Follow along with the video below to see how to install our site as a web app on your home screen.
Note: This feature may not be available in some browsers.
#################### LANGUAGE SETTINGS ####################
##
## System-Wide Language
## Contains the name of the default system language file
## used in the phone. The filename should be one of the
## files listed in the LANGUAGES parameter. If no
## filename is specified, or if the filename does not
## match one of the LANGUAGES values, the phone shall use
## its built-in English text strings. 0 to 32 ASCII
## characters. Filename must end in .xml
##
## NOTE:
## For 96xx SIP Release 1.0 phones only, all language
## filenames begin with Mls_Spark_. For example,
## Mls_Spark_English.xml
##
## For 96xx SIP Release 2.0 and later and for 16CC phones,
## all language filenames begin with Mlf_
##
## SET SYSTEM_LANGUAGE Mlf_English.xml
##
## The language files of 96x0 SIP 2.6.13 and later in the 96x0 SIP firmware distributions are different than 96x1
## and therefore their filenames were changed to Mlf_S96x0_<Language name>.xml.
## Mlf_<language name>.xml filename convention is used by:
## 1. 96x1 SIP Release 6.0 and later and
## 2. 96xx SIP Release 2.0 up to 2.6.13 (excluded).
## In mutual environment of 96x0 SIP and 96x1 SIP phones there shall be use of IF conditional statement
## base on MODEL/GROUP to assign different language files for each phone family.
## SET SYSTEM_LANGUAGE Mlf_English.xml
## SET SYSTEM_LANGUAGE Mlf_S96x0_English.xml
##
## Installed Languages
## Specifies the language files to be installed/downloaded
## to the phone. Filenames may be full URL, relative
## pathname, or filename. (0 to 1096 ASCII characters,
## including commas). Filenames must end in .xml.
##
## NOTE:
## For 96xx SIP Release 1.0 phones only, all language
## filenames begin with Mls_Spark_ For example,
## Mls_Spark_English.xml
##
## For 96xx SIP Release 2.0 and later and for 16CC phones,
## all language filenames begin with Mlf_
##
## The language files of 96x0 SIP 2.6.13 and later in the 96x0 SIP firmware distributions are different than 96x1
## and therefore their filenames were changed to Mlf_S96x0_<Language name>.xml.
## Mlf_<language name>.xml filename convention is used by:
## 1. 96x1 SIP Release 6.0 and later and
## 2. 96xx SIP Release 2.0 up to 2.6.13 (excluded).
## In mutual environment of 96x0 SIP and 96x1 SIP phones there shall be use of IF conditional statement
## base on MODEL/GROUP to assign different language files for each phone family.
##
## SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml
## SET LANGUAGES Mlf_S96x0_German.xml,Mlf_S96x0_ParisianFrench.xml,Mlf_S96x0_LatinAmericanSpanish.xml
######################################################################################
##
## AVAYA IP TELEPHONE CONFIGURATION FILE TEMPLATE
## *** 11 May 2015 ***
##
## 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:
##
## 96x1 H.323 R6.6
## B189 H.323 R6.6
## 96x1 SIP R6.5
## 96x0 H.323 R3.2.4
## 96x0 SIP R2.6.13
## 46xx H.323 R2.9.2
## 46xx SIP R2.2.2
## 364x SIP R1.1
## 3631 H.323 R1.3.0
## 16xx H.323 R1.3.3
## 16CC SIP R1.0
## 1603 SIP R1.0
## 1692 H.323 R1.4
## Softphone SIP R2.1
## H1xx SIP R1.0
##
######################################################################################
##
## 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 AvayatelephonewithMACaddress$MACADDR
##
######################################################################################
##
## List of MODEL4 values for models which support MODEL4 as testable parameter in the
## configuration file (for example: IF $MODEL4 SEQ 1603 GOTO SETTINGS16XX).
## 1603
## 1608
## 1616
## 1692
## 16CC
## 3631
## 364X
## 4601
## 4602
## 4610
## 4620
## 4621
## 4622
## 4625
## 4630
## 9610
## 9620
## 9630
## 9640
## 9650
## 9670
## 9608
## 9611
## 9621
## 9641
## B189
## H175
##
######################################################################################
##
## COMMON SETTINGS
##
## Settings in this section will be processed by all telephones,
## but not all parameters are supported by all telephones or all software releases.
## Settings for parameters that are not supported will be ignored.
## For more information, see the Administrator's Guide available at support.avaya.com
##
############### LAYER 2 VLAN AND QOS SETTINGS ##############
##
## L2Q specifies whether layer 2 frames generated by the telephone will have IEEE 802.1Q tags.
## Value Operation
## 0 Auto - frames will be tagged if the value of L2QVLAN is non-zero (default).
## 1 On - frames will always be tagged.
## 2 Off - frames will never be tagged.
## Note: This parameter may also be set via DHCP or LLDP.
## This parameter is supported by:
## 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
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 96x0 SIP R1.0 and later
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP R1.0 and later
## SET L2QAUD 7
##
## L2QVID specifies the layer 2 priority value for video frames generated by the telephone.
## Valid values are 0 through 7; the default value is 5.
## This parameter is supported by:
## H1xx SIP R1.0 and later
## SET L2QVID 7
##
## L2QSIG specifies the layer 2 priority value for signaling frames generated by the telephone.
## Valid values are 0 through 7; the default value is 6.
## Note: This parameter may also be set via LLDP or H.323 signaling,
## which would overwrite any value set in this file.
## This parameter is supported by:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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
## 46xx H.323 R2.3.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 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 recieved 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 LANSEP==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.
## 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 (default).
## 1 Enabled if VLANSEP, L2Q, L2QVLAN and PHY2VLAN are set appropriately
## This parameter is supported by:
## 96x1 H.323 R6.6 and later
## 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:
## 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
## 46xx H.323 R2.3.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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
## 46xx H.323 R2.3.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 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:
## 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
## 46xx H.323 R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 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:
## 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:
## 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
## 46xx H.323 R2.1 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## SET DHCPSTD 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:
## 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
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 96x0 SIP R1.0 and later
## 46xx H.323 R1.8 and later
## 46xx SIP R2.2 and later
## 16xx H.323 R1.0 and later
## 16CC SIP 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:
## 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:
## 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
## 46xx H.323 R1.6 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later
## 3631 H.323 R1.0 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP R1.0 and later
## SET DNSSRVR 198.152.15.15
##
## DOMAIN specifies a character string that will be appended to parameter values
## that are specified as DNS names, before the name is resolved.
## The value can contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 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
## 46xx H.323 R1.6 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later (up to 63 characters only)
## 3631 H.323 R1.0 and later
## 16xx H.323 R1.0 and later
## 16CC SIP R1.0 and later
## 1603 SIP 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:
## 96x1 H.323 R6.0 and later
## 96x0 H.323 R2.0 and later
## SET QKLOGINSTAT 0
##
################### SERVER SETTINGS (H.323) ################
##
## MCIPADD specifies a list of H.323 call server IP addresses.
## Addresses can be in dotted-decimal (IPv4), colon-hex (IPv6, if supported), or
## DNS name format, separated by commas without any intervening spaces.
## The list can contain up to 255 characters; the default value is null ("").
## A value set in this file will replace any value set for MCIPADD via DHCP.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 46xx H.323 R1.0 and later
## 3631 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:
## 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:
## 96x1 H.323 R6.0 and later
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 46xx H.323 R2.1 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:
## 96x1 H.323 R6.0 and later
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 46xx H.323 R2.8.1 and later
## 16xx H.323 R1.0 and later
## 1692 H.323 R1.4 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:
## 96x1 H.323 R6.0 and later
## B189 H.323 R1.0 and later
## 96x0 H.323 R1.0 and later
## 46xx H.323 R2.1 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:
## 96x1 H.323 R6.6 and later
## B189 H.323 R6.6 and later
## SET UDT 960
##
## 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:
## H1xx SIP R1.0 and later
## 96x1 H.323 R6.0 and later releases
## B189 H.323 R1.0 and later
## 96x0 H.323 R3.1 and later releases
## SET GRATARP 0
##
######### GUEST LOGIN (AND VISITING USER) SETTINGS (H.323 only) #########
##
## 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
## 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.
## 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.
## SET GUESTWARNING 5
##
################# SERVER SETTINGS (SIP) ################
##
## 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:
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later (up to 60 characters only)
## 16CC SIP R1.0 and later
## 1603 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
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later
## 16CC SIP R1.0 and later
## 1603 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 SIP controller designators,
## separated by commas without any intervening spaces,
## where 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).
## [:port] is an optional port number.
## [;transport=xxx] is an optional transport type where xxx can be tls, tcp, or udp.
## If a port number is not specified a default value of 5060 for TCP and UDP or 5061 for TLS is used.
## If a transport type is not specified, a default value of tls is used.
## The value can contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## 96x0 SIP R2.4.1 and later
## 1603 SIP R1.0 and later
## H1xx SIP R1.0 and later; udp is not supported.
## SET SIP_CONTROLLER_LIST proxy1:5060;transport=tls,proxy2:5060;transport=tls
##
## 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:
## Not supported in 96x1 SIP R6.2 and later; the default value is simultaneous.
## 96x1 SIP R6.0.x; the default value is alternate.
## 96x0 SIP R2.4.1 and later; the default value is alternate.
## SET SIPREGPROXYPOLICY simultaneous
##
## SIMULTANEOUS_REGISTRATIONS specifies the number of Session Managers
## with which the telephone will simultaneously register.
## Valid values are 1, 2 or 3; the default value is 3.
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## 96x0 SIP R2.6 and later
## H1xx SIP R1.0 and later
## 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)
## This parameter is supported by:
## 96x1 SIP R6.4 and later
## 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 supported by:
## 96x1 SIP R6.0 and later
## 96x0 SIP R2.4.1 and later
## 1603 SIP R1.0 and later
## H1xx SIP R1.0 and later
## SET ENABLE_PPM_SOURCED_SIPPROXYSRVR 1
##
## CONFIG_SERVER specifies the address of the Avaya configuration server.
## Zero or one IP address in dotted decimal or DNS name format,
## optionally followed by a colon and a TCP port number.
## The value may contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 96x0 SIP R2.6.7 and later
## H1xx SIP R1.0 and later
## SET CONFIG_SERVER ppm.myco.com
##
## CONFIG_SERVER_SECURE_MODE specifies whether HTTP or HTTPS is used to access the configuration server.
## Value Operation
## 0 use HTTP (default for 96x0 R2.0 through R2.5)
## 1 use HTTPS (default for other releases and products)
## 2 use HTTPS if SIP transport mode is TLS, otherwise use HTTP
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.0 and later
## 1603 SIP R1.0 and later
## SET CONFIG_SERVER_SECURE_MODE 1
##
## 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
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later (only supports one address)
## 16CC SIP R1.0 and later
## 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
## 16CC SIP R1.0 and later
## 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
## 16CC SIP R1.0 and later
## 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R1.0 through R2.4
## 16CC SIP R1.0 and later
## SET ENABLE_AVAYA_ENVIRONMENT 1
##
##
######### NON-AVAYA ENVIRONMENT SETTINGS (SIP ONLY) ########
##
## MWISRVR specifies a list of addresses of Message Waiting Indicator servers.
## Addresses can be in dotted-decimal or DNS name format,
## separated by commas without any intervening spaces.
## The value can contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## 96x0 SIP R2.0 and later
## H1xx SIP R1.0 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:
## 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:
## 96x1 SIP R6.0 and later
## 96x0 SIP R2.0 and later
## H1xx SIP R1.0 and later
## SET PHNNUMOFSA 3
##
################## TIME SETTINGS (SIP ONLY) #################
##
## SNTPSRVR specifies a list of addresses of SNTP servers.
## Addresses can be in dotted-decimal or DNS name format,
## separated by commas without any intervening spaces.
## The list can contain up to 255 characters; the default value is null ("").
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later (only supports one address)
## 16CC SIP R1.0 and later
## 1603 SIP R1.0 and later
## H1xx SIP R1.0 and later
## SET SNTPSRVR 192.168.0.5
##
## 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:
## H1xx SIP R1.0 and later
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 364x SIP R1.1 and later (see Note below)
## 16CC SIP R1.0 and later
## 1603 SIP R1.0 and later
## Note: For the 364x, only values of 00, 15, 30 and 45 are supported for minutes,
## other values are set to 00.
## 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:
## H1xx SIP R1.0 and later
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 16CC SIP R1.0 and later
## 1603 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 except the 46xx is 2SunMar2L
## (the second Sunday in March at 2AM local time).
## The default value for 46xx telephones is 1SunApr2L
## (the first Sunday in April at 2AM local time),
## which is now obsolete for North America so it should be set below.
## See the Administrator's Guide for format and setting alternatives.
## This parameter is supported by:
## H1xx SIP R1.0 and later
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 16CC SIP R1.0 and later
## 1603 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 except the 46xx is 1SunNov2L
## (the first Sunday in November at 2AM local time).
## The default value for 46xx telephones is LSunOct2L
## (the last Sunday in October at 2AM local time),
## which is now obsolete for North America so it should be set below.
## See the Administrator's Guide for format and setting alternatives.
## This parameter is supported by:
## H1xx SIP R1.0 and later
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## 46xx SIP R2.2 and later
## 16CC SIP R1.0 and later
## 1603 SIP R1.0 and later
## SET DSTSTOP 1SunNov2L
##
################## 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.5 and later
## 1603 SIP R1.0 and later
## Note: For Avaya Distributed Office configurations prior to R2.0, this parameter must be set to 60.
## SET WAIT_FOR_REGISTRATION_TIMER 60
##
## REGISTERWAIT specifies the number of seconds between re-registrations with the current server.
## This parameter is supported by:
## 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.
## 46xx SIP R2.2 and later; valid values are 0 to 65535; the default value is 3600.
## 364x SIP R1.1 and later; valid values are 60 to 3600; the default value is 3600.
## 16CC SIP R1.0 and later; valid values are 10 to 1000000000; the default value is 3600.
## 1603 SIP R1.0 and later; valid values are 30 to 86400; the default value is 900.
## SET REGISTERWAIT 1000
##
## RECOVERYREGISTERWAIT specifies a number of seconds.
## If no response is received to a REGISTER request within the number of seconds specified
## by WAIT_FOR_REGISTRATION_TIMER, the telephone will try again after a randomly selected
## delay of 50% to 90% of the value of RECOVERYREGISTERWAIT.
## Valid values are 10 through 36000; the default value is 60.
## This parameter is supported by:
## 96x1 SIP R6.0 and later; not supported in R6.2 and later.
## 96x0 SIP R2.4.1 and later
## 1603 SIP R1.0 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:
## 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:
## 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:
## 96x1 SIP R6.0 and later.
## H1xx SIP R1.0 and later
## 96x0 SIP R2.0 and later.
## 364x SIP R1.1 and later (currently used only for avaya-ccs-profile
## subscription as message-summary is controlled by REGISTERWAIT timer).
## 1603 SIP R1.0 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:
## 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.0 and later
## 1603 SIP R1.0 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:
## 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R1.0 and later
## 1603 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R1.0 and later
## 1603 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R1.0 and later
## 1603 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:
## 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; the default value is 60.
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.6 and later
## 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:
## 96x1 SIP R6.0 and later, not supported in R6.2 and later.
## 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.4.1 and later
## 364x SIP R1.1 and later
## 1603 SIP R1.0 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.4.1 and later
## 364x SIP R1.1 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:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.4.1 and later
## 364x SIP R1.1 and later
## 1603 SIP R1.0 and later
## SET RDS_INITIAL_RETRY_ATTEMPTS 20
##
############# CONFERENCING SETTINGS (SIP ONLY) #############
##
## CONFERENCE_FACTORY_URI specifies the URI for Avaya Aura Conferencing.
## 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:
## 96x1 SIP R6.2.1 and later
## H1xx SIP R1.0 and later
## SET CONFERENCE_FACTORY_URI "93375000@avaya.com"
##
## 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:
## 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:
## 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)
## SET ENABLE_PRESENCE 1
##
## PRESENCE_SERVER specifies the an 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:
## 96x1 SIP R6.2 and later
## 96x0 SIP R2.6.6 and later
## H1xx SIP R1.0 and later
## 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 supported by:
## 96x1 SIP R6.3 and later (values 0-1)
## H1xx SIP R1.0 and later (values 0-2)
## SET PRESENCE_ACL_CONFIRM 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
## 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:
## H1xx SIP R1.0 and later
## 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:
## H1xx SIP R1.0 and later
## SET AWAY_TIMER_VALUE 0
##
########### 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:
## 96x1 SIP R6.2 and later
## SET INSTANT_MSG_ENABLED 1
##
############### 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:
## 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:
## 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:
## 96x1 SIP R6.3 and later.
## SET EXCHANGE_SERVER_MODE 1
##
## PROVIDE_EXCHANGE_CONTACTS specifies whether menu item(s) for Exchange Contacts are displayed.
## Value Operation
## 0 Not displayed
## 1 Displayed (default)
## This parameter is supported by:
## 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:
## 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:
## 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 value can contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 96x1 SIP R6.0 and later
## H1xx SIP R1.0 and later
## 96x0 SIP R2.5 and later
## SET EXCHANGE_USER_DOMAIN exchange.myco.com
##
## EXCHANGE_EMAIL_DOMAIN specifies the Exchange email domain.
## The value can contain 0 to 255 characters; the default value is null ("").
## This parameter is supported by:
## 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:
## 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:
## 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:
## 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:
## 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:
## 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;participantcode;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;hostcode;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;meetingid;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 meetingpin;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:
## 96x1 SIP R6.0 and later
## 96x0 SIP R1.0 and later
## SET SPEAKERSTAT 1
##
## MUTE_ON_REMOTE_OFF_HOOK controls the speakerphone muting for a remote-initiated
## (a shared control or OOD-REFER) speakerphone off-hook.
##
## Valid values are 0 and 1
## 0 - the speakerphone is Unmuted
## 1 - the speakerphone is Muted
##
## The default value is 1 (Muted) for 96x1 SIP R6.3
## The default value is 0 (Unmuted) for 96x1 SIP R6.3.1 and later, H1xx SIP R1.0 and later
##
## This parameter is supported by:
## 96x1 SIP R6.3 and later
## H1xx SIP R1.0 and later
##
## 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
##
## 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:
## 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:
## 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:
## 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:
## 96x1 SIP R6.2 and later; valid values 0 through 3; the default value is 0.
## 96x1 SIP R6.0.x; valid values 0 through 1; the default value is 0.
## 96x0 SIP R2.6.5 and later; valid values 0 through 3; the default value is 0.
## 96x0 SIP R2.0 through R2.6.4; valid values 0 through 1; the default value is 0.
## SET DISPLAY_NAME_NUMBER 0
##
## HOTLINE specifies zero or one hotline number.
## Valid values can contain up to 30 dialable characters (0-9, *, #).
## The default value is null ("").
## This parameter is supported by:
## 96x1 SIP R6.2 and later
## SET HOTLINE ""
##
## PLAY_TONE_UNTIL_RTP specifies whether locally-generated ringback tone will stop
## as soon as SDP is received for an early media session, or whether it will continue
## until RTP is actually received from the far-end party.
## Value Operation
## 0 Stop ringback tone as soon as SDP is received
## 1 Continue ringback tone until RTP is received (default)
## This parameter is supported by:
## 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:
## 96x1 SIP R6.2 and later
## SET PLUS_ONE 1
##
## 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:
## 96x1 SIP R6.2 and later
## SET TEAM_BUTTON_RING_TYPE 3
##
## 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 and later
## 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:
## 96x1 SIP R6.2 and later
## H1xx SIP R1.0 and later
## SET LOCALLY_ENFORCE_PRIVACY_HEADER 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
## This parameter is supported by:
## 96x1 SIP R6.2 and later
## H1xx SIP R1.0 and later
## 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:
## 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
SETTINGS96X1H323
####### Add settings for 96x1 H.323 telephones below #######
##
## Note: Starting R6.6 release language file name convention was changed from ?mlf_s96x1_...? to ?mlf_96x1_...?
## In addition, the template English filename was changed from "..._template_english.txt to "..._template_en.txt".
##
## LANGSYS specifies the name of a language file to use for the default language.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results in built-in English language strings being used.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
SET LANGSYS mlf_96x1_v132_portuguese.txt
##
## LANG1FILE specifies the name of the language file for the first user-selectable language.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results no user-selectable language for this parameter.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
SET LANG1FILE mlf_96x1_v132_portuguese.txt
##
## LANG2FILE specifies the name of the language file for the second user-selectable language.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results no user-selectable language for this parameter.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
## SET LANG2FILE mlf_96x1_v131_russian.txt
##
## LANG3FILE specifies the name of the language file for the third user-selectable language.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results no user-selectable language for this parameter.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
## SET LANG3FILE mlf_96x1_v131_spanish_latin.txt
##
## LANG4FILE specifies the name of the language file for the fourth user-selectable language.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results no user-selectable language for this parameter.
## This parameter is supported by:
## 96x1 H.323 R6.0 and later
## SET LANG4FILE mlf_96x1_v131_korean.txt
##
## LANGLARGEFONT specifies the name of the language file for the display of large text.
## The file name can contain 0-32 ASCII characters.
## The default value is null, which results in the Text Size option not being offered.
## This parameter is supported by:
## 96x1 H.323 R6.1 and later
SET LANGLARGEFONT mlf_96x1_v132_portuguese.txt
##
## VOXFILES specifies a list of voice language files that determine the
## list of Voice Dialing Languages that is presented to the user.
## The list can contain up to 255 characters; the default value is null ("").
## File names are separated by commas without any intervening spaces.
## The first file in the list will be downloaded by default.
## The first three characters of the filename indicate the language supported as follows:
## DUN Dutch
## ENG U.K. English
## ENU U.S. English
## FRF Parisian French
## GED German
## ITI Italian
## PTB Brazilian Portuguese
## SPE European Spanish
## This parameter is supported by:
## 96x1 H.323 R6.2 and subsequent dot releases, but not by R6.3 and later
SET VOXFILES PTB_S20_v3.tar,PTB_S20_FL_v1.tar
##
##### End of 96x1 H.323 product line-specific settings #####
##
################# END OF 96X1 SETTINGS #######################