNALING_ADDR_MODE, MEDIA_NEG_PREFERENCE, ## MEDIA_ADDR_MODE, IPV6DADXMITS, AGCHEAD, ENABLE_REDIAL_LIST, PROVIDE_KEY_REPEAT_DELAY, BAKLIGHTOFF, CONF_TRANS_ON_PRIMARY_APPR, AUTO_SELECT_ANY_IDLE_APPR, ## ENABLE_BLIND_TRANSFER, ENTRYNAME, CC_INFO_TIMER, MSGNUM, SPEAKERSTAT, WBCSTAT, QLEVEL_MIN, FAST_RESPONSE_TIMEOUT, FORBIDDEN_SESSION_REMOVAL_TIMER, ## PRESENCE_ACL_CONFIRM, ENABLE_OOD_MSG_TLS_ONLY, LOG_DIALED_DIGITS, RECORDINGTONE, RECORDINGTONE_INTERVAL, RECORDINGTONE_VOLUME, MAX_TRUSTCERTS, FIPS_ENABLED, ## LOCAL_LOG_LEVEL, ENABLE_RECORDING, WARNING_FILE, ELD_SYSNUM, AUDIOENV, RINGTONES, RINGTONES_UPDATE, AUDASYS, CONFERENCE_TYPE, INGRESS_DTMF_VOL_LEVEL ## b. Add new parameters: BACKGROUND_IMAGE, BACKGROUND_IMAGE_DISPLAY, BACKGROUND_IMAGE_SELECTABLE,SCREENSAVER_IMAGE, SCREENSAVER_IMAGE_DISPLAY, SCREENSAVER_IMAGE_SELECTABLE, ## MEDIA_PRESERVATION, PRESERVED_CALL_DURATION, WLAN_ESSID, ENABLE_NETWORK_CONFIG_BY_USER, WLAN_COUNTRY, WLAN_ENABLE_80211D, WLAN_SECURITY, WEP_DEFAULT_KEY, WEP_KEY_LEN, ## WLAN_PASSWORD, WLAN_WPA2E_EAP_METHOD, WLAN_WPA2E_IDENTITY, WLAN_WPA2E_ANONYMOUS_IDENTITY, WLAN_L2QUAD, WLAN_L2QSIG, WLAN_DSCPAUD, WLAN_DSCPSIG, ENABLE_WEBSERVER, ## WEBSERVER_ON_HTTP, WEB_HTTP_PORT, WEB_HTTPS_PORT, FORCE_WEB_ADMIN_PASSWORD, SOFTKEY_CONFIGURATION, BACKLIGHT_SELECTABLE, WEP_KEY1, WEP_KEY2, WEP_KEY3, WEP_KEY4, ## WLAN_WPA2E_EAP_PHASE2, SHOW_LAST_EXTENSION. ## c. Support new enumeration (2) for ENABLE_IPOFFICE. ## d. Support new enumeration (2) for WIFISTAT. ## e. Support new enumeration (0) for BRANDING_VOLUME. ## f. Add note to COUNTRY configuration parameters that WLAN_COUNTRY shall be used instead for J100 SIP R2.0.0.0. ## 2. Mark DSCPVID as supported by Avaya Vantage Basic Application SIP R1.1.0.1 and later; used in IP office environment only (for Aura environment ## DSCPVID is taken from PPM and configured using SMGR) and H1xx only. ## 3. Only values 0-1 for SSH_ALLOWED are supported 96x1 SIP R7.1.0.0. Default value is 0. ## 4. Add ALLOW_DND_SAC_LINK_CHANGE and DND_SAC_LINK and add note that they are supported by 96x1 SIP R6.4, J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only). ## 5. Add TRUST_AGENTS_AVAYA_SMARTLOCK_STAT and add note that it is supported by Avaya Vantage Devices SIP R1.1.0.0 and later. ## ## 28-May-2018: ## 1. J169/J179 H.323 R6.7: ## a. Mark ALL 96x1 H.323 configuration parameters as supported by J169/J179 H.323 phones (with exception of USB, Bluetooth related parameters which are NOT supported by ## J169/J179 H.323 phones). In particular, the following parameters as supported: L2Q, L2QVLAN, L2QAUD, L2QSIG, VLANSEP, VLANSEPMODE, PHY2VLAN, PHY2TAGS, DSCPAUD, ## DSCPSIG, WBCSTAT, QLEVEL_MIN, DHCPSTD, VLANTEST, REUSETIME, DNSSRVR, DOMAIN, QKLOGINSTAT, CLEAR_EXTPSWD_ON_LOGOUT, MCIPADD, VUMCIPADD, STATIC, UNNAMEDSTAT, ## REREGISTER, UDT, H323SIGPROTOCOL, GRATARP, GUESTLOGINSTAT, GUESTDURATION, GUESTWARNING, ADMIN_HSEQUAL, PHNEMERGNUM, APPSTAT, OPSTAT, OPSTAT2, SYSAUDIOPATH, ## PHNSCRALL, EOEDITDIAL, FBONCASCREEN, PHNSCRCOLUMNS, CADISPMODE, CALLAPPRSELMODE, CLDISPCONTENT, CLDELCALLBK, LOGMISSEDONCE, LOGUNSEEN, LOGBACKUP, CLBACKUPTIMESTAT, ## CLBACKUPTIME, CALL_LOG_JOURNAL, DEFAULTRING, TIMERSTAT, HEADSETBIDIR, AUTOANSSTAT, AUTOANSSTRING, AUTOANSALERT, HEADSYS, CALLCTRSTAT, OPSTATCC, AGTACTIVESK, ## AGTCALLINFOSTAT, AGTFWDBTNSTAT, AGTGREETINGSTAT, AGTGREETLOGOUTDEL, AGTVUSTATID, AGTLOGINFAC, AGTLOGOUTFAC, AGTSPKRSTAT, AGTTIMESTAT, AGTTRANSLTO, ## AGTTRANSLCLBK, AGTTRANSLPRI, AGTTRANSLPK, AGTTRANSLICOM, CCLOGOUTIDLESTAT, LOCALZIPTONEATT, AGENTGREETINGSDELAY, AGTCAINFOLINE, RECORDINGTONE, ## RECORDINGTONE_INTERVAL, RECORDINGTONE_VOLUME, CCBTNSTAT, CONFSTAT, DROPSTAT, HOLDSTAT, XFERSTAT, TRUSTCERTS, TLSSRVRVERIFYID, SERVER_CERT_RECHECK_HOURS, ## CERT_WARNING_DAYS, TLS_SECURE_RENEG, TLS_VERSION, MYCERTURL, MYCERTCN, MYCERTDN, MYCERTCAID, MYCERTKEYLEN, MYCERTRENEW, MYCERTWAIT, SCEPPASSWORD, ## MYCERTKEYUSAGE, PKCS12URL, DOT1XSTAT, DOT1X, DOT1XEAPS, DOT1XWAIT, FIPS_ENABLED, OCSP_ENABLED, OCSP_ACCEPT_UNK, OCSP_NONCE, OCSP_URI, OCSP_URI_PREF, ## OCSP_TRUSTCERTS, TPSLIST, SUBSCRIBELIST, PUSHCAP, PUSHPORT, WMLHOME, WMLPROXY, WMLPORT, WMLEXCEPT, WMLHELPSTAT, BAKLIGHTOFF, SCREENSAVERON, SCREENSAVER, ## WMLIDLETIME, WMLIDLEURI, PHY1STAT, PHY2STAT, PHY2_AUTOMDIX_ENABLED, PROCSTAT, PROCPSWD, MUTECRAFTOPTIONS, SNMPSTRING, SNMPADD, LLDP_XMIT_SECS, LOGSRVR, ## LOGLOCAL, LOGTOFILE, ENABLE_RECORDING, SSH_ALLOWED, SSH_BANNER_FILE, SSH_IDLE_TIMEOUT, AUTHCTRLSTAT, APPLICATIONWD, SLMSTAT, SLMCAP, SLMCTRL, SLMPERF, ## SLMPORT, SLMSRVR, ENHDIALSTAT, PHNCC, PHNDPLENGTH, PHNIC, PHNLD, PHNLDLENGTH, PHNOL, CTASTAT, AGCHAND, AGCHEAD, AGCSPKR, AUDIOENV, RINGTONESTYLE, ## HTTPSRVR, HTTPDIR, HTTPPORT, AUTH, TLSSRVR, TLSDIR, TLSPORT, RTCPMON, ICMPDU, ICMPRED, BRURI, BRAUTH, AUDASYS, LANG0STAT, AMADMIN, IDLEFEATURES, ## DIALFEATURES, RINGBKFEATURES, TALKFEATURES, TEAMBTNDISPLAY, RINGPRIORITY, LEDMODE, NVVPNMODE, NVSGIP, VPNALLOWTAGS, DHCPSRVR, NVVPNCFGPROF, ## NVIKEXCHGMODE, NVIKECONFIGMODE, NVVPNAUTHTYPE, NVVPNUSER, NVVPNPSWDTYPE, NVVPNCOPYTOS, NVVPNENCAPS, NVIKEPSK, NVIKEID, NVIKEIDTYPE, NVIPSECSUBNET, ## NVIKEDHGRP, NVPFSDHGRP, NVIKEP1ENCALG, NVIKEP2ENCALG, NVIKEP1AUTHALG, NVIKEP2AUTHALG, NORTELAUTH, NVXAUTH, VPNCODE, VPNPROC, ALWCLRNOTIFY, ## DROPCLEAR, NVMCIPADD, NVHTTPSRVR, NVTLSSRVR, NVIKEOVERTCP, NVIKEP1LIFESEC, NVIKEP2LIFESEC, NVVPNSVENDOR, NVVPNUSERTYPE, VPNTTS, NDREDV6, DHCPPREF, ## DHCPSTAT, IPPREF, IPV6STAT, PINGREPLYV6, GRATNAV6, LANGSYS, LANG1FILE, LANG2FILE, LANG3FILE, LANG4FILE, LANGLARGEFONT, AUDIOSTHS, AUDIOSTHD ## WMLHOME, WMLIDLEURI. ## b. MYCERTKEYLEN supports "2048" value only. ## c. Mark EASG_SITE_CERTS, EASG_SITE_AUTH_FACTOR, CERT_WARNING_DAYS_EASG as supported. ## 2. Avaya Vantage Basic application SIP R1.1.0.1: ## a. Mark CONFERENCE_FACTORY_URL parameter as supported for IPO environment only. ## 3. MYCERTWAIT is not supported by J100 phones. ## 4. Add note that MAX_TRUSTCERTS limits number of trusted certificates and NOT trusted certificate files. A a certificate file may include more than one trusted certificate. ## 5. J169/J179 SIP R1.5.0 supports values 0-1 for SSH_ALLOWED. ## ## 10-July-2018: ## 1. J100 SIP R3.0.0.0 and later (J129, J139, J169, J179): ## a. J139 is supported by J100 SIP R3.0.0.0. Mark the following as supported by J139 in J100 SIP R3.0.0.0: BAKLIGHTOFF, ENABLE_MLPP, MLPP_NET_DOMAIN, MLPP_MAX_PREC_LEVEL, ## ENABLE_PRECEDENCE_SOFTKEY, DSCPAUD_FO, DSCPAUD_FL, DSCPAUD_IM, DSCPAUD_PR, DSCPMGMT, PHY1STAT, PHY2STAT, PHY2_AUTOMDIX_ENABLED, HANDSET_PROFILE_DEFAULT, ## HANDSET_PROFILE_NAMES, ENABLE_IPOFFICE, MEDIA_PRESERVATION, PRESERVED_CALL_DURATION, ENABLE_SHOW_EMERG_SK, ENABLE_SHOW_EMERG_SK_UNREG, LANGUAGES, SYSTEM_LANGUAGE, ## SIP_CONTROLLER_LIST, SIMULTANEOUS_REGISTRATIONS, SIPREGPROXYPOLICY, TRUSTCERTS, SPEAKERSTAT, L2Q, L2QVLAN, L2QAUD, L2QSIG, VLANSEP, VLANSEPMODE, PHY2VLAN, PHY2PRIO, ## PHY2TAGS, DSCPAUD, DSCPSIG, WBCSTAT, QLEVEL_MIN, DHCPSTD, VLANTEST, REUSETIME, DNSSRVR, DOMAIN, ENABLE_UDP_TRANSPORT, SIP_CONTROLLER_LIST_2, SIPDOMAIN, ## ENABLE_PPM_SOURCED_SIPPROXYSRVR, ENABLE_AVAYA_ENVIRONMENT, DIALPLAN, PHNNUMOFSA, SNTPSRVR, SNTP_SYNC_INTERVAL, GMTOFFSET, DSTOFFSET, DSTSTART, DSTSTOP, ## WAIT_FOR_REGISTRATION_TIMER, REGISTERWAIT, WAIT_FOR_UNREGISTRATION_TIMER, WAIT_FOR_INVITE_RESPONSE_TIMEOUT, OUTBOUND_SUBSCRIPTION_REQUEST_DURATION, ## NO_DIGITS_TIMEOUT, INTER_DIGIT_TIMEOUT, FAILED_SESSION_REMOVAL_TIMER, TCP_KEEP_ALIVE_STATUS, TCP_KEEP_ALIVE_TIME, TCP_KEEP_ALIVE_INTERVAL, CONTROLLER_SEARCH_INTERVAL, ## ASTCONFIRMATION, FAST_RESPONSE_TIMEOUT, RDS_INITIAL_RETRY_TIME, RDS_MAX_RETRY_TIME, RDS_INITIAL_RETRY_ATTEMPTS, SIP_TIMER_T1, SIP_TIMER_T2, SIP_TIMER_T4, ## FORBIDDEN_SESSION_REMOVAL_TIMER, CONFERENCE_FACTORY_URI, EVENT_NOTIFY_AVAYA_MAX_USERS, ENABLE_PRESENCE, PRESENCE_ACL_CONFIRM, MUTE_ON_REMOTE_OFF_HOOK, ## SDPCAPNEG, ENFORCE_SIPS_URI, 100REL_SUPPORT, PLAY_TONE_UNTIL_RTP, LOCALLY_ENFORCE_PRIVACY_HEADER, ENABLE_SIP_USER_ID, ENABLE_STRICT_USER_VALIDATION, ## BRANDING_VOLUME, ENABLE_OOD_MSG_TLS_ONLY, PHNEMERGNUM, PHNMOREEMERGNUMS, LOG_DIALED_DIGITS, MAX_TRUSTCERTS, ENABLE_PUBLIC_CA_CERTS, TLSSRVRID, FQDN_IP_MAP, ## SERVER_CERT_RECHECK_HOURS, CERT_WARNING_DAYS, DELETE_MY_CERT, TLS_VERSION, HTTPPROXY, HTTPEXCEPTIONDOMAINS, MYCERTURL, MYCERTCN, MYCERTDN, MYCERTCAID, ## MYCERTRENEW, SCEPPASSWORD, PKCS12URL, PKCS12_PASSWD_RETRY, DOT1XSTAT, DOT1X, DOT1XEAPS, OCSP_ENABLED, OCSP_ACCEPT_UNK, OCSP_NONCE, OCSP_URI, OCSP_URI_PREF, ## OCSP_TRUSTCERTS, OCSP_HASH_ALGORITHM, OCSP_USE_CACHE, OCSP_CACHE_EXPIRY, ENABLE_WEBSERVER, WEBSERVER_ON_HTTP, WEB_HTTP_PORT, WEB_HTTPS_PORT, FORCE_WEB_ADMIN_PASSWORD, ## ENABLE_PHONE_LOCK, PHONE_LOCK_IDLETIME, ENABLE_G711A, ENABLE_G711U, ENABLE_G722, ENABLE_G726, G726_PAYLOAD_TYPE, ENABLE_G729, ENABLE_OPUS, OPUS_PAYLOAD_TYPE, ## SEND_DTMF_TYPE, DTMF_PAYLOAD_TYPE, SYMMETRIC_RTP, MATCHTYPE, PHNMUTEALERT_BLOCK, PROCSTAT, PROCPSWD, ADMIN_PASSWORD, ADMIN_LOGIN_ATTEMPT_ALLOWED, ADMIN_LOGIN_LOCKED_TIME, ## SNMPSTRING, SNMPADD, LLDP_ENABLED, LOGSRVR, LOCAL_LOG_LEVEL, LOG_CATEGORY, SSH_ALLOWED, SSH_BANNER_FILE, SSH_IDLE_TIMEOUT, EASG_SITE_CERTS, EASG_SITE_AUTH_FACTOR, ## CERT_WARNING_DAYS_EASG, AUTHCTRLSTAT, SLMSTAT, SLMCAP, SLMCTRL, SLMPERF, SLMPORT, SLMSRVR, ENHDIALSTAT, PHNCC, PHNDPLENGTH, PHNIC, PHNLD, PHNLDLENGTH, PHNOL, ## AGCHAND, AGCHEAD, AGCSPKR, AUDIOENV, HTTPSRVR, HTTPDIR, HTTPPORT, AUTH, TLSSRVR, TLSDIR, TLSPORT, DES_STAT, RTCPMON, RTCPMONPORT, RTCPMONPERIOD, ICMPDU, ICMPRED, ## BRURI, MSGNUM, IPV6STAT, SIGNALING_ADDR_MODE, MEDIA_NEG_PREFERENCE, MEDIA_ADDR_MODE, IPV6DADXMITS, CALLFWDDELAY, CALLFWDADDR, ENABLE_AUTO_ANSWER_SUPPORT, ## AUTO_ANSWER_MUTE_ENABLE, ENABLE_DND, ENABLE_DND_PRIORITY_OVER_CFU_CFB, HOLD_REMINDER_TIMER, CONFERENCE_TYPE, RTCP_XR, MEDIAENCRYPTION, ENCRYPT_SRTCP, ## SUBSCRIBE_LIST_NON_AVAYA, ENABLE_3PCC_ENVIRONMENT, USER_STORE_URI, PROVIDE_OPTIONS_SCREEN, PROVIDE_NETWORKINFO_SCREEN, PROVIDE_LOGOUT, DISPLAY_SSL_VERSION, ## ENABLE_CALL_LOG, ENABLE_REDIAL, CONTACT_NAME_FORMAT, DAYLIGHT_SAVING_SETTING_MODE, RTP_PORT_LOW, RTP_PORT_RANGE, SIG_PORT_LOW, SIG_PORT_RANGE, INGRESS_DTMF_VOL_LEVEL, ## DISCOVER_AVAYA_ENVIRONMENT, PSTN_VM_NUM, ENABLE_REMOVE_PSTN_ACCESS_PREFIX, LOCAL_DIAL_AREA_CODE, PHNLAC, FORCE_SIP_USERNAME, FORCE_SIP_PASSWORD, FORCE_SIP_EXTENSION, ## AUDIOSTHS, AUDIOSTHD, ENABLE_BLIND_TRANSFER, PROVIDE_KEY_REPEAT_DELAY, HEADSET_PROFILE_DEFAULT, HEADSET_PROFILE_NAMES, HEADSYS, ENTRYNAME, ENABLE_RECORDING, ## RECORDINGTONE_INTERVAL, RECORDINGTONE_VOLUME, SCREENSAVERON, BACKLIGHT_SELECTABLE, WARNING_FILE, ELD_SYSNUM, RINGTONESTYLE, PROVIDE_CF_RINGTONE, ## CONF_TRANS_ON_PRIMARY_APPR, AUTO_SELECT_ANY_IDLE_APPR, LANGLARGEFONT, AUDASYS ## b. As J139 does not support Wi-Fi/BT module the following parameters were marked as not supported: WIFISTAT, ENABLE_NETWORK_CONFIG_BY_USER, WLAN_COUNTRY, ## WLAN_ENABLE_80211D, WLAN_ESSID, WLAN_SECURITY, WEP_DEFAULT_KEY, WEP_KEY_LEN, WEP_KEY1, WEP_KEY2, WEP_KEY3, WEP_KEY4, WLAN_PASSWORD, WLAN_WPA2E_EAP_METHOD, ## WLAN_WPA2E_EAP_PHASE2, WLAN_WPA2E_IDENTITY, WLAN_WPA2E_ANONYMOUS_IDENTITY, WLAN_L2QUAD, WLAN_L2QSIG, WLAN_DSCPAUD, WLAN_DSCPSIG. ## c. Exchange Integration parameters are not supported: EXCHANGE_SERVER_LIST, PROVIDE_EXCHANGE_CONTACTS, ## PROVIDE_EXCHANGE_CALENDAR, EXCHANGE_USER_DOMAIN, EXCHANGE_AUTH_USERNAME_FORMAT, EXCHANGE_EMAIL_DOMAIN, ENABLE_EXCHANGE_REMINDER, EXCHANGE_REMINDER_TIME, ## EXCHANGE_SNOOZE_TIME, EXCHANGE_REMINDER_TONE, EXCHANGE_NOTIFY_SUBSCRIPTION_PERIOD ## d. Unsupported customization features: RINGTONES ## e. Mark the following parameters as supported by J100 SIP R3.0.0.0: TPSLIST, SUBSCRIBELIST, PUSHCAP and PUSHPORT. ## 2. Add recommendation note to use value 20 for BAKLIGHTOFF for ENERGY STAR® compliance on applicable phones. ## 3. Add note to CALLFWDSTAT about J129 support. ## 4. J100 SIP R2.0.0.0 and later: ## a. The default of SNTPSRVR to "0.avaya.pool.ntp.org,1.avaya.pool.ntp.org,2.avaya.pool.ntp.org,3.avaya.pool.ntp.org". ## b. Mark PROVIDE_CF_RINGTONE as supported by J100 SIP R2.0.0.0 (J169/J179). ## 5. The default of LOCAL_LOG_LEVEL for Avaya Vantage devices is 4 (Warning). ## 6. Add notes about the usage of RINGTONES by Avaya Vantage devices. ## 7. Avaya Vantage R2.0.0.0: ## a. The default of SNTPSRVR to "0.avaya.pool.ntp.org,1.avaya.pool.ntp.org,2.avaya.pool.ntp.org,3.avaya.pool.ntp.org". SNTPSRVR supports FQDN in R2.0.0.0 and later. ## b. All Avaya Vantage K165/K175 parameters are supported by K155. K155 is supported in R2.0.0.0 and later. ## c. BAKLIGHTOFF default was changed to 10 minutes. For K155 only the range is 1-60. ## d. Support new configuration parameters: USER_INSTALL_APPS_UNKNOWN_SOURCES, HTTPPROXYAUTOCONFIGURL and HTTPPROXYSOURCE. ## 8. Add note to TLSPORT and FILE_SERVER_URL that by default Utility Server uses TCP port 411 with https://. ## 9. Avaya Equinox 3.4 running on Avaya Vantage: ## a. Add the following applicable parameters for Avaya Equinox 3.4 when running on Avaya Vantage: ENFORCE_SIPS_URI. ## b. Add note to ENABLE_AVAYA_CLOUD_ACCOUNTS that since Avaya Equinox on Avaya Vantage does not support Avaya Spaces integration, this parameter shall be configured to 0. ## 10. Add note to ENABLE_EQUINOX_MEETING_ACCOUNT_DISCOVERY that since Avaya Equinox on Avaya Vantage does not support Avaya Equinox Meeting Online account, this parameter shall be configured to 0. ## 11. Remove HW_ENCODER_STAT configuration parameter as it not required by Avaya Vantage. ## 12. Remove RECORDINGTONE and EXTEND_RINGTONE from list of supported parameters by J100 SIP phones series. ## 13. Mark AWAY_TIMER_VALUE and AWAY_TIMER parameters as supported by J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only); J139 SIP R3.0.0.0 and later and 96x1 SIP R6.4 and later. ## 14. Mark LANGLARGEFONT as supported by 100 SIP R2.0.0.0 and later (J169/J179 only). ## 15. Remove RECORDINGTONE from list of parameters supported by 96x1 SIP R7.0.0 and later. ## 16. Re-add 1603, 1608 and 1616 H.323 configuration parameters. ## ## 07-Nov-2018: ## 1. Add ENABLE_OOD_RESET_NOTIFY and mark it as supported by J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later. ## 2. Add HTTP_PROXY_CSDK_ENABLE and mark is as supported by Avaya Equinox 3.4 and later. ## 3. Update the description of TIMEFORMAT. ## 4. Fix the second example of MEDIA_ADDR_MODE. ## 5. Add note to PHY2STAT that K155 supports values 0-1. ## 6. Mark ENABLE_RECORDING as supported by Avaya Vantage Devices R1.1.0.0 and later. ## 7. ACSPORT - remove copy/paste error of the default value. ## 8. Update note and example in HEADSET_PROFILE_NAMES. ## ## 11-Dec-2018: ## 1. Add note to LOCAL_LOG_LEVEL that supports range of 3-7 for Avaya Vantage. ## 2. Add note that SPEAKERSTAT is not supported by J129. ## 3. Mark that ENABLE_OPUS is supported by J100 SIP R2.0.0.0 and later for all environments (Aura, IP Office and 3PCC). ## 4. Explicitly mention that the second set of values for MATCHTYPE are supported by J129 SIP R1.0.0.0 (or R1.1.0.0), ## J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later). ## 5. J100 SIP R4.0.0.0 and later: ## a. Support the following parameters: DHCPSTDV6, 3PCC_SERVER_MODE, BLF_LIST_URI, CALL_PICKUP_FAC, CALL_PICKUP_BARGEIN_FAC, ## CALL_UNPARK_FAC, ALLOW_BLF_LIST_CHANGE, XSI_URL, XSI_CHANNEL_DURATION, XSI_HEARTBEAT, FORCE_XSI_USER_ID, FORCE_XSI_WEB_PASSWORD, ## PRIMARY_LINE_TYPE, PRIMARY_LINE_BARGE_IN_ENABLED, SCA1_ENABLED, SCA2_ENABLED, SCA3_ENABLED, SCA1_MAX_CALL_APPEARANCES, SCA2_MAX_CALL_APPEARANCES, SCA3_MAX_CALL_APPEARANCES, ## SCA1_SIPUSERID, SCA2_SIPUSERID, SCA3_SIPUSERID, SCA1_USERNAME, SCA2_USERNAME, SCA3_USERNAME, SCA1_PASSWORD, SCA2_PASSWORD, SCA3_PASSWORD, ## SCA1_EXTENSION, SCA2_EXTENSION, SCA3_EXTENSION, SCA1_BARGE_IN_ENABLED, SCA2_BARGE_IN_ENABLED, SCA3_BARGE_IN_ENABLED, SCA_LINE_SEIZE_DURATION, ## PROVIDE_SHARED_LINE_CONFIG, WAIT_FOR_CALL_OPERATION_RESPONSE, BW_ENABLE_DIR, BW_ENABLE_DIR_ENTERPRISE, BW_ENABLE_DIR_ENTERPRISE_COMMON, ## BW_ENABLE_DIR_GROUP, BW_ENABLE_DIR_GROUP_COMMON, BW_ENABLE_DIR_PERSONAL, BW_ENABLE_DIR_CUSTOM, BW_DIR_ENTERPRISE_DESCRIPTION, ## BW_DIR_ENTERPRISE_COMMON_DESCRIPTION, BW_DIR_GROUP_DESCRIPTION, BW_DIR_GROUP_COMMON_DESCRIPTION, BW_DIR_PERSONAL_DESCRIPTION, ## BW_DIR_CUSTOM_DESCRIPTION, BW_DIR_ENTERPRISE_EXTENSION, BW_DIR_ENTERPRISE_COMMON_EXTENSION, BW_DIR_GROUP_EXTENSION, ## BW_DIR_GROUP_COMMON_EXTENSION, BW_DIR_PERSONAL_EXTENSION, BW_DIR_CUSTOM_EXTENSION, DUAL_IPPREF, PRIVACY_SLAAC_MODE, SCEPENCALG, ## BLOCK_CERTIFICATE_WILDCARDS, WLAN_MAX_AUTH_FAIL_RETRIES ## b. Support IPv6 only (value 2) in IPV6STAT. Default value is changed to 1 (Dual Stack) from 0 (IPv4 only). ## c. Support default value 3 for DHCPSTAT. ## d. REUSETIME controls both DHCPv4 and DHCPv6 clients. ## e. Add note to SIGNALING_ADDR_MODE that the single IPv6 mode phone ignores SIGNALING_ADDR_MODE and SIP_CONTROLLER_LIST and selects the SIP controller's IPv6 addresses from SIP_CONTROLLER_LIST_2. ## f. Add note that SIP_CONTROLLER_LIST list is used on IPv4-only and dual mode phones (if SIP_CONTROLLER_LIST_2 is not provided). ## g. Add note to SIP_CONTROLLER_LIST_2 that it is used on it is used on IPv6-only phones to provide the list of SIPv6 servers and that SIPv4 servers are ignored in IPv6-only mode. ## IPv4 only phones use SIP_CONTROLLER_LIST. Dual mode phones use SIP_CONTROLLER_LIST if SIP_CONTROLLER_LIST_2 is not provided. ## h. Support DES_STAT==3 to not rely on end users to press "YES" to enforce DES as part of bootup of fresh endpoint. ## 6. Add double quotes to the example of SCEPPASSWORD. ## 7. Updated the list of phones which support TLSSRVR, TLSPORT and TLSDIR. ## 8. Mark CNGLABEL as not supported by J100 and 96x1 SIP phones. ## 9. Mark as supported by B189 R6.7.1: SLMSTAT, SLMCAP, SLMCTRL, SLMPERF, SLMPORT, SLMSRVR. ## ########################################################################################