###################################################################################### ## ## AVAYA IP TELEPHONE CONFIGURATION FILE TEMPLATE ## *** 13 Apr 2021 *** ## ## 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 R4.0.9.0 (J129, J139, J159, J169, J179, J189) ## 96x1 SIP R7.1.13.0 ## Avaya Vantage Devices SIP R3.0.0.3 (K155/K175) ## Avaya Vantage builtin Unified Communication Experience R3.0.0.3 (Known also as Avaya Vantage Connect) ## J159/J169/J179/J189 H.323 R6.8.5 ## 96x1 H.323 R6.8.5 ## B189 H.323 R6.8.5 ## Avaya Vantage Devices SIP R2.2.0.4 (K155/K165/K175) ## Avaya Vantage Connect Application SIP R2.2.0.4 ## Avaya IX Workplace 3.8 (running on Avaya Vantage Devices)(f.k.a Avaya Equinox) ## 96x1 H.323 R6.7.1 ## B189 H.323 R6.7.1 ## 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. ## ###################################################################################### ## ## IMPORTANT: It is recommended to use the 46xxsettings.txt file as reference for ## configuration of relevant parameters and avoid the use of the whole file ## as it is. This in order to reduce the file size, network traffic and ## time required for downloading and parsing of this file by the endpoints. ## ###################################################################################### ## ## 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 ## J159 - Supported by J100 SIP R4.0.3.0 and later, Supported by H.323 R6.8.5 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. ## J189 - Supported by J100 SIP R4.0.6.1 and later, Supported by H.323 R6.8.5 and later. ## H175 ## K155 - supported in R2.0.0.0 and later. ## K165 - supported up to R2.x version. ## K175 ## ## List of MODEL6 values for models which support MODEL6 as testable parameter in the ## configuration file (for example: IF $MODEL6 SEQ K155CW GOTO SETTINGSK155CW). ## K155CW (R3.0.0.0+) ## K175NW (R3.0.0.0+) ## K175CW (R3.0.0.0+) ## K175CN (R3.0.0.0+) ## ## Note: Avaya Vantage Connect 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. ## In R3.0.0.0 and later, there is also support for $MODEL6 and $GEN as testable parameters. $GEN equal to "B" used by Avaya Vantage Devices ## running R3.0.0.0 and later. ## ## Note: Avaya Vantage Connect 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 Connect 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. ## ## Note: Avaya Breeze Client SDK applications (e.g. Avaya IX Workplace, Avaya Vantage Connect, etc.) keep their previous configuration parameters values even if not explicitly configured in this file. ## The value is not returned to default when the parameter is removed or masked as on Avaya SIP hard endpoints (96x1/J100/Avaya Vantage). To return to default, the parameter shall be explicitly configured with "". ## ## 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. ## ###################################################################################### ## ## 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 ## ############### CONDITIONAL PARAMETERS ############## ## ## APPNAME_IN_USE specifies the current software version used by the phone. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 SIP R1.5.0 ## Example: ## IF $APPNAME_IN_USE SEQ 4.0.8.0.7 GOTO SPECIALHANDLING ## ############### 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later - if L2QVLAN == 0, L2Q is treated as 2 (disabled) ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Connect Application SIP R2.1.0.0 and later (This parameter may also be set via AADS or LLDP which would overwrite any value set in this file) ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.1.0.0 and later (This parameter may also be set via AADS or LLDP which would overwrite any value set in this file). ## 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: ## Avaya Vantage Connect Application SIP R2.1.0.0 and later (This parameter may also be set via AADS or LLDP which would overwrite any value set in this file). ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya IX Workplace R3.5.5 and later used in IP office environment only (for Aura environment ## DSCPAUD is taken from PPM and configured using SMGR). The parameter can be configured from AADS and LLDP which have higher precedence than this file. ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Connect Application SIP R1.1.0.1 and later; used in IP office and OpenSIP environments only (for Aura environment ## DSCPAUD is taken from PPM only and configured using SMGR). R2.2.0.0 and later - the parameter is supported in Aura, IP Office and OpenSIP environments. ## Parameter precedence from high to low: LLDP, AADS, 46xxsettings.txt file. The parameter is supported for IPv4 and IPv6 packets. ## 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 IX Workplace R3.5.5 and later used in IP office environment only (for Aura environment ## DSCPAUD is taken from PPM and configured using SMGR). The parameter can be configured from AADS and LLDP which have higher precedence than this file. ## Avaya Vantage Connect Application SIP R1.1.0.1 and later; used in IP office and OpenSIP environments only (for Aura environment ## DSCPVID is taken from PPM only and configured using SMGR). R2.2.0.0 and later - the parameter is supported in Aura, IP Office and OpenSIP environments. ## Parameter precedence from high to low: LLDP, AADS, 46xxsettings.txt file. The parameter is supported for IPv4 and IPv6 packets. ## 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: ## Avaya IX Workplace R3.5.5 and later used in IP office environment only (for Aura environment ## DSCPAUD is taken from PPM and configured using SMGR). The parameter can be configured from AADS and LLDP which have higher precedence than this file. ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Connect Application SIP R1.1.0.1 and later; used in IP office and OpenSIP environments only (for Aura environment ## DSCPSIG is taken from PPM only and configured using SMGR). R2.2.0.0 and later - the parameter is supported in Aura, IP Office and OpenSIP environments. ## Parameter precedence from high to low: LLDP, AADS, 46xxsettings.txt file. The parameter is supported for IPv4 and IPv6 packets. ## 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Devices SIP R3.0.0.0 and later ## J100 SIP R4.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; R2.2.0.0 and later supports IPv6 address as well. ## 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 ## SET DNSSRVR e9e4:35a:cef2::1 ## ## 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R2.2.0.4 and later; When SHOW_LAST_EXTENSION is set to 0, the SIP extension in case of SIP login or unified login username in case of unified login ## will NOT be displayed when doing logout or pressing "Cancel" during "logging-in" phase. When SHOW_LAST_EXTENSION is set to 1, the SIP extension in case of SIP login or ## unified login username in case of unified login will be displayed when doing logout or pressing "Cancel" during "logging-in" phase. ## In R3.0.0.0 and later, this parameter is only applicable when ENABLE_PLATFORM_LOGIN_SCREEN == 1. ## J100 SIP R2.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET SHOW_LAST_EXTENSION 1 ## ## PRESERVE_LOGIN_PASSWORD specifies whether to preserve SIP password in case of SIP login or unified login password in case of unified login when canceling logging-in. ## Value Operation ## 0 SIP password in case of SIP login or unified login password in case of unified login will NOT be stored/displayed when pressing "Cancel" during "logging-in" phase (default) ## 1 SIP password in case of SIP login or unified login password in case of unified login will be stored/displayed when pressing "Cancel" during "logging-in" phase (SHOW_LAST_EXTENSION MUST be set to "1" as well). ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.2.0.4 and later; In R3.0.0.0 and later, this parameter is only applicable when ENABLE_PLATFORM_LOGIN_SCREEN == 1. ## SET PRESERVE_LOGIN_PASSWORD 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. ## R2.0.1+ - only https:// is supported. http:// is not supported. ## R2.2.0.0 and later supports IPv6 address as well. ## 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 ## SET USER_AUTH_FILE_SERVER_URL https://e9e4:35a:cef2::2:8443 ## 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. ## ## USER_AUTH_FILE_SERVER_SSO specifies what authentication method is used when accessing AADS / user authenticated file server. ## Value Operation ## 1 Unified Credentials (default) ## 3 Avaya Authorization ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later, value 1 is supported from Avaya Vantage Devices SIP R1.0.0.0 and later, value 3 is supported from Avaya Vantage Devices SIP R3.0.0.0 and later. ## SET USER_AUTH_FILE_SERVER_SSO 3 ## ## AVAYA_AUTHORIZATION_REDIRECTION_LIST specifies a list of FQDNs of identity providers to which the endpoint can be redirected from AADS when USER_AUTH_FILE_SERVER_SSO is "3". ## The default value is "". ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later ## SET AVAYA_AUTHORIZATION_REDIRECTION_LIST example.shibboleth.com ## ## ENABLE_PLATFORM_LOGIN_SCREEN specifies whether user is required to provide SIP/user enterprise credentials before reaching Android home screen or not. ## Value Operation ## 0 User is not required to provide SIP/user enterprise credentials to reach Android home screen (default) ## 1 User is required to provide SIP/user enterprise credentials to reach Android home screen ## Note: Value 0 is mainly applicable for cases where Avaya Vantage Device is used by single end user. ## Note: Value 1 implies the same logging-in user experience as in Avaya Vantage Devices running pre R3.0.0.0. ## Note: Android lock screen can be enforced in both cases. When value 0 is configured, then Lock Password/PIN, etc. is configured by the end user. When value 1 is configured, then Lock Password is the SIP password ## (when SIP login is used (USER_AUTH_FILE_SERVER_URL=="")) or the user enterprise password (when user enterprise credentials login is used (USER_AUTH_FILE_SERVER_URL<>"")). ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later ## SET ENABLE_PLATFORM_LOGIN_SCREEN 1 ## ## ALLOW_LOGOUT_WHEN_LOCKED specifies whether end users/administrators will be able to logout an existing user when lock screen is presented. ## Value Operation ## 0 End users have no option to do logout of the existing user when the device is locked (default - in R2.0.1.0 and later). ## Administrator can do logout using settings application (administrator access) if PROVIDE_LOGOUT (R2.0.1.0 and later) is 1. ## 1 End users and administrators can do logout of the existing user (default in pre-R2.0.1.0). ## Administrator can do logout using settings application (administrator access) if PROVIDE_LOGOUT (R2.0.1.0 and later) is 1. ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later; not applicable when Avaya Vantage Open application is used. ## In pre R2.0.1.0, value 2 is supported - value 2 provides a logout option to administrator only using settings application (The logout option is available for administrator only also when ## the device is not locked, but logged-in). In R2.0.1.0 and later the Logout option in the settings application is controlled only by PROVIDE_LOGOUT parameter ## and users can use it if the device is not locked. In R3.0.0.0, this parameter is only supported when ENABLE_PLATFORM_LOGIN_SCREEN is 1. ## SET ALLOW_LOGOUT_WHEN_LOCKED 0 ## ## FAST_LOGIN_AFTER_BOOT specifies whether to accelerate the login process after reboot (applicable for Avaya Breeze Client SDK applications which supports fast login). ## Value Operation ## 0 No acceleration. ## 1 Only if the Active Breeze Client SDK Application is Avaya Vantage Connect, the login process is accelerated. This is the default value. ## 2 Login process is accelerated for any active Breeze Client SDK application. ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.2.0.0 and later ## SET FAST_LOGIN_AFTER_BOOT 2 ## ################### ONLINE HELP ################ ## ## ONLINE_HELP_URL specifies the URL from which Avaya Vantage will present help menus in the settings application. The default is "". ## When ONLINE_HELP_URL is "", then online help menus are used from Avaya support site. ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.1.0.0 and later; R2.2.0.0 and later supports IPv6 address as well. Link local IPv6 address is not supported. ## SET ONLINE_HELP_URL https://example.com/help_menu/ ## SET ONLINE_HELP_URL http://[e9e4:35a:cef2::1]/help/ ## ################### AVAYA AURA DEVICE SERVICES (AADS) CONTACTS SERVICES ################ ## ## 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 IX Workplace 3.1.2 and later ## Avaya Vantage Connect 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 IX Workplace 3.1.2 and later ## Avaya Vantage Connect Application SIP R1.0.0.0 and later; R2.2.0.0 and later supports IPv6 address as well. ## 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 IX Workplace 3.1.2 and later ## Avaya Vantage Connect 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 IX Workplace 3.1.2 and later ## Avaya Vantage Connect 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 IX Workplace 3.1.2 and later ## SET CONTACT_MATCHING_SEARCH_LOCATION 2 ## ## ACSSSO specifies whether Device Services uses Unified Login. ## Value Operation ## 0 Disable - user is required to enter Device Services credentials manually on the application. ## 1 Enable (default) ## 3 Avaya Authorization ## Note: ACSSSO is enforced to 1 by Avaya Vantage devices up to R2.0.1.0. Avaya Vantage devices running R2.0.1.0 and later do not restrict any ACSSSO value. ## Note: Avaya Vantage Connect supports ACSSSO==1 only. ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later (values 1 and 3 only), ACSSSO value "3" requires USER_AUTH_FILE_SERVER_SSO to set to "3". ## Avaya IX Workplace 3.0 and later (values 0-1); value 3 is supported in 3.6 and later. ## SET ACSSSO 0 ## ################### AVAYA MULTIMEDIA MESSAGING ################ ## ## 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 IX Workplace 3.1.2 and later ## SET ESMENABLED 1 ## ## ENABLE_IM 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 Vantage built-in Unified Communication Experience R3.0.0.3 and later; when ENABLE_IM is "0", there will be no IM tab in Avaya Vantage built-in Unified Communication Experience. ## When ENABLE_IM is "0" or ACTIVE_CSDK_BASED_PHONE_APP <> "com.avaya.android.vantage.basic", there will be no IM icon on Android home screen. ## SET ENABLE_IM 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 IX Workplace 3.1.2 and later ## Note: Avaya Vantage built-in Unified Communication Experience R3.0.0.3 and later supports value 0 only. ## SET ESMHIDEONDISCONNECT 1 ## ## ESMSRVR specifies IP address or FQDN of Avaya Multimedia Messaging server. Default value is "". ## This parameter is supported by: ## Avaya IX Workplace 3.1.2 and later ## Avaya Vantage built-in Unified Communication Experience R3.0.0.3 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 IX Workplace 3.1.2 and later ## Avaya Vantage built-in Unified Communication Experience R3.0.0.3 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 IX Workplace 3.1.2 and later ## Avaya Vantage built-in Unified Communication Experience R3.0.0.3 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 IX Workplace 3.1.2 and later ## Avaya Vantage built-in Unified Communication Experience R3.0.0.3 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 IX Workplace 3.1.2 and later ## SET ADDRESS_VALIDATION 1 ## ## ESMSSO specifies whether Messaging uses Unified Login. ## Value Operation ## 0 Disable - user is required to enter Messaging credentials manually on the application. ## 1 Enable (default) ## 3 Avaya Authorization ## Note: ESMSSO is enforced to 1 by Avaya Vantage devices up to R2.0.1.0. Avaya Vantage devices running R2.0.1.0 and later do not restrict any ESMSSO value. ## This parameter is supported by: ## Avaya IX Workplace 3.0 and later ## Avaya Vantage built-in Unified Communication Experience R3.0.0.3 and later; ESMSSO value "3" requires USER_AUTH_FILE_SERVER_SSO to be set to "3". ## SET ESMSSO 0 ## ## ESMSENDREADRECEIPTS specifies whether to enable or disable the use of read receipts for all users. ## Value Operation ## 0 Indicates that read receipts are disabled. ## 1 Indicates that read receipts are enabled. This is the default value. ## This parameter is supported by: ## Avaya IX Workplace 3.7 and later ## Note: Avaya Vantage built-in Unified Communication Experience R3.0.0.3 and later supports value 0 only. ## SET ESMSENDREADRECEIPTS 0 ## ################### EXCHANGE WEB SERVICES (EWS) ################ ## ## 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 Vantage Connect Application SIP R2.2.0.0 and later ## Avaya IX Workplace 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 Vantage Connect Application SIP R2.2.0.0 and later; not applicable when EWSSSO == 4. ## Avaya IX Workplace 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 Vantage Connect Application SIP R2.2.0.0 and later; not applicable when EWSSSO == 4. ## Avaya IX Workplace 3.1.2 and later ## SET EWSDOMAIN "avaya.com" ## ## EWSSSO specifies whether EWS uses Unified Login. ## Value Operation ## 0 Disable - user is required to enter EWS credentials manually on the application. ## 1 Enable (default) ## 4 Microsoft Modern Authentication ## Note: EWSSSO is enforced to 1 by Avaya Vantage devices up to R2.0.1.0. Avaya Vantage devices running R2.0.1.0 and later do not restrict any EWSSSO value. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later; value 4 is supported in R2.2.0.2 and later. ## Avaya IX Workplace 3.1 and later ## SET EWSSSO 0 ## ## ENABLE_CALENDAR specifies whether Calendar tab is presented or not. ## Value Operation ## 0 Calendar tab is hidden ## 1 Calendar tab is presented ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later; the default is 0 for pre R3.0.0.0 and 1 for R3.0.0.0 and later ## SET ENABLE_CALENDAR 1 ## ################### UNIFIED PORTAL ################ ## ## 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 Vantage built-in Unified Communication Experience R3.0.0.0 and later, “Start my Meeting” icon will be presented when UNIFIEDPORTALENABLED is "1", UNIFIED_PORTAL_SSO is "1" and ## CONFERENCE_PORTAL_URI is valid. ## Avaya IX Workplace 3.2 and later ## SET UNIFIEDPORTALENABLED 1 ## ## UNIFIED_PORTAL_SSO specifies whether Unified Portal uses Unified Login. ## Value Operation ## 0 Disable - user is required to enter Unified Portal credentials manually on the application. ## 1 Enable (default) ## Note: UNIFIED_PORTAL_SSO is enforced to 1 by Avaya Vantage devices up to R2.0.1.0. Avaya Vantage devices running R2.0.1.0 and later do not restrict any UNIFIED_PORTAL_SSO value. ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later, only value "1" is supported. When value "0" is configured, then “Start my Meeting” icon will not be presented. ## Avaya IX Workplace 3.2 and later ## SET UNIFIED_PORTAL_SSO 0 ## ################### 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. ## Value Operation ## 0 Disabled ## 1 Enabled (default) ## This parameter is supported by: ## Avaya IX Workplace 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. ## Value Operation ## 0 Avaya Spaces integration is disabled ## 1 Avaya Spaces integration is enabled (default) ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later ## Avaya IX Workplace 3.4 and later ## SET ENABLE_AVAYA_CLOUD_ACCOUNTS 0 ## ## ENABLE_SPACES_MESSAGING specifies whether to enable or disable Spaces Messaging (Avaya IP Office deployments). ## Value Operation ## 0 Avaya Spaces Messaging is disabled ## 1 Avaya Spaces Messaging is enabled (default) ## This parameter is supported by: ## Avaya IX Workplace 3.7 and later ## SET ENABLE_SPACES_MESSAGING 0 ## ## ENABLE SPACES DASHBOARD specifies whether to present the "Spaces Dashboard" icon. ## Value Operation ## 0 "Spaces Dashboard" icon is not presented ## 1 "Spaces Dashboard" icon is presented (default) ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later ## SET ENABLE SPACES DASHBOARD 0 ## ## SPACES_CALL_END_NAV specifies which screen is presented when there is end of Avaya Spaces call. ## Value Operation ## 0 Stay in Avaya Spaces once Avaya Spaces call end ## 1 Navigate to the Avaya Vantage home screen once Avaya Spaces call end ## 2 Navigate back to the application which called Avaya Spaces once Avaya Spaces call end ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.1 and later; Up to R3.0.0.3 (excluded) - the default is 0, R3.0.0.3 and later the default was changed to 2. ## SET SPACES_CALL_END_NAV 1 ## ################### APPLICATION RATING ################ ## ## ENABLE_IN_APP_RATING specifies whether the user can view the option to rate the application. ## Value Operation ## 0 The user cannot view the option to rate the application. ## 1 The user can view the option to rate the application (default). ## This parameter is supported by: ## Avaya IX Workplace 3.7 and later ## SET ENABLE_IN_APP_RATING 0 ## ################### LDAP/S CLIENT CONFIGURATION ################ ## ## DIRENABLED_PLATFORM specifies whether the Avaya Vantage LDAP/S client is enabled or disabled. ## Value Operation ## 0 LDAP/S client is disabled (default) ## 1 LDAP/S client is enabled ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later ## SET DIRENABLED_PLATFORM 1 ## ## DIRUSERNAME specifies LDAP/S client username. It is not recommended to configure LDAP/S client username in the 46xxsettings.txt file ## if HTTPS protocol is not used and if there is no HTTP authentication or mutual certificate based authentication else the 46xxsettings.txt ## file will be retrieved by anyone. For J100 SIP phones, this parameter can be configured securely using WEB interface. ## The default is "". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later; the following characters are allowed: 0 - 9, a - z, A - Z ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later ## SET DIRUSERNAME directoryUsername ## ## DIRPASSWORD specifies LDAP/S client password. It is not recommended to configure LDAP/S client password in the 46xxsettings.txt file ## if HTTPS protocol is not used and if there is no HTTP authentication or mutual certificate based authentication else the 46xxsettings.txt ## file will be retrieved by anyone. For J100 SIP phones, this parameter can be configured securely using WEB interface. ## The default is "". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later, the following characters are allowed: 0 - 9, a - z, A - Z. ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later ## SET DIRPASSWORD directoryPassword ## ## DIRSRVR specifies the IP address or fully qualified domain name of LDAP/S server. The default is "". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later; R2.2.0.0 and later supports IPv6 address as well. ## SET DIRSRVR ldap.customer.com ## ## DIRSRVRPRT specifies the port number of the LDAP/S server. ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, the default is 389 in R4.0.3.0 up to R4.0.4.0 (excluded) and 636 in R4.0.4.0 and on, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later; the default is 636. ## SET DIRSRVRPRT 640 ## ## DIRTOPDN specifies the LDAP search base. The default is "". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## Avaya Vantage Devices SIP R2.0.1.0 and later ## SET DIRTOPDN "OU=Global Users,dc=global,dc=company,dc=com" ## ## DIRSECURE specifies whether to use TLS or TCP for LDAP/S. ## For server authentication, the trusted certificate file shall be placed on an HTTP or HTTPS server and include certificate file name in the TRUSTCERTS parameter value. ## Client authentication uses identity certificate installed using SCEP or PKCS12. ## STARTTLS uses the same port as an LDAP protocol. DIRSRVRPRT must be the same as a port configured for LDAP (not for LDAPS) protocol on the server side - by default, 389. ## LDAPS protocol uses a different port from LDAP. Therefore, DIRSRVRPRT need to be configured to the server port for LDAPS connection - by default, 636. ## Value Operation ## 0 LDAP over TCP ## 1 Avaya Vantage - LDAPS over TLS (default) or LDAP Start TLS (RFC 4513) per DIRSTARTTLS. ## J100 - LDAP Start TLS (RFC 4513) ## 2 LDAPS over TLS ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 up to R4.0.4.0 (excluded) (values 0-1; default is 1, value 1 - establish TLS connection using STARTTLS extended operation) ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later (values 0-2; default is 2, value 1 - establish TLS connection using STARTTLS extended operation, ## value 2 - establish TLS connection using Secure LDAP protocol (LDAPS)). ## 96x1 SIP R7.1.10 and later (values 0-2; default is 2) ## Avaya Vantage Devices SIP R2.0.1.0 and later (values 0-1 only, 1 is default) ## SET DIRSECURE 0 ## ## DIRSTARTTLS specifies whether to use LDAPS over TLS or LDAP Start TLS (RFC 4513). ## Value Operation ## 0 LDAPS over TLS (default) ## 1 LDAP Start TLS (RFC 4513) ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.2.0.0 and later ## SET DIRSTARTTLS 1 ## ## DIRREFERRALS specifies whether to enable LDAP/S referrals. ## Value Operation ## 0 LDAP/S referrals are not supported. ## 1 LDAP/S referrals are supported (default). ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.1.0 and later ## SET DIRREFERRALS 0 ## ## DIRSEARCH_FIELDS specifies the LDAP attributes that will be searched. The default is "cn,sn,telephoneNumber". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIRSEARCH_FIELDS "givenName,mail,middle initials,telephoneNumber,sn,mobile,o,department,Rank,office,DoD SIP URI". ## ## DIRSHOW_FIELDS specifies the LDAP attributes to show in the detail view. This parameter is also used for mapping specified LDAP attributes to user-friendly display fields. ## The format of each DIRSHOW_FIELDS parameter shall be as follows: ## SET DIRSHOW_FIELDS "[LDAP Attributes]=[Field Names],[LDAP Attribute 1]=[Field Name1]" ## The default is "cn,sn,telephoneNumber,mail". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIRSHOW_FIELDS "dn=Distinguished Name,rank,gn=First Name,office=Office,middle initials=Middle Initial,Display Name=Full Name,sn=Last Name,job title=Job,cn=Common Name,o=Office,c=Country,department=Department,street=Street,mail=Mail Box,l,telephoneNumber=Phone number,st,mobile=Mobile,postalCode=Postal code,facsimileTelephoneNumber=Fax,DoD SIP URI=Number" ## ## DIRNAME_FIELDS specifies the LDAP attributes used for name in the search results and the order they will be displayed. The default is "cn". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIRNAME_FIELDS "cn,sn". ## ## DIRNUMBER_FIELDS specifies the LDAP attributes that contain a callable number. The primary number is the first attribute listed in this config. ## The default is "telephoneNumber". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIRNUMBER_FIELDS "telephoneNumber,mobile,DoD SIP URI" ## ## DIRAUTHTYPE specifies the authentication method when a username is provided in DIRUSERNAME parameter. ## Value Operation ## 0 Authentication using LDAP simple authentication (default). Normally DIRUSERNAME parameter should contain DN name of an LDAP record, and DIRPASSWORD should contain password associated with the record. ## 1 Authentication using Simple Authentication and Security Layer (SASL). ## If a connection established over TLS (DIRSECURE set to 1 or 2), then both digest (MD5) and basic (clear) authentication mechanisms are supported. ## If the connection established over TCP (DIRSECURE set to 0), then digest (MD5) authentication algorithm is only supported. ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIRAUTHTYPE 1 ## ## DIR_TO_LOCAL_MAPPING specifies a mapping of LDAP fields to local contact fields. The entire contact mapping is considered invalid ## if there is no valid rule for either first name or last name or there is no valid rule for at least one contact number. ## Local contact fields names can be assigned from the following list: "lastName", "firstName", "nickname", "URI", "extension", ## "email", "department", "zipCode", "country". Number types list: "work", "home", "mobile", "other". ## The default value is "fn=firstName,ln=lastName,cn=nickname,telephoneNumber=work". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIR_TO_LOCAL_MAPPING "firstName=firstName,lastName=lastName,BusinessPhoneNumber=work" ## ## DIR_LDAP_DESCRIPTION specifies a custom label to be used for the LDAP directory in the Contacts application. ## When DIR_LDAP_DESCRIPTION is "" (Default value), then the default label presented is "LDAP Directory". ## This parameter is supported by: ## J139, J159, J169, J179 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.10 and later ## SET DIR_LDAP_DESCRIPTION "Users from LDAP" ## ## DIR96X1_CONTACT_SOURCE_DEFAULT specifies the default directory to be used when a contact search is performed. ## Value Operation ## 0 System (default) ## 1 LDAP ## This parameter is supported by: ## 96x1 SIP R7.1.10 and later ## SET DIR96X1_CONTACT_SOURCE_DEFAULT 1 ## ################### 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.6.2 and later releases ## SET H323SIGPROTOCOL 1 ## ## GCFIPADDRREPORT specifies which IP address to be reported to the H.323 call server. The default value is 0. ## Note: In cases where the H.323 endpoint registers to H.323 call server NOT behind NAT after being registered to H.323 call server behind NAT (or vice versa), then GCFIPADDRREPORT shall be set to 1. ## Value Operation ## 0 If DiscoveryResponse was received in the past, then the H.323 endpoint reports the IP address in the DiscoverResponse else report the endpoint IP address. ## 1 If GCF with DiscoverResponse was received by the H.323 endpoint, then the endpoint is behind NAT and shall use the IP address reported in the DiscoverResponse. ## If endpoint gets GCF without DiscoverResponse then the endpoint is NOT behind NAT and shall use its own IP address. ## Note: The IP address is reported in the rasAddress and CallSignalAddress fields in the RRQ message. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.7.1 and later ## B189 H.323 R6.7.1 and later ## SET GCFIPADDRREPORT 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: ## J100 SIP R2.0.0.0 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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 ## ## ARP_REQUEST_LIMIT specifies the number of ARP requests per second from a single IP address to be replied by the H.323 endpoint. ## Range is 2-10 with default value 2. ## Up to 10 different IP addresses are supported at a time. ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.7.1 and later ## B189 H.323 R6.7.1 and later ## SET ARP_REQUEST_LIMIT 3 ## ######### 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, J159 and J189 H.323 R6.8.5 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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, J159 and J189 H.323 R6.8.5 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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, J159 and J189 H.323 R6.8.5 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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. ## R3.0.0.0 and later the default is "com.avaya.android.vantage.basic". No support of "com.avaya.android.flare". ## Note: For Avaya Vantage Connect Application ## SET ACTIVE_CSDK_BASED_PHONE_APP "com.avaya.android.vantage.basic" ## Note: For Avaya IX Workplace 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 Connect 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; R2.2.0.0 and later supports IPv6 address as well. ## SET PUSH_APPLICATION "com.avaya.android.vantage.basic.apk,flare-android.apk,clock.apk,outlook.apk" ## SET PUSH_APPLICATION "https://[e9e4:35a:cef2::a018]:8000/example_dir/example.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 Connect application) support special handling of pin after initial login ## to prevent pinning of the application without having the login screen. ## Avaya Vantage Connect 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 Connect Application User preferences menu using ADMIN_PASSWORD if configured, else PROCPSWD if configured. In R2.1.0.0, ## there is support for complex administrator password configured in SMGR. If it is configured then complex administrator password configured in SMGR will take precedence, else 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. There is an option to exit from the Avaya Vantage Android Launcher ## using complex administrator password configured in SMGR (R2.1.0.0 and later), else ADMIN_PASSWORD if configured, else PROCPSWD if configured. ## 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 Connect 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" ## The above example is for case where Avaya Android launcher is used (com.avaya.endpoint.avayakiosk). The applications that are pinned are: Avaya Vantage Connect Application and the Android Calculator. ## com.avaya.endpoint.login is used to control whether the logout icon is presented or not. If PIN_APP includes com.avaya.endpoint.login, then the logout icon will be displayed. ## In Pre R3.0.0.0, lock icon is presented if ENABLE_PHONE_LOCK is "1". In R3.0.0.0 and later lock icon is presented if ENABLE_PHONE_LOCK is "1" AND ENABLE_PLATFORM_LOGIN_SCREEN is "1" OR if the "Screen lock" field ## in the settings application <> "None" and ENABLE_PLATFORM_LOGIN_SCREEN is "0". ## SET PIN_APP com.avaya.android.vantage.basic,com.avaya.endpoint.avayakiosk,com.avaya.vantageremote ## The above example is for using Avaya Vantage in Kiosk mode where only Avaya Vantage Connect and Avaya Vantage Expansion applications are available. ## ## DEFAULT_PIN_APP specifies which application out of the applications to be pinned shall be presented after reboot/powerup. The android package name of this application shall be configured. ## The default value is "". ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.1 and later and R2.2.0.5 and later. ## SET DEFAULT_PIN_APP "com.avaya.android.vantage.basic" ## SET DEFAULT_PIN_APP "com.avaya.vantageremote" ## ## 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; R2.1.0.0 - this parameter is used to control installation of applications from Google Play store and from unknown sources (e.g. Browser, USB mass storage device, etc.). ## R2.2.0.0 and later supports IPv6 address as well. ## SET APPS_CONTROL_FILE https://149.49.77.1/appcontrol.xml ## SET APPS_CONTROL_FILE ../appcontrol.xml ## SET APPS_CONTROL_FILE https://[e9e4:35a:cef2::a018]:8000/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 up to R3.0.0.0 (excluded). Avaya Vantage Devices SIP R3.0.0.0 do not include Google Mobile Services (GMS) by default. When Google Play Store/Google Play Services are sideloaded by ## settings GOOGLE_APPS to 2 or settings GOOGLE_APPS to 3 (and user choose to install Google Play Store/Google Play Services), then there will be no option to disable Google Play Store. ## Note: The parameter is also used by "Avaya Vantage Connect Application" to present "Rate us" menu only when ## USER_INSTALL_APPS_GOOGLE_PLAY_STORE is set to 1. The note is applicable for Avaya Vantage Devices SIP R1.0.0.0 to R3.0.0.0 (Excluded). ## SET USER_INSTALL_APPS_GOOGLE_PLAY_STORE 0 ## ## GOOGLE_APPS specifies whether to install or uninstall Google Play Store/Google Play Services. ## Value Operation ## 0 No change - If Google Play Store/Google Play Services are installed then they will remain installed. No new Google Play Store/Google Play Services installation (default) ## 1 Uninstall of Google Play Store/Google Play Services if installed. Return to factory defaults is triggered at the end of the uninstallation. ## 2 Install Google Play Store/Google Play Services. Internet Connectivity is required. ENABLE_PUBLIC_CA_CERTS shall be set to "1" or TRUSTCERTS shall be configured to include the root CA certificate of the identity certificates ## of des.avaya.com AND sourceforge.net servers. ## 3 Provide to the end user an option to install / remove Google Play Store/Google Play Services. If Google Play Store/Google Play Services are not installed, then the installation wizard provides an option to install ## Google Play Store/Google Play Services. The settings application provides an option to install / remove Google Play Store/Google Play Services. ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later; value 3 is supported in R3.0.0.3 and later. ## Note: Installation or uninstallation of Google Play Store/Google Play Services is system update and cannot be done without access to the HTTP/HTTPS file server which includes Avaya Vantage Device Software. ## SET GOOGLE_APPS 2 ## ## 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 ## R2.0.1.0 - ## 0 - All applications which can be used as source for unknown applications will NOT be able to download/install applications from unknown sources. ## User cannot change the status in the settings application for any such application. ## 1 - All applications which can be used as source for unknown applications will have this flag as disabled by default and users can change the status of this flag in the settings application. ## Value 2 is as 1. Default value is 1. ## 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 "" ## ################### 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 (OpenSIP) support is only provided by J129 SIP R1.1.0.0, J100 SIP R2.0.0.0 and later, Avaya Vantage Connect R2.0.1.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya IX Workplace 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 Connect 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, udp or auto. ## "auto" means that the preferred transport is determined based on Name Authority Pointer (NAPTR) record retrieved for the SIP Controller FQDN configured. SRV records can be then used to retrieve port and the final FQDN for use. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; DNS name format is supported for OpenSIP environment only ## For OpenSIP environment, only one SIP controller is supported. ## "auto" is supported in OpenSIP environment only. "auto" is supported by J100 SIP R4.0.0.0 and later. ## 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 OpenSIP environment. In OpenSIP 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 IX Workplace 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. ## IPv6 address is supported as well for Aura environment only. SIP_CONTROLLER_LIST_2 is not supported. ## Avaya Vantage Connect Application SIP R1.0.0.0 and later; DNS name format is supported; TCP/TLS are supported in Avaya Aura, Avaya IP Office and OpenSIP environments. UDP is supported in OpenSIP environment only. ## OpenSIP environment is supported from R2.0.1.0 and later. 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. R2.2.0.0 and later - IPv6 address is supported as well for Aura environment only. Please note that for dual stack controllers (IPv4/IPv6), ## then FQDN shall be used to avoid multiple registration to the SAME controller over both IPv4 and IPv6 addresses. SIP_CONTROLLER_LIST_2 is not supported. ## 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 ## SET SIP_CONTROLLER_LIST proxy.example.com;transport=auto ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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" ## ## MAX_DNS_DISCOVERED_SIP_CONTROLLERS specifies the maximum number of sip controllers to be used for redundancy from DNS lookups. ## The parameter is applicable only when 3PCC_SERVER_MODE = 3 (Netsapiens). For BroadSoft number of controllers is always 2, for other modes it's 1. ## For Netsapiens mode it is accepted to take value between 2 and 6 (2 is the default). ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## SET MAX_DNS_DISCOVERED_SIP_CONTROLLERS 3 ## ## 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, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later and J189 SIP R4.0.6.1 and later only for OpenSIP 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later, the default is simultaneous ## The parameter shall be configured to "alternate" in IP Office and OpenSIP 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. The parameter is mainly used in Aura environment. ## Valid values are 1, 2 or 3; the default value is 3. ## This parameter is supported by: ## Avaya IX Workplace 3.4 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Connect 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## ## ENABLE_PPM parameter enables PPM. ## Value Operation ## 0 disable PPM ## 1 enable PPM (default). ## This parameter is supported by: ## Avaya IX Workplace 3.4 and later ## SET ENABLE_PPM 0 ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; for IP office and OpenSIP 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 ## ## SMGR_AUTO_FAVORITE specifies whether all the Avaya Aura System Manager (SMGR) configured Features and supported Autodials, regardless of them being marked as Favorite or not, are put onto the J139, J159, J169, J179 or J189 Phone screen. ## This will only be applied to the phone if the phone is connected to Aura 8.0.0.0 or earlier. ## Value Operation ## 0 do not auto favorite (default) ## 1 auto favorite ## This parameter is supported by: ## J100 SIP R4.0.0.1 and later (J169/J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## SET SMGR_AUTO_FAVORITE 1 ## ## ENABLE_SIPURI_HOST_VALIDATION specifies whether to accept SIP URI with unrecognized host part in INVITE message. ## Value Operation ## 0 Accept SIP URI with unrecognized host part in INVITE message ## 1 Do not accept SIP URI with unrecognized host part in INVITE message (default) ## The parameter is supported in ALL environments (Aura, IPO and OpenSIP). ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## SET ENABLE_SIPURI_HOST_VALIDATION 0 ## ######### 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R6.0 and later ## 96x0 SIP R2.0 and later ## H1xx SIP R1.0 and later ## SET PHNNUMOFSA 3 ## ## ENABLE_DIGIT_MAPPING specifies if DIGIT_MAPPING config parameter will be used for dial plan configuration, if the parameter is disabled DIALPLAN and ELD config parameters will be used. ## Value Operation ## 0 DIALPLAN and ELD rules will be applied to the dial number (default). ## 1 DIGIT_MAPPING rules will be applied to the dial number. ## The parameter ENABLE_DIGIT_MAPPING is not applied in Avaya Aura and Avaya IP Office (CCMS) environments. ## This parameter is supported by: ## J100 SIP R4.0.6.0 and later, J189 SIP R4.0.6.1 and later ## SET ENABLE_DIGIT_MAPPING 1 ## ## DIGIT_MAPPING specifies a digit map which can be used to match digits to ensure a complete number is dialed, transform dialed digits, and block numbers from being dialed. ## ',' is used for rules separation. The following elements can be used to configure the rules: ## "Literals" - matches digit sequences with exactly the same literals. ## "x" - wildcard matches any character ## ". Matching Function" - matches 0 or more of the previous element ## "[ ] Set" - enclose a set of numerals and/or characters that are used to match a single digit or character. ## "[^ ] Exclusion Set" - An exclusion set matches any single alphanumeric character that is not within the set. ## "! Call Bar" - To block users from calling numbers that match a rule, add an exclamation mark - ! - in front of that rule in the digit map. ## "< elements : literals >" - Element to Literal Transformation allows replacing of characters sequence matching elements with the given literals. Use this rule to remove digits from a dialed number, add digits to a dialed number, or transform a dialed number ## The default is "". ## The parameter DIGIT_MAPPING is not applied in Avaya Aura and Avaya IP Office (CCMS) environments. ## For more information please refer to the administration guide. ## This parameter is supported by: ## J100 SIP R4.0.6.0 and later, J189 SIP R4.0.6.1 and later ## SET DIGIT_MAPPING "!99xx , <189:88>xx" ## ## PRESERVE_MEDIA_CONNECTIONS specifies whether to preserve media connections for servers which do not comply with RFC 3264 8.2 Removing a Media Stream. The parameter is supported in OpenSIP environment only. ## Value Operation ## 0 Do not preserve media connections (default) ## 1 Preserve media connections. ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET PRESERVE_MEDIA_CONNECTIONS 1 ## ################## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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. ## R2.2.0.0 and later supports IPv6 address as well. ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## H1xx SIP R1.0 and later ## SET SNTPSRVR 192.168.0.5 ## SET SNTPSRVR e9e4:35a:cef2::2 ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 SIP R2.5 and later ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; valid values are 30 to 86400; the default value is 900 ## Avaya Vantage Connect 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: ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later. The parameter is ## only supported in non-Aura environments (in Aura environment the setting comes from PPM). ## 96x1 SIP R6.0-R6.1, not supported in R6.2 and later as the setting comes from PPM ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 OpenSIP 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J139 SIP R3.0.0.0 and later and J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later only for OpenSIP 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, J139 SIP R3.0.0.0 and later and J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later only for OpenSIP 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, J139 SIP R3.0.0.0 and later and J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later only for OpenSIP 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. The parameter is supported in Aura and OpenSIP environments. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.0.0 and later ## SET FORBIDDEN_SESSION_REMOVAL_TIMER 5 ## ## CALL_SESSION_TIMER_EXPIRATION specifies the SIP session timer in seconds per RFC 4028. ## Value Operation ## 0 Disabled (Default) ## 90-65535 Value to be set in Session-Expires header ## Note: The recommended value per RFC 4028 is 1800 (30 minutes). ## Note: The parameter is supported in OpenSIP (BroadSoft) environment. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later ## SET CALL_SESSION_TIMER_EXPIRATION 1800 ## ############# CONFERENCING SETTINGS (SIP ONLY) ############# ## ## CONFERENCE_FACTORY_URI specifies the URI for Avaya Aura Conferencing or Network Conferencing in OpenSIP 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 Connect 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya IX Workplace 3.1.2 and later (for IPO environment only). ## 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 IX Workplace 3.1.2 and later ## SET CONFERENCE_ACCESS_NUMBER "15112345678" ## ## CONFERENCE_PORTAL_URI specifies the URI of the Conference Portal. The default value is null (""). ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later ## Avaya IX Workplace 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 IX Workplace 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 IX Workplace 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 IX Workplace 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 Vantage Connect Application SIP R2.2.0.0 and later ## Avaya IX Workplace 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 IX Workplace 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.2.0.3 and later (default is 1), supported in both Avaya Aura and Avaya IP Office environments. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later (default is 1); For IP office and OpenSIP 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, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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. The parameter is applicable for Aura environment only. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later, the parameter is NO longer supported ## in R4.0.8.0. ## 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 ## 2 Activate the SendAllCall when user enables DoNotDisturb / Activate DoNotDisturb when SendAllCall is enabled. ## 3 Forced do not activate the SendAllCall when user enables DoNotDisturb ## 4 Forced activate the SendAllCall when user enables DoNotDisturb ## 5 Forced activate the SendAllCall when user enables DoNotDisturb / Activate DoNotDisturb when SendAllCall is enabled. ## This parameter is supported by: ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later (J169/J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later; The value of this parameter is used if the ## ALLOW_DND_SAC_LINK_CHANGE is set to 0. Values 2-5 are supported by J100 SIP R4.0.8.0 (except J129). ## 96x1 SIP R6.4 and later; The value of this parameter is used if the ALLOW_DND_SAC_LINK_CHANGE is set to 0. Values 0-1 are supported. ## Avaya IX Workplace 3.1.2 and later; Values 0-1 are supported. ## Avaya Vantage Connect Application SIP R2.2.0.3 and later; Values 0-1 are supported. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 10 minutes. ## The parameter is supported in Aura environment only. ## This parameter is supported by: ## Avaya IX Workplace 3.1.2 and later; ## Avaya Vantage Connect Application SIP R2.2.0.3 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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, J100 SIP R2.0.0.0 and later (J169/J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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 (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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later; 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_METHOD_DEFAULT specifies the Exchange authentication method configured by an administrator. ## When Basic authentication (Forced) or OAuth (Forced) is configured, user is not allowed to change the authentication method from phone user interface. The Exchange authentication method will be per administrator configuration. ## When Basic authentication or OAuth is configured, user can change the administrator configured authentication method from phone user interface. ## Value Operation ## 0 Basic authentication (Default) ## 1 OAuth ## 2 Basic authentication (Forced) ## 3 OAuth (Forced) ## This parameter is supported by: ## 96x1 SIP R7.1.11.0 and later (9601 does not support this parameter) ## J100 SIP R4.0.7.0 and later (J129 does not support this parameter) ## SET EXCHANGE_AUTH_METHOD_DEFAULT 1 ## ## EXCHANGE_USER_ACCOUNT_DEFAULT specifies the Exchange user account configured by an administrator. This parameter is only applicable when authentication method is OAuth. ## If phone user hasn't configured any user name on the phone user interface then value configured in this parameter would be used. ## The value can contain 0 to 255 characters; the default value is null (""). ## This parameter is supported by: ## 96x1 SIP R7.1.11.0 and later (9601 does not support this parameter) ## J100 SIP R4.0.7.0 and later (J129 does not support this parameter) ## SET EXCHANGE_USER_ACCOUNT_DEFAULT john ## ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## 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), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later; 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; the parameter is not supported by J129. ## 96x1 SIP R6.0 and later ## 96x0 SIP R1.0 and later ## SET SPEAKERSTAT 1 ## ## AUDIOPATH_DEFAULT Specifies a default audio path. ## Note: The parameter is supported in ALL environments (Aura, IP Office and OpenSIP) ## Value Operation ## 1 Speaker (default) ## 2 Headset ## 3 Speaker (Forced, user cannot change this configuration) ## 4 Headset (Forced, user cannot change this configuration) ## This parameter is supported by: ## J100 SIP R4.0.6.0 and later (with exception of J129), J189 SIP R4.0.6.1 and later ## SET AUDIOPATH_DEFAULT 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; ## For OpenSIP environments using tls signaling, ENFORCE_SIPS_URI must be set to 0. ## Avaya IX Workplace 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, *, #). "," is used for one second delay. ## The default value is null (""). When Hotline is configured, then any off hook event will dial to the number specified in HOTLINE. No option to dial to other numbers. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later; no access to Avaya Vantage Connect settings menus. ## J169/J179 SIP R1.5.0 ## 96x1 SIP R6.2 and later ## SET HOTLINE "" ## ## HOTLINE_CALL_TYPE specifies whether hotline call type is audio or video. ## Value Operation ## 0 Audio Call (default) ## 1 Video Call ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET HOTLINE_CALL_TYPE 1 ## ## HOTLINE_ADMIN_MESSAGE specifies admin's message displayed on dialer screen when Hotline is activated (HOTLINE <>""). ## Valid values: On K165/K175 - up to 255 characters. On K155 - up to 68 characters. ## The default value is "Lift handset to place a call to Hotline number". ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET HOTLINE_ADMIN_MESSAGE "Hello. This is a test message." ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## ## PHONEKEY specifies list of pre-config keys. This parameter is used for mapping feature, application, line and autodial keys available in the Phone screen. ## The format of each PHONEKEY parameter shall be as follows: ## SET PHONEKEY "Key=[n1];Type=[Feature|Application|Line|Autodial];Name=[name];attr1=[value];attr2=[value];Label=[label (optional)][;Forced (optional)] ## [n1] corresponds to the number of the phone key to be configured. The allowed values are positive integers from 1 to 96. ## [Feature|Application|Line|Autodial] corresponds to the functionality to be assigned to a key. The allowed values are: feature, application , line or autodial. Please note that ## in the Avaya Aura® environment, it is recommended to configure the line keys on Avaya Aura® System Manager. ## [name], depending on the functionality entered in Type, can be either of the following: ## - the name of the feature that will be accessed by pressing the customized phone key, e.g. callpickup,groupcallpickup, etc. blf is supported in OpenSIP environment. ## PHONEKEY can be configured with blf and particular phone extension as attr1 allows BLF configuration in OpenSIP when BLF_LIST_URI is not specified. ## - the name of the application, e.g., lock, logout, etc. ## - the type of the phone line that will be accessed. The allowed values are: primary, bca (Bridged Call Appearance), sca (Shared Call Appearance, supported in OpenSIP environment only.). ## - autodialing of a defined phone extension - e.g. autodial. PHONEKEY is the only way to configure autodial in Open SIP environment. ## [attr1] and [attr2] are optional or mandatory attributes per the [Type] and [Name] values. For example, for bca attr1 is the bridged line number and attr2 is the extension. ## [label] is the key label that will be displayed on the Phone screen. ## [Forced] determines whether the user can move or delete a key. A few scenarios: ## - If Forced is set and location is empty, the key is applied. ## - If Forced is set for an occupied key location, it overrides the existing key or moves it to a different location. ## - If Forced is not set and the location is empty, the new definition for this location is applied. ## - If Forced is not set and the location is already occupied, the new definition is dropped. ## - If Forced is not set and the location is empty but this key type is already configured for a different location, the new definition is dropped. ## Multiple PHONEKEY lines can be specified, one for each key. All the PHONEKEY values and keywords are case-insensitive except values set in Label. ## The parameter is supported in OpenSIP and Aura environments. ## For full list of allowed Feature, Application, Line and Autodial values please refer to the phone's administration and installation guide. ## The default is "". ## This parameter is supported by: ## J100 SIP R4.0.1.0 and later (J169/J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later. ## SET PHONEKEY "Key=1;Type=application;Name=lock" ## SET PHONEKEY "Key=1;Type=autodial;Name=autodial;attr1=6837" ## SET PHONEKEY "Key=1;Type=line;Name=bca;attr1=1;attr2=6837" ## SET PHONEKEY "Key=1;Type=feature;Name=sac;attr1=6837" ## SET PHONEKEY "Key=2;Type=feature;Name=team;attr1=6836" ## SET PHONEKEY "key=6;Type=Feature;Name=BLF;Attr1=4004004;Label=Kichu4004" ## ## 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 IX Workplace 3.4 and later ## Avaya Vantage Connect 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R6.2 and later ## SET TEAM_BUTTON_RING_TYPE 3 ## ## SHOW_TEAM_BUTTON_VISUAL_ALERT specifies team buttons visual alert notification at monitoring station. ## Value Operation ## 0 Don't show visual alert ## 1 Show visual alert (default) ## On screen per team button toggle switch at monitoring station to control team button visual alert stays disabled / greyed out if value of this setting is 0. ## This parameter is supported by: ## Avaya IX Workplace 3.4.1 and later ## SET SHOW_TEAM_BUTTON_VISUAL_ALERT 0 ## ## SHOW_TEAM_BUTTON_CALLER_ID specifies whether to show or hide caller id on team button call pick up alert at monitoring station. ## Value Operation ## 0 hide caller id ## 1 show caller id (default) ## This parameter is supported by: ## Avaya IX Workplace 3.4.1 and later ## SET SHOW_TEAM_BUTTON_CALLER_ID 0 ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## 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 and J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later only for OpenSIP environment. ## Avaya Vantage Devices SIP R2.2.0.1 and later ## 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, J139 SIP R3.0.0.0 and later and J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later only for OpenSIP 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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), Value 0 is supported from R2.2.0.4 and later. ## 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, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.0.0 and later ## SET ENABLE_BLIND_TRANSFER 0 ## ## USE_CONTACT_IN_REFERTO defines which transfer target address should be used in Refer-To (header of REFER SIP request on attended transfer).  ## Value Operation ## 0 use TO URI of the transfer target in Refer-To header of REFER SIP request ## 1 use CONTACT URI of the transfer target in Refer-To header of REFER SIP request (default) ## This parameter is supported by: ## J100 SIP R4.0.1.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET USE_CONTACT_IN_REFERTO 0 ## ## PREF_MUTE_MIC_WHEN_JOINING_MEETING specifies whether microphone is muted when joining meetings. ## Value Operation ## 0 Do not mute the microphone when joining meeting ## 1 Mute the microphone when joining meeting (default) ## This parameter is supported by: ## Avaya IX Workplace 3.5.5 and later ## Avaya Vantage Connect Application SIP R2.2.0.3 and later ## SET PREF_MUTE_MIC_WHEN_JOINING_MEETING 0 ## ## PREF_BLOCK_CAMERA_WHEN_JOINING_MEETING specifies whether camera is blocked when joining meetings. ## Value Operation ## 0 Do not block the camera when joining meeting ## 1 block the camera when joining meeting (default) ## This parameter is supported by: ## Avaya IX Workplace 3.5.5 and later ## Avaya Vantage Connect Application SIP R2.2.0.3 and later ## SET PREF_BLOCK_CAMERA_WHEN_JOINING_MEETING 0 ## ## CALL_DECLINE_POLICY specifies if a Decline softkey is displayed for an incoming call and if enabled it defines which response code will be used to decline the call. ## The parameter is supported in all environments (Avaya Aura, Avaya IP Office and OpenSIP). ## Value Operation ## 0 Decline softkey is not displayed (default) ## 1 486 method is used ## 2 603 method is used ## This parameter is supported by: ## J100 SIP R4.0.4.0 and later (J139, J159, J169, J179 only), J189 SIP R4.0.6.1 and later ## SET CALL_DECLINE_POLICY 1 ## ## KEEP_CURRENT_CA specifies whether the selected line on the phone screen will remain selected if the line is a call appearance with a call that is just ended. ## The call can be on a primary call appearance, a bridged call appearance or a shared call appearance. ## Value Operation ## 0 Disable - select a higher priority call or reset to select the first line if the phone is idle ## 1 Enable - keep the current line selection (default) ## This parameter is supported by: ## J100 SIP R4.0.4.0 and later (J139, J159, J169, J179 only), J189 SIP R4.0.6.1 and later ## SET KEEP_CURRENT_CA 0 ## ## APPLICATION_HEADER_APPEARANCE_CONTEXT specifies how the Application Header Line on phone screen displays contextual information regarding various states of the lines. ## In J100 SIP R4.0.5.0 and later when a highlighting a Call Appearance, Shared Call Appearance or BLF key the Application Header line will display: ## ‘SHORT_FORM_USER_ID: DISPLAY_NAME’, when idle ## Note: If this parameter is enabled but both SHORT_FORM_USER_ID and DISPLAY_NAME are empty, then the userid of the registered SIP user will be displayed. ## ‘Calling: Called Number’, when placing an outbound call ## ‘Incoming: Caller Number’, when receiving an incoming call ## ‘Connect: Far End Number’, when an active call ## ‘On-Hold: Far End Number’, when a call is held ## Value Operation ## 0 Disable - no contextual information on the header line (default) ## 1 Enable with extra context - enables contextual information for various states, including name of the shared line key or BLF key, when applicable. ## 2 Enable - enables contextual information for various states, including name of BLF key, when applicable. ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later (values 0-1), J100 SIP R4.0.6.0 and later (values 0-2), J189 SIP R4.0.6.1 and later (Not supported by J129) ## SET APPLICATION_HEADER_APPEARANCE_CONTEXT 1 ## ## SHORT_FORM_USER_ID - Define this parameter if the users system extension number is different than the users ## FORCE_SIP_USERNAME and the user is monitoring other users via BLF. ## When the monitoring user attempts to place a call to a monitored BLF key it will prevent the J100 from displaying ## an incoming call for the BLF monitored user. ## SHORT_FORM_USER_ID is also displayed in the Application Header Line when APPLICATION_HEADER_APPEARANCE_CONTEXT is enabled ## The parameter is applicable for OpenSIP environment only. ## The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## SET SHORT_FORM_USER_ID "144" ## ## DISPLAY_NAME specifies the display name in the INVITE sent by J100. The display name will be presented in the remote party if the SIP controller supports a phone providing its own display name. ## The parameter is supported in OpenSIP environment only. The parameter is not applicable for AURA and IPO/IPO CCMS environments. ## In J100 SIP R4.0.5.0 and later, DISPLAY_NAME is also displayed as part of the Application Header Line when APPLICATION_HEADER_APPEARANCE_CONTEXT is enabled ## This parameter cannot contain forbidden symbols ";<>/&. The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## SET DISPLAY_NAME "Jane Doe" ## ############ DATA PRIVACY ############# ## ## ENABLE_GDPR_MODE specifies whether to enable / disable private user data protection on the phone. ## Value Operation ## 0 private user data protection is disabled (default) ## 1 private user data protection is enabled ## This parameter is supported by: ## J100 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 96X1 SIP R7.1.8.0 and later ## SET ENABLE_GDPR_MODE 1 ## ## RTCPGDPRCOM specifies whether to send the extension information in peer to peer RTCP packets or not. ## Value Operation ## 0 Peer to Peer RTCP packets include extension information (pre-R6.8.3 behavior) ## 1 Peer to Peer RTCP packets do not include extension information (default) ## Note: RTCP monitoring packets include the extension information (no matter what RTCPGDPRCOM value is). ## Note: For customers with data privacy concerns related to extension information sent in RTCP packets then the value 1 shall be used AND RTCP monitoring shall not be enabled. ## Note: Some customers use the extension information in peer to peer RTCP packets for call recording purposes. These customers shall set the value to 0. ## This parameter is supported by: ## J169/J179 H.323 R6.8.4 and later, J159 and J189 H.323 R6.8.5 and later ## 96X1 H.323 R6.8.4 and later ## B189 H.323 R6.8.4 and later ## SET RTCPGDPRCOM 0 ## ## ENABLE_PUBLISH_MAC_ADDRESS specifies whether to publish MAC address in the SIP signaling and in PPM messages (PPM is supported in Avaya Aura environment only). ## Value Operation ## 0 MAC address is not published in the SIP signaling and PPM messages (default) ## 1 MAC address is published in the SIP signaling and PPM messages ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.0.4.0 and later. When enabled, Ethernet MAC address is added to SIP Instant ID for all SIP messages (including in emergency call ## when the device is in unnamed registration (Aura environment only)). In addition, depends on active interface, Ethernet or Wi-Fi MAC address is published in new ## Avaya proprietary field (+av.mac-address) in the SIP REGISTER message's Contact header. The MAC address can be used by third party servers for finding location and reporting ## the location for emergency calls. For correct report of the Ethernet MAC address in SMGR, then ENABLE_PUBLISH_MAC_ADDRESS shall be set to 1. ## SET ENABLE_PUBLISH_MAC_ADDRESS 1 ## ############ 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R6.2 and later ## SET PROVIDE_KEY_REPEAT_DELAY 2 ## ## ENABLE_TALKBACK specifies whether Android TalkBack is enabled or disabled. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## Avaya Vantage SIP R2.1.0.0 and later ## SET ENABLE_TALKBACK 1 ## ################### 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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,,,,” ## ###################### LONG TERM ACOUSTIC EXPOSURE PROTECTION SETTINGS #################### ## ## ACOUSTIC_EXPOSURE_PROTECT_MODE_DEFAULT specifies the Long-term acoustic exposure protection mode default setting. ## Value Operation ## 1 Off (default) ## 2 Dynamic ## 3 4 hours ## 4 8 hours ## This parameter is supported by: ## J169/J179 H.323 R6.8.2 and later, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.8.2 and later ## J100 SIP R4.0.1.0 and later (J139, J169, J179 only), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later. ## SET ACOUSTIC_EXPOSURE_PROTECT_MODE_DEFAULT 2 ## ###################### 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later for OpenSIP environment only. ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Devices SIP R1.1.0.1 and later for IP Office (R1.1.0.1+) and OpenSIP (R2.0.1.0+) environments only (The emergency information is retrieved in Aura environment from PPM). ## 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later for OpenSIP environment only. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Devices SIP R1.1.0.1 and later for IP Office (R1.1.0.1+) and OpenSIP (R2.0.1.0+) environments. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.3 and later ## SET SYSAUDIOPATH 1 ## ## LOCKED_PREFERENCES specifies list of parameters configured in the Avaya IX Workplace 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 Vantage SIP R2.1.0.0 and later; used to lock certain menus in the settings application from end users configuration (administrators can always configure these fields). Possible values are: ## DHCP_SSON - for locking DHCP SSON option field. ## FILE_SERVER_URL - for locking file server field. ## GROUP - for locking GROUP field. ## DNSSRVR - for locking DNS server field. ## DOMAIN - for locking DNS Domain field. ## IPADD - for locking IP address information (including netmask and router information). ## LOCKED_PREFERENCES in this file is being overwritten by AADS configuration. ## Avaya IX Workplace 3.1.2 and later ## SET LOCKED_PREFERENCES "NAME_SORT_ORDER,NAME_DISPLAY_ORDER" ## ## OBSCURE_PREFERENCES specifies list of parameters configured in the Avaya IX Workplace 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 Vantage SIP R2.1.0.0 and later; used to hide certain menus in the settings application from end users (administrators can always view these fields). Possible values are: ## DHCP_SSON - for hiding DHCP SSON option field. ## FILE_SERVER_URL - for hiding file server field. ## GROUP - for hiding GROUP field. ## DNSSRVR - for hiding DNS server field. ## DOMAIN - for hiding DNS Domain field. ## IPADD - for hiding IP address information (including netmask and router information). ## SIPDOMAIN - for hiding SIP Domain field. ## SIP_CONTROLLER_LIST - for hiding SIP controller list. ## L2Q - for hiding tagging information. ## L2QVLAN - for hiding VLAN information. ## SNTPSRVR - for hiding SNTP server information. ## OBSCURE_PREFERENCES in this file is being overwritten by AADS configuration. ## Avaya IX Workplace 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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 Slide switch is High ## This parameter is supported by: ## J189 H.323 R6.8.5 and later ## J159 SIP 4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## J189 H.323 R6.8.5 and later ## 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) ## 2 USB port is disabled (Android OS and Recovery). ## This parameter is supported by: ## Avaya Vantage Devices SIP R1.0.0.0 and later (values 0-1); Value 0 will disable USB on Android OS only (USB can be used with BRM), Value 2 will disable USB on Android OS and BRM. Value 2 is supported in R2.2.0.1 and later. ## SET ENABLE_USB_GENERAL_PURPOSE 0 ## ## ENABLE_USBDEVICE specifies whether support of USB devices on the phone is enabled or disabled. ## Value Operation ## 0 USB device is disabled. ## 1 USB device is enabled (default) ## This parameter is supported by: ## J100 SIP R4.0.9.0 and later (supported by J159, J189 only) ## SET ENABLE_USBDEVICE 0 ## ## ENABLE_USBHEADSET specifies whether USB Headset is enabled or disabled. ## Value Operation ## 0 USB headset is disabled. ## 1 USB headset is enabled (default) ## This parameter is supported by: ## J100 SIP R4.0.7.1 and later ## SET ENABLE_USBHEADSET 0 ## ## ENABLE_USBSTICK specifies whether USB flash drive is enabled or disabled. ## Value Operation ## 0 USB flash drive is disabled. ## 1 USB flash drive is enabled (default) ## This parameter is supported by: ## J100 SIP R4.0.9.0 and later (supported by J159, J189 only) ## SET ENABLE_USBSTICK 0 ## ## ENABLE_USBKEYBOARD specifies whether USB keyboard is enabled or disabled. ## Value Operation ## 0 USB keyboard is disabled. ## 1 USB keyboard is enabled (default) ## This parameter is supported by: ## J100 SIP R4.0.9.0 and later (supported by J159, J189 only) ## SET ENABLE_USBKEYBOARD 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: ## J179 SIP R2.0.0.0 and later, J189 SIP R4.0.6.1 and later; J159 SIP R4.0.8.0 and later ## 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 ## ########### NFC SETTINGS ############## ## ## NFCSTAT specifies whether the user is given an option to enable NFC. ## Value Operation ## 0 NFC is disabled and the user is not given an option to enable it ## 1 The user is given an option to enable NFC (default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later; supported by Avaya Vantage Devices with built-in Wi-Fi/Bluetooth/NFC support only. ## SET NFCSTAT 0 ## ########### 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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: *.-!$%&'()+,:;/\=@~# ## The space character, ASCII 0x20, is NOT supported. ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WLAN_PASSWORD Avaya123 ## ## WLAN_WPA2E_EAP_METHOD specifies the pre-configured Wi-Fi network 802.1x EAP method. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e. ## Value Operation ## PEAP Connect using PEAP (Default) ## TLS Connect using TLS ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WLAN_WPA2E_EAP_METHOD TLS ## ## WLAN_WPA2E_EAP_PHASE2 is the pre-configured Wi-Fi network 802.1x phase 2 Method. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e and WLAN_WPA2E_EAP_METHOD is PEAP. ## Value Operation ## none No phase 2 authentication (Default, but not currently supported) ## MSCHAPV2 As of J100 2.0, MUST be set to this value for forward compatibility ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later; only J129 and J179 support a pluggable Wi-Fi/BT module (J139/J169 does not support Wi-Fi/BT), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WLAN_WPA2E_EAP_PHASE2 MSCHAPV2 ## ## WLAN_WPA2E_IDENTITY specifies the pre-configured Wi-Fi network 802.1x identity. ## This parameter is only applicable when WIFISTAT enables Wi-Fi and WLAN_SECURITY is wpa2e and: ## - WLAN_WPA2E_EAP_METHOD is PEAP and WLAN_WPA2E_EAP_PHASE2 is MSCHAPV2 ## or ## - WLAN_WPA2E_EAP_METHOD is TLS ## The value can contain 1 to 32 characters; the default value is null (""). ## Valid characters are: ## A-Z, a-z, 0-9, and the following: *.-!$%&'()+,:;/\=@~# ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WLAN_WPA2E_ANONYMOUS_IDENTITY foo@example ## ########## WLAN LAYER 2 QOS SETTINGS ######## ## ## WLAN_L2QAUD 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WLAN_L2QAUD 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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), J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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), J189 SIP R4.0.6.1 and later ## 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), J189 SIP R4.0.6.1 and later ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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. ## 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), J189 SIP R4.0.6.1 and later, ## Up to R4.0.7.0 (excluded) - valid values are 1 through 24, R4.0.7.0 and later - valid values are 0 through 24. ## '0' means the subscription would never expire. This value shall be used when Service Observing (SO) feature is supported via CTI. ## 96x1 SIP R6.2 and later; Valid values are 1 through 24 ## SET CC_INFO_TIMER 8 ## ########## CALL CENTER SETTINGS (96x1/J100 H.323 and J100 SIP) ########## ## ## 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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 (H.323 environment only). ## Value Operation ## 0 Agent greetings will be disabled ## 1 Agent greetings will be enabled (default) ## This parameter is supported by: ## J100 SIP R4.0.7.0 and later (J169, J179, J189 only), Aura environment only. ## J169/J179 H.323 R6.7 and later, J189 H.323 R6.8.5 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 (default) ## 1 Agent greetings will be deleted when the agent logs out ## This parameter is supported by: ## J100 SIP R4.0.7.0 and later (J169, J179 and J189 phones only), Aura environment only. ## J169/J179 H.323 R6.7 and later, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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: ## J100 SIP R4.0.7.0 and later (J169, J179, J189), Aura environment only. ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 Connect 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, J189 H.323 R6.8.5 and later ## Avaya Vantage Connect 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 Connect Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J189 H.323 R6.8.5 and later ## Avaya Vantage Connect 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, J189 H.323 R6.8.5 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 Connect Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J189 H.323 R6.8.5 and later ## Avaya Vantage Connect 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 Connect Application SIP) ## This parameter is supported by: ## Avaya Vantage Connect 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 Connect Application SIP) ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J189 H.323 R6.8.5 and later ## Avaya Vantage Connect 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 ## ## IGNORESTAT specifies whether the Ignore button is enabled or disabled when CCBTNSTAT is 0. ## While the other parameters (HOLDSTAT, CONFSTAT, etc.) are associated with call center, ## IGNORESTAT can be used in call-center and non-call center environments. ## Value Operation ## 0 The Ignore button is disabled when CCBTNSTAT is 0 ## 1 The Ignore button is enabled (default) ## This parameter is supported by: ## J169/J179 H.323 R6.8.3 and later, J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.8.3 and later ## SET IGNORESTAT 1 ## ################## APPLICATION CUSTOMIZATION SETTINGS ################## ## ## ENABLE_JOIN_EQUINOX_MEETING specifies whether to present the "Join Meeting" icon in the dialer tab. ## Value Operation ## 0 Hide "Join Meeting" icon in the dialer tab ## 1 Present "Join Meeting" icon in the dialer tab (default) ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET ENABLE_JOIN_EQUINOX_MEETING 0 ## ## SETTINGS_MENUS_ACCESS specifies whether user and administrator can access settings menus or administrator only (using administrator password). ## Value Operation ## 0 Both users and administrator can access the settings menus without entering administrator password (default) ## 1 Only administrator can access the settings menus by entering the administrator password ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET SETTINGS_MENUS_ACCESS 1 ## ## COMPANION_APPLICATION specifies the Android package name of the companion application to be used with Avaya Vantage Connect application running on the same Avaya Vantage device. ## The default is "". When default, then Branding Logo on the top left corner of the Avaya Vantage Connect application is displayed and pressing on it will have no effect. ## When COMPANION_APPLICATION is configured with the Android package name of the companion application installed on the device, then in addition to the Branding Logo there will be ## an icon that once pressed will invoke the companion application. The default companion application icon can be replaced using COMPANION_APPLICATION_BRANDING_FILE. ## When Avaya Connect Expansion Module application is installed on the same device where Avaya Vantage Connect is running and it is paired/connected with Avaya Vantage Connect and ## COMPANION_APPLICATION is "", then in addition to the Branding Logo there will be an icon that once pressed will invoke the Avaya Connect Expansion Module application. The default ## companion application icon can be replaced using COMPANION_APPLICATION_BRANDING_FILE. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET COMPANION_APPLICATION "com.arowana.houdini_tab_portrait" ## ## COMPANION_APPLICATION_BRANDING_FILE specifies the URL from which the companion application icon will be downloaded from. The companion application icon is presented near ## the Branding Logo on the top left corner of the Avaya Vantage Connect application. Up to one URL shall be specified. The default value ## is "" (in such case the default icons will be presented). URL shall be absolute path (start with http:// or https://). IPv4 address, IPv6 address or FQDN is supported. ## 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 Connect Application SIP R2.2.0.0 and later ## SET COMPANION_APPLICATION_BRANDING_FILE "http://example.com/IT/companion_application_branding_file_example.png" ## ################## EXPANSION MODULE APPLICATION SETTINGS ################## ## ## BUTTON_MODULE_ENABLE specifies whether to enable/disable Expansion Module or allow end users to enable/disable it. ## Value Operation ## 0 The Expansion Module support by Avaya Vantage Connect is disabled (default). All Avaya Vantage Connect Expansion Module settings are hidden. ## No option to enable Expansion module in the Avaya Vantage Connect settings menus. No option to pair/connect Avaya Connect Expansion Module application ## with Avaya Vantage Connect. ## 1 Expansion Module is enabled. No option to disable Expansion Module using Avaya Vantage Connect Expansion Module settings. ## Avaya Connect Expansion Module application shall be able to discover Avaya Vantage Connect, pair and connect with Avaya Vantage Connect application. ## 2 The Expansion module is disabled on Avaya Vantage Connect by default. However, users can enable/disable Expansion Module using settings menus. ## Once enabled, Avaya Connect Expansion Module application shall be able to discover Avaya Vantage Connect, pair and connect with Avaya Vantage Connect application. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later; R3.0.0.3 - when BUTTON_MODULE_ENABLE is set to "1" and ACTIVE_CSDK_BASED_PHONE_APP is "com.avaya.android.vantage.basic", then ## there will be automatic pairing of the Expansion application to Avaya Vantage UC Experience. ## Avaya Vantage R3.0.0.3 and later - When BUTTON_MODULE_ENABLE is set to "0", the Expansion Module application icon will be hidden. When BUTTON_MODULE_ENABLE is set to "1" or "2", ## the Expansion Module application icon will appear. ## SET BUTTON_MODULE_ENABLE 0 ## ## BUTTON_MODULE_CONNECTION_PORT specifies the TCP port on which the Avaya Vantage Connect will listen to incoming TCP connections from Avaya Connect Expansion Module applications. ## Valid values 1024-65535. Default port is 1389. ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R2.2.0.0 and later ## SET BUTTON_MODULE_CONNECTION_PORT 10000 ## ################## AMAZON ALEXA SETTINGS ################## ## ## ENABLE_ALEXA specifies whether to enable Amazon Alexa application. ## Value Operation ## 0 Disabled (default) ## 1 Enabled ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later. ## SET ENABLE_ALEXA 1 ## ################## TRANSDUCER SETTINGS ################## ## ## AUDIO_DEVICE_CALL_CONTROL_ENABLED specifies whether headset call control keys are enabled for Answer/End/Mute operations. Answer/End/Mute is supported with USB headsets. ## Answer/End is supported with BT headsets. ## Value Operation ## 0 Disabled ## 1 Enabled (default). ## This parameter is supported by: ## Avaya IX Workplace 3.7.1 and later; the default is changed to 1 in 3.7.4 and on (until then it is 0) ## Note: Avaya Vantage Connect Application SIP R2.2.0.0 and later supports by default headset call control keys. No option to disable them. ## SET AUDIO_DEVICE_CALL_CONTROL_ENABLED 0 ## ################## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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.). ## R2.2.0.0 and later supports IPv6 address as well. ## 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 ## SET TRUSTCERTS http://[e9e4:35a::cef2]/dir_example/trust.pem ## 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 Connect Application and Avaya IX Workplace 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya IX Workplace 3.1.2 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Not used by Avaya Vantage Open application. ## 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 ## ## ENABLE_RFC5922 specifies whether SIP domain will be verified per RFC 5922 as part of certificate hostname validation. ## ENABLE_RFC5922 is only applicable if TLSSRVRID value is "1". ## The parameter is supported in ALL environments (Avaya Aura, Avaya IP Office, OpenSIP). ## Value Operation ## 0 SIP domain verification per RFC 5922 is not enabled. ## 1 SIP domain verification per RFC 5922 is enabled (default). The configured SIP Domain must be found in the Subject Alternative Name (SAN) field as a SIP URI according to RFC 5922. ## If there are no SAN entries of any kind, a match of the SIP Domain in the CN is permitted. ## This parameter is supported by: ## J100 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## SET ENABLE_RFC5922 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, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET BLOCK_CERTIFICATE_WILDCARDS 1 ## ## KEYUSAGE_REQUIRED specifies whether to check the server certificate for the presence of a Key Usage extension. ## Value Operation ## 0 Do not check Key Usage extension (default) ## 1 Check Key Usage extension in the server certificate. If missing then the server certificate will be rejected. ## This parameter is supported by: ## J100 SIP R4.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.9.0 and later ## SET KEYUSAGE_REQUIRED 1 ## ## CERTIFICATE_MIN_RSA_KEY_LENGTH specifies the minimum RSA key length to be used for validating the certificate received from the server during TLS Handshake. ## Server certificate will be rejected in case the configured value is greater than server certificate's key length. ## When FIPS_ENABLED is 1, then the parameter is ignored and 2048 bits is the minimum RSA length. ## Value Operation ## 0 1024 bits (default) ## 1 2048 bits ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.1 and later; the default is 1 (2048 bits). ## Avaya Vantage built-in Unified Communication Experience R3.0.0.1 and later ; the default is 1 (2048 bits). ## Avaya IX Workplace R3.8 and later ## SET CERTIFICATE_MIN_RSA_KEY_LENGTH 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, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Connect Application SIP R2.1.0.0 and later, R2.2.0.3 the default was changed to 1. ## Avaya IX Workplace 3.5.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Not used by Avaya Vantage Open application. R2.2.0.3 the default was changed to 1. ## 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 ## ## CIPHER_SUITE_BLACKLIST specifies a list of a blacklisted ciphers, which will not be included during TLS connection negotiation. ## The default value is "". ## This parameter is supported by: ## Avaya IX Workplace 3.7 and later ## SET CIPHER_SUITE_BLACKLIST "TLS_RSA_WITH_AES_128_GCM_SHA256,TLS_ECDH_ECDSA_WITH_AES_128_GCM_SHA256" ## ################ 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; HTTPPROXY is NOT supported for SCEP, but for Android HTTP based applications. ## R2.2.0.0 and later supports IPv6 address as well. ## 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 ## SET HTTPPROXY [e9e4:35a:cef2::1]:8000 ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; HTTPPROXY is NOT supported for SCEP, but for Android HTTP based applications. In R3.0.0.0 and later, there is support of wildcards in the domain names. ## 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; R2.2.0.0 and later supports IPv6 address as well. ## SET HTTPPROXYAUTOCONFIGURL http://example.net/example.com/inet ## SET HTTPPROXYAUTOCONFIGURL example.com/inet ## SET HTTPPROXYAUTOCONFIGURL https://[e9e4:35a:cef2::1]/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 IX Workplace 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 IX Workplace 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"; R2.2.0.0 and later supports IPv6 address as well. ## H1xx SIP R1.0.1 and later ## SET CAPTIVE_PORTAL_SERVER http://clients3.google.com/generate_204 ## SET CAPTIVE_PORTAL_SERVER http://[e9e4:35a:cef2::1]:80/examplepath/generate_204 ## ###################### STUN SETTINGS ##################### ## ## The parameters below are applicable only when 3PCC_SERVER_MODE=0 or 3PCC_SERVER_MODE=2 and ENABLE_3PCC_ENVIRONMENT=1 ## BroadSoft mode (3PCC_SERVER_MODE=1) is not supported. ## ## STUN_SERVER_ADDRESS specifies STUN Server address, in IPv4, IPv6 or FQDN format. The default is "". ## The remaining NAT/STUN-related parameters in this section will be used by the phone ONLY if ## STUN_SERVER_ADDRESS is set, and will be ignored otherwise. ## If specified as an IPv4 address, the STUN server port will default to 3478. ## Valid characters are: ## 0-9, a-z, A-Z, .: ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET STUN_SERVER_ADDRESS stun.aura.avaya.com ## ## STUN_UDP_INITIAL_TIMEOUT_MSEC specifies initial timeout, in milliseconds, to wait for a Response to a STUN Request sent over UDP. ## The timeout value is internally doubled after each retransmission. ## Valid values are 500 (1/2 sec) through 3000 (3 sec); the default value is 500. ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET STUN_UDP_INITIAL_TIMEOUT_MSEC 1000 ## ## STUN_UDP_MAX_TRANSMISSIONS specifies the number of times the phone will transmit a STUN Request until a Response is received, after ## which the Request will be treated as failed. ## Valid values are 1 through 7; the default value is 7. ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET STUN_UDP_MAX_TRANSMISSIONS 3 ## ## STUN_UDP_MAX_MEDIA_TRANSMISSIONS specifies the number of times the phone will transmit a STUN Request to get NAT bindings for the phone’s RTP/RTCP IP address and ports. ## Retransmissions continue until a response is received, or until a total number of requests have been sent.  ## Initial timeout, in milliseconds, to wait for a Response to a STUN Request sent over UDP for  media is 500 msec. The timeout value is internally doubled after each [re]transmission. ## Valid values are 1 through 4; the default value is 3. ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET STUN_UDP_MAX_MEDIA_TRANSMISSIONS 3 ## ## NAT_SIGNALING_KEEPALIVE_ENABLED Specifies whether or not the telephone sends keep-alives to refresh NAT bindings for the phone’s private signaling IP address and port. ## Value Operation ## 0 Keep-alive messages are not sent ## 1 Keep-alive messages are sent (default) ## This parameter is supported by: ## J100 SIP R4.0.2.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET NAT_SIGNALING_KEEPALIVE_ENABLED 0 ## ## NAT_SIGNALING_KEEPALIVE_OVERRIDE_SEC specifies interval, in seconds, between keep-alives used to refresh NAT bindings for the phone’s private signaling IP address and port. ## Valid values are 15 through 900 (15 minutes); the default value is 29. ## Value Operation ## 15-900 The phone will use this value as the keep-alive interval for every SIP registration and dialog ## This parameter is supported by: ## J100 SIP R4.0.0.2 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET NAT_SIGNALING_KEEPALIVE_OVERRIDE_SEC 20 ## ###################### SCEP SETTINGS ##################### ## ## Note for all endpoints with exception of J100 SIP R4.0.1.0/96x1 SIP 7.1.5.0 and later: ## When FIPS_ENABLED is set to 1 (for endpoints which support FIPS mode), SCEP shall not be used. ## Note for J100 SIP R4.0.1.0/96x1 SIP 7.1.5.0 and later: ## When FIPS_ENABLED is set to 1 (for endpoints which support FIPS mode), SCEP is supported. An identity certificate ## can be generated using SCEP when FIPS_ENABLED is set to 1. SCEPENCALG must be set to 1. ## ## General note: If identity certificate was generated before FIPS_ENABLED is set to 1, the phone will keep using it. ## It is NOT recommended to use identity certificate generated using SCEP when FIPS_ENABLED is 0 and then ## configure the phone 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). Only J100 SIP R4.0.1.0/96x1 SIP 7.1.5.0 and later supports SCEP ## when FIPS_ENABLED is set to 1. Other endpoints will need to download a PKCS12 file from a CA configured to work in FIPS mode. ## J100 SIP R4.0.1.0/96x1 SIP 7.1.5.0 and later also support download of a PKCS12 file. ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later - the URL can be https or http. ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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. ## R2.2.0.0 and later supports IPv6 address as well. ## 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 ## SET MYCERTURL http://[e9e4:35a::cef2]:80/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 "$SERIALNO". ## This parameter is supported by: ## J129 SIP R1.0.0.0 (or R1.1.0.0), J169/J179 SIP R1.5.0, J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later ## Avaya Vantage Devices SIP R1.0.0.0 and later; Avaya Vantage Open application does not use identity certificate. ## 96x1 H.323 R6.0 and later ## B189 H.323 R6.6 and later ## 96x1 SIP R6.0 and later ## H1xx SIP R1.0 and later ## 96x0 H.323 R3.1 and later ## 96x0 SIP R1.0 and later ## Note that prior to R2.6.8, 96x0 SIP releases only support ## "$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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; only "2048" is supported ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 and J189 H.323 R6.8.5 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, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R2.2.0.0 and later (When FIPS_ENABLED==1, then AES-256 is used) ## J100 SIP R4.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.4.0 and later ## SET SCEPENCALG 1 ## ## SCEPBEFOREUPGRADE specifies in case where both SCEP and software upgrade shall be done, whether SCEP will be done before software upgrade or afterwards. ## Value Operation ## 0 SCEP will be done after software upgrade (default) ## 1 SCEP will be done before software upgrade ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.6.8.1 and later ## B189 H.323 R6.8.1 and later ## SET SCEPBEFOREUPGRADE 1 ## ## SCEP_ENTITY_CLASS is used with Avaya Aura SMGR only (for other environments this parameter shall be ""). The parameter is used ## to identify the entity class for which identity certificates will be generated using SCEP. The value of "entity-class" is configured in Avaya Aura SMGR. ## The default value is "". ## J100 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 96X1 SIP R7.1.8.0 and later ## SET SCEP_ENTITY_CLASS "$123" ## ###################### PKCS12 SETTINGS ##################### ## ## PKCS12URL specifies the URL to be used to download a PKCS #12 file ## containing an identity certificate and its private key. ## 0 to 255 ASCII characters, zero or one URL. The value can be a string that contains either ## "$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 $MACADDR is added to the URL then the PKCS12 filename on the file server shall include MAC address ## without colons (i.e., 6 pairs of ASCII hexadecimal characters AABBCCDDEEFF with hex characters A-F ## encoded as upper-case characters). For example, if Ethernet MAC address of a specific phone ## is: 00-24-D7-E4-2E-98 and the PKCS12URL is: pkc12file_$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. ## SIP endpoints note: ## 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. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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. ## Avaya Vantage Devices SIP R1.0.0.0 (build 2304) and later; Avaya Vantage Open application does not use identity certificate. ## R2.2.0.0 and later supports IPv6 address as well. ## J129 SIP R1.0.0.0 and later ## 96x1 H.323 R6.6 and later ## SET PKCS12URL pkc12file_$MACADDR.cer ## SET PKCS12URL http://example.com/exampledir/pkc12file_$MACADDR.cer ## SET PKCS12URL https://example.com/exampledir/pkc12file_$SERIALNO.cer ## SET PKCS12URL http://[e9e4:35a:cef2::1]:80/path/example.p12 ## ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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) ## 16xx H.323 R1.1 and later ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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 ## 16xx H.323 R1.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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 - phone continues DHCP operation only disregarding 802.1x authentication status (default) ## 1 Waits for 802.1X to complete ## 2 Does not wait for successful 802.1X - phone continues boot disregarding 802.1x authentication status ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later (value 2 is supported in R6.8.1 and later). ## 96x1 H.323 R6.3 and later (value 2 is supported in R6.8.1 and later). ## B189 H.323 R1.0 and later (value 2 is supported in R6.8.1 and later). ## 96x0 H.323 R3.2.2 and later ## SET DOT1XWAIT 1 ## ## DOT1XEAPTLSONLYWITHCERT specifies 802.1x EAP-TLS is activated when there is identity certificate installed. ## Value Operation ## 0 802.1x EAP-TLS is activated immediately when DOT1XSTAT is set to 1 or 2 and DOT1XEAPS is "TLS". This is the default behavior in pre R6.8.1. ## 1 802.1x EAP-TLS is activated when DOT1XSTAT is set to 1 or 2 and DOT1XEAPS is "TLS" and only if identity certificate is installed and at least one trusted certificate is installed (default). ## This parameter is supported by: ## J169/J179 H.323 R6.8.1 and later, J159 and J189 H.323 R6.8.5 and later ## 96x1 H.323 R6.6.8.1 and later ## B189 H.323 R6.8.1 and later ## SET DOT1XEAPTLSONLYWITHCERT 0 ## ################ 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: ## Avaya Vantage SIP R2.2.0.0 and later (OpenSSL FIPS Object Module - CERT #2398 is being used. Please note that Android BoringSSL ## does not have an embedded FIPS 140-2-validated cryptographic module). Features which uses OpenSSL FIPS Object Module are: Configuration and Software files download from HTTPS file server, ## AADS, SSH, PPM, SCEP, Debug report generation. ## Avaya Vantage Connect Application SIP R2.2.0.0 and later (OpenSSL FIPS Object Module - CERT #2398 is being used. Please note that Avaya Vantage Connect Application ## uses Android Java Cryptographic APIs which are using Android BoringSSL which does not have an embedded FIPS 140-2-validated cryptographic module). ## Features which uses OpenSSL FIPS Object Module are: SIP. ## J169/J179 SIP R1.5.0; J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 and later; OCSP is supported for certain PLATFORM features only: Configuration and software files download from HTTPS file server, PPM, AADS, ## SCEP over HTTPS and DES. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Devices SIP R3.0.0.0 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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. ## J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later and J139 SIP R4.0.4.0 and later support 5-digit values (default is 00000) as J169/J179 SIP R4.0.1.0 ## J100 SIP R4.0.1.0 and later support 5-digit values (default is 00000), as in R3.0.0.0 and later, but also support by J169/J179 only display (WML browser) Push requests - Both "22222" and "00222" values have the same meaning. ## 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, J159 and J189 H.323 R6.8.5 and later; Support 3 and 4-digit values (default is 2222). J159 does NOT support display of WML browser Push requests. ## 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 listening 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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 ## ## PUSH_MODE specifies which combination of non-secure and secure push is to be used. ## Value Operation ## 0 Only non-secure Push (http) is enabled ## 1 Only secure Push (https) is enabled ## 2 Both secure and non-secure Push are enabled (Default) ## NOTE: If there is no identity certificate installed on the phone with the serverAuth attribute, then only http can be used. This means that if PUSH_MODE = 1, Push will be disabled and if PUSH_MODE=2 then only non-secure will be used. ## In this last case, the Subscribe Push must indicate "nonsecure". ## This parameter is supported by: ## J100 SIP R4.0.8.0 and later ## 96x1 SIP R7.1.12.0 and later ## SET PUSH_MODE 1 ## ## PUSHPORT_SECURE specifies the port to be used to listen for incoming secure Push requests over https if secure push is enabled. ## Valid values are 80 through 65535; the default value is 8443. ## NOTE: The default value is set to 8443 to avoid a conflict with the web server in the phone using the default value. ## This parameter is supported by: ## J100 SIP R4.0.8.0 and later ## 96x1 SIP R7.1.12.0 and later ## SET PUSHPORT_SECURE 411 ## ################ HTTP/S WEB SERVER ################# ## ## ENABLE_WEBSERVER specifies whether the HTTP/S WEB Server is enabled or disabled. ## Value Operation ## 0 Disabled ## 1 Enabled ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later, J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; ## The default of ENABLE_WEBSERVER of OpenSIP J100 models is 1. This is to enable web server by default in OpenSIP environments. ## The default of ENABLE_WEBSERVER of Aura/IPO J100 models is 0. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET WEBSERVER_ON_HTTP 0 ## ## 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, J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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. ## The default is "27238". ## Please note that the configured password shall meet the below conditions else the default password is used: ## 1. Range is 8 - 31 characters. ## 2. There must be at least one number, one letter and one symbol (allowed symbols are ! @ # $ % ^ & * ( ) - + : ; " ' , | { } [ ] ## < > / ? \ _ . ~ ` =). ## 3. Maximum number of consecutive characters from the same character class(number, upper case letter, lower case letter, symbol) is 4. ## 4. Maximum number of consecutive identical characters is 2. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## SET FORCE_WEB_ADMIN_PASSWORD HelloWorld!01 ## ################# 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, J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0 ## 96x1 H.323 R6.0 and later. ## 96x1 SIP R6.2 and later ## SET WMLHELPSTAT 0 ## ## ENABLE_WMLPUSH_ALERTING specifies the behavior of WML browser during incoming call. ## Value Operation ## 0 WML bowser disappears when the phone starts ringing and an incoming call appears instead (Default) ## 1 WML browser still appears when the phone starts ringing and user can answer a call by off-hook from WML browser application. ## This parameter is supported by: ## J169/J179 SIP R4.0.6.0 and later, J189 SIP R4.0.6.1 and later ## SET ENABLE_WMLPUSH_ALERTING 1 ## ################# 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 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. ## R2.0.1 and later - range is 0-999 with default 60 for all Avaya Vantage devices including K155. ## R2.2.0.4 and later - the parameter is configured by administrators only in settings file or AADS only (and not configured by end users ## using Android settings application --> Display menu --> Sleep field). ## 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), J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later (J129 default is 10, other phones 0). ## 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, J159 and J189 H.323 R6.8.5 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 (""). ## This parameter is supported by: ## J169/J179 H.323 R6.7 and later, J159 and J189 H.323 R6.8.5 and later. J159 main screen resolution is 320x240 pixels and color depth of 18 bits. ## J189 J159 main screen resolution is 800x480 pixels and color depth of 18 bits. ## 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 ## ## SCREENSAVER_QD 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 (""). ## Screen resolution is 240x320 pixels and color depth of 18 bits. ## Avaya screen saver will be presented if this parameter is "" or incorrect/invalid. ## This parameter is supported by: ## J159 and J189 H.323 R6.8.5 and later ## SET SCREENSAVER_QD 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: ## Avaya Vantage Devices SIP R2.0.4.0 and later; Avaya Vantage screensaver image resolution shall be 1280x720 with 24 bits color depth for K155 and 800x1280 with 24 bits color depth for K165/K175. ## The following file types are supported by Avaya Vantage: PNG, JPG (JPEG), GIF and BMP (GIF is presented without animation). Up to 16 screensaver images can be downloaded. ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## 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. ## J159/J169/J179 screen resolution is 320 pixels x 240 pixels. J159/J179 color depth is 16 bits. ## J189 screen resolution is 800 x 480 pixels with color depth of 16 bits. ## 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: ## Avaya Vantage Devices SIP R2.0.4.0 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later; To select one of the six built-in default images, ## specify a value from 0 to 5 for this setting where 0 corresponds to "Default image 1" and 5 corresponds to "Default image 6". ## 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: ## Avaya Vantage Devices SIP R2.0.4.0 and later ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J159 SIP R4.0.3.0 and later, J139 SIP R4.0.4.0 and later, J189 SIP R4.0.6.1 and later ## SET SCREENSAVER_IMAGE_SELECTABLE 0 ## ## SCREENSAVER_IMAGE_SECONDARY specifies a list of screensaver images for the secondary display. The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.3.0 and later (J159 only), J189 SIP R4.0.6.1 and later ## 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. ## J159 / J189 secondary screen resolution is 240 pixels x 320 pixels with color depth of 16 bits. ## The files shall be stored in the same directory defined by HTTPDIR / TLSDIR. ## SET SCREENSAVER_IMAGE_SECONDARY "secondary_screensaver_example1.jpg,secondary_screensaver_example2.jpeg" ## ## SCREENSAVER_IMAGE_DISPLAY_SECONDARY specifies the administrator choice of secondary screensaver image. ## The filename shall be one of the filenames listed in SCREENSAVER_IMAGE_SECONDARY. ## If SCREENSAVER_IMAGE_SELECTABLE_SECONDARY is set to 1 then the end user may override this setting. ## The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.3.0 and later (J159 only), J189 SIP R4.0.6.1 and later ## SET SCREENSAVER_IMAGE_DISPLAY_SECONDARY secondary_screensaver_example1.jpg ## ## SCREENSAVER_IMAGE_SELECTABLE_SECONDARY specifies whether end users are allowed to choose secondary screensaver images ## (and overrides administrator choice as configured using SCREENSAVER_IMAGE_DISPLAY_SECONDARY parameter). ## Value Operation ## 0 End user is not allowed to choose secondary screensaver image and will not see the secondary screensaver image selection in the Settings -> Display menu. ## 1 End user is allowed to choose the secondary screensaver image from the Settings -> Display menu (Default) ## This parameter is supported by: ## J100 SIP R2.0.0.0 and later (J159 only), J189 SIP R4.0.6.1 and later ## SET SCREENSAVER_IMAGE_SELECTABLE_SECONDARY 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 (Default) ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.2.0.4 (user configuration is in Android settings application --> "Display" menu --> "Sleep" field) ## J100 SIP R2.0.0.0 and later (J169 and J179 only), J139 SIP R3.0.0.0 and later, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later (user configuration is in "User Menu->Settings->Display" submenu) ## 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, J159 and J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J159 and J189 H.323 R6.8.5 and later ## J169/J179 SIP R1.5.0; J100 SIP R4.0.1.0 and later (J169/J179 only), J189 SIP R4.0.6.1 and later ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later; Please note that when ENABLE_PHONE_LOCK is "1", then ## J100 SIP phones present the "Lock" application (Pressing Menu button --> Applications). Users can use "Phone Keys Customization" (Pressing Menu button --> Settings --> Phone) to present the "Lock" application in the main phone screen. ## There is no Lock softkey or feature button. ## When ENABLE_PHONE_LOCK is "0", then there is no "Lock" application and there is no option to present "Lock" as part of "Phone Keys Customization" in the main phone screen. ## Avaya Vantage Devices SIP R1.0.0.0 and later; Please note that ENABLE_PHONE_LOCK is used as enable/disable ## of lock screen when Avaya Breeze Client SDK application defined in ACTIVE_CSDK_BASED_PHONE_APP is installed. ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 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: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later, range is 0-20 with default value 0. ## Avaya Vantage Devices SIP R1.0.0.0 and later; ## H1xx SIP R1.0 and later; ## SET PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS 10 ## ## PHONE_LOCK_PASSWORD_LOCKED_TIME specifies the lock time in minutes as result of reaching the number of attempts specified in PHONE_LOCK_PASSWORD_FAILED_ATTEMPTS. ## The range is 5-1440. The default value is 5. ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## SET PHONE_LOCK_PASSWORD_LOCKED_TIME 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/ ## ## PHONE_LOCK_PIN specifies PIN the user have to enter to unlock their phone. SIP password isn't accepted if PIN is specified. ## Only digits are supported. The length can only be 4-20 digits long. Invalid values are ignored. The default value is "". ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## SET PHONE_LOCK_PIN 1694 ## ############# 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later (in all environments - Aura, IP Office and OpenSIP) ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later (in all environments - Aura, IP Office and OpenSIP) ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later (in all environments - Aura, IP Office and OpenSIP) ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later (in all environments - Aura, IP Office and OpenSIP) ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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: ## Avaya Vantage Connect Application SIP R2.0.1.0 and later (in all environments - Aura, IP Office and OpenSIP) ## 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## 4 Enabled SUPER_WIDEBAND ## This parameter is supported by: ## Avaya Vantage Connect Application SIP R1.0.0.1 and later; supported in both Aura and IPO environments. R2.0.1.0 and later in all environments - Aura, IP Office and OpenSIP, ## values 0-3 are supported. ## 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, J159 SIP R4.0.3.0 and later; J189 SIP R4.0.7.1, values 0-3 are supported by all ## phone models, value 4 is supported by J189 only. ## Up to R2.0.0.0 (excluded) for IP office and OpenSIP 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 IX Workplace 3.1.2 and later; values 0-3 are supported. ## 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: ## Avaya Vantage Connect Application SIP R2.1.0.0 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, J159 SIP R4.0.3.0 and later ## Avaya IX Workplace 3.1.2 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 and later ## Avaya IX Workplace 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## ## CODEC_PRIORITY specifies the order of codecs in the SDP offer. The default value is "". ## Supported values: OPUS or G722 or G711U or G711A or G726 or G729. ## Default order if CODEC_PRIORITY is "" : OPUS,G722,G711U,G711A,G726,G729. ## If the CODEC_PRIORITY parameter is defined, then upon building INVITE, list the codecs set in CODEC_PRIORITY regardless of enable/disable setting for the codec. ## Opus is not supported by J189 SIP R4.0.6.1 and later. ## This parameter is supported by: ## J100 SIP R4.0.5.0 and later, J189 SIP R4.0.6.1 and later ## 96x1 SIP R7.1.9.0 and later ## SET CODEC_PRIORITY "G722,OPUS,G711A,G729,G726" ## ############ 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 Connect Application SIP R1.0.0.1 and later ## Avaya IX Workplace 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 IX Workplace 3.1.2 and later ## Avaya Vantage Connect Application SIP R1.0.0.0 and later ## SET VIDEO_MAX_BANDWIDTH_ANY_NETWORK 1000 ## ## FORWARD_ERROR_CORRECTION specifies Forward Error Correction (FEC) for video streams. ## Value Operation ## 0 Disabled ## 3 Scopia Proprietary FEC in SDP negotiation (Default) ## This parameter is supported by: ## Avaya Vantage built-in Unified Communication Experience R3.0.0.0 and later ## Avaya IX Workplace 3.4.4 and later ## SET FORWARD_ERROR_CORRECTION 0 ## ############ CAMERA SETTINGS ########### ## ## CAMERASTAT specifies whether to enable or disable the embedded camera and external third-party USB camera. ## Value Operation ## 0 Disabled. User has no option to control whether to enable/disable the camera. ## 1 Enabled. User has no option to control whether to enable/disable the camera. ## 2 User has an option to control whether to enable/disable the camera. ## This parameter is supported by: ## Avaya Vantage Devices SIP R2.0.1.0 and later, the default is 1 in pre R3.0.0.3 and 2 in R3.0.0.3 and later. In pre R3.0.0.3 user can disable/enable the camera from the Android settings application. ## In R3.0.0.3, user can turn off/on the camera from Android quick settings. ## SET CAMERASTAT 0 ## ## 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: ## Avaya Vantage Devices SIP R3.0.0.0 and later ## H1xx SIP R1.0 and later ## SET CAMERA_ANTIFLICKER_POWERLINE_FRQ 1 ## ############ 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 ## ## ALLOW_AUDIO_ROUTE_TO_HDMI specifies whether audio will be routed to HDMI port or not. ## Value Operation ## 0 Audio will NOT be routed to HDMI port (default). ## 1 Audio will be routed to HDMI port per Android transducer precedence list. ## Note: When connecting external screen using HDMI, Android can route the audio to the HDMI port (per Android transducer list precedence) no matter whether there are built-in speakers in the external screen or not. ## This parameter can be used to disable route audio to HDMI port. ## Note: Call audio and alarm sounds are not routed to HDMI port. Only Media and Notification sounds are sent to HDMI port when ALLOW_AUDIO_ROUTE_TO_HDMI is "1". ## This parameter is supported by: ## Avaya Vantage Devices SIP R3.0.0.0 and later ## SET ALLOW_AUDIO_ROUTE_TO_HDMI 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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, J159 SIP R4.0.3.0 and later, J189 SIP R4.0.6.1 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 ## ## OVERRIDE_SOFTKEY_ACTIVE specifies if default softkeys will be shown for CA lines in ACTIVE state. ## Value Operation ## 0 Default softkeys will be shown (default) ## 1 Default softkeys will NOT be shown ## This parameter is supported by: ## J100 SIP R4.0.6.0 and later (Not supported by J129), J189 SIP R4.0.6.1 and later ## SET OVERRIDE_SOFTKEY_ACTIVE 0 ## ## SOFTKEY_ACTIVE appends new softkeys to the existing active call softkeys (hold/transfer/endcall/conf/newcall/details). Softkeys supports in-band DTMF FAC codes. This could also be used for charge codes. ## The following format shall be used: ## ADD SOFTKEY_ACTIVE "type=;action=;label=