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Change menu language 9608
- Thread starter Aerap87
- Start date
In your 46xxsettings file, you specify which additional languages to load beyond English. Each firmware release comes in a zip file and included in that is the language packs for that firmware.
Code:
#################### 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.xmlkyle555 posted the section of the 46xxsettings.txt file associated with languages if the 9608 is running SIP firmware. There is a similar section if the phone is running H323 firmware. The same concept applies - you have to load the language files into the phone before they can be chosen in the menu.
- Thread starter
- #4
I'm putting the file 46xxsettings the language, the more the extension does not load.
I've done several changes and nothing.
follows the file below:
I've done several changes and nothing.
follows the file below:
Code:
######################################################################################
##
## 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 DialingYeah... that won't work.
You need to uncomment the lines you want to use in the settings file (remove the # at the beginning). And you need to load certain language XML files before you can specify what the default language would be.
Example:
## SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml
Change to:
SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml
That line will tell the phone to download those language files. Once that is specified, you can tell the phone to use a certain language as a default - like you did in your settings file.
What model of phone are you trying to set to Brazilian Portuguese, and is it SIP or H.323?
You need to uncomment the lines you want to use in the settings file (remove the # at the beginning). And you need to load certain language XML files before you can specify what the default language would be.
Example:
## SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml
Change to:
SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml,Mlf_LatinAmericanSpanish.xml
That line will tell the phone to download those language files. Once that is specified, you can tell the phone to use a certain language as a default - like you did in your settings file.
What model of phone are you trying to set to Brazilian Portuguese, and is it SIP or H.323?
- Thread starter
- #6
- Thread starter
- #8
I'm putting in H323 file yes.
follows:
follows:
Code:
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 #######################- Thread starter
- #10
Did you put all of the language files from the h.323 installation zip file on the server? Does the name of the language file in the 46xxsettings.txt file match the name from the zip file? If you are setting the default language (LANGSYS) to Portuguese, then you do not need to specify the same against LANG1FILE.
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