The following source numbers affect all users on the IP Office system. They are entered through the Source Numbers tab of the NoUser user. These source numbers are informally referred to as NUSNs.

Changes to these source numbers require a system reboot to become effective.

  • ATM4U_PCS7_RINGDETECT

    For some cellular or mobile interfaces connected to a IP500 ATM4U card, the card may not detect the ring signal. For PCS4 and higher card, this NoUser source number can be used activate alternate ring detection.

  • ALLOW_5410_UPGRADES

    This option must be present for 5410 phones to update their firmware.

  • B_DISABLE_SIP_IPADDR

    Disables the blacklisting of SIP device registration based on the device IP address. Refer to the Avaya IP Office™ Platform Security Guidelines manual.

  • BST_MESSAGE_FOR_YOU

    Replace the date and time shown on BST phones when idle with Message for you or Messages for you when the user has new voicemail messages. This source number can also be set as a source number for individual users.

  • CIPHERS_LEVEL_H323=<N>

    Sets the minimum cipher strength the IP Office accepts on TLS connections for H.323 phones and trunks. Not used for clients where ciphers are enabled and chosen based on those offered by the TLS server.

    • Supported for IP Office R11.1.2.x releases. For IP Office R11.3.1 and higher, this NUSN is replaced by the System > Certificates > H.323 Security Level security setting.

    • Note: The default level 1 (medium strength) is used if no source number is specified.

    The value <N> is set as follows:

    • Low (0) - Accept low, medium, and high-strength ciphers. Low and medium on IP500 V2 systems.

    • Medium (1) - Accept medium and high-strength ciphers. Medium on IP500 V2 systems.

    • High (2) - Accept high-strength ciphers. Not supported for IP500 V2 systems.

  • CIPHERS_LEVELS_SIP=<N>

    Sets the minimum cipher strength the IP Office accepts on TLS connections for SIP phones and trunks. Not used for clients where ciphers are enabled and chosen based on those offered by the TLS server.

    • Supported for IP Office R11.1.2.x releases. For IP Office R11.3.1 and higher, this NUSN is replaced by the System > Certificates > SIP Security Level security setting.

    • Use the same values as CIPHERS_LEVELS_H323 but sets the cipher level the IP Office accepts for SIP TLS connections.

  • DECT_REVERSE_RING

    By default, when this parameter is not set, calls on DECT phones associated with a CTI application will ring as a priority call. When this parameter is set, DECT phones ring as a normal, external or internal call.

  • DISTINCT_HOLD_RINGBACK

    Used to display a specific message about the call type for calls returning after timing out from being parked or held. If set, such calls display Return Call - Held or Return Call – Parked rather than connected party name or line name.

  • ENABLE_J100_FQDN

    Use FQDN rather than IP addresses in the server address values provided to J100 Series phones. This requires that the FQDN values are correctly routable by the customer DNS servers and that the phones use the DNS server address (either obtained through DHCP or set manually).

  • ENABLE_J100_AUTO_UPDATE_POLICY

    Add settings for J100 Series phone auto-upgrade support to the system's auto-generated 46xxsettings.txt file. Refer to the IP Office SIP Telephone Installation Notes manual.

  • Enable_OTT

    Enable one touch transfer for all users. See One Touch Transferring. This source number can also be set as a source number for individual users.

  • EQNX_CONTACT_MATCHING_MIN_DIGITS=<N>

    By default the Avaya Workplace Client requires at least 10 digits for contact matching (8 for Bahrain). This NoUser source number can be used to define the minimum digits for contact matching for countries where national dial plan phone numbers are less than 10 digits.

  • FORCE_HANDSFREE_TRANSFER

    If set, when using the handsfree announced transfer process (see Handsfree Announced Transfers), both the transfer enquiry and transfer completion calls are auto-answered. Without this setting only the transfer enquiry call is auto-answered.

  • HIDE_CALL_STATE

    Used to hide the call status information, for example Dial and Conn, shown on older DS phones such as 2400, 4400 and 5400 Series. Used in conjunction with the LONGER_NAMES source number.

  • HOLD_MUSIC_TIMEOUT=<seconds>

    By default, line alternate music sources remain connected for 30 seconds after they stop being used. You can use this source number to change the disconnect timeout. The supported range is 1 to 600 seconds.

  • LONGER_NAMES

    Used to increase the length of names sent for display on older DS phones such as 2400, 4400 and 5400 Series.

  • MEDIA_NAT_DM_INTERNAL=N

    Used in conjunction with the setting System | VoIP | Allow Direct Media Within NAT Location. When Allow Direct Media Within NAT Location is enabled, the default behavior is to attempt direct media between all types of devices (H323 and SIP remote workers and IP Office Lines behind a NAT). For routers using H323 ALG or SIP ALG, it can be desirable to only attempt direct media between certain device types. In this case, set this NoUser user source number where N is the sum of the following values:

    • 1 = Include H323 phones.

    • 2 = Include SIP phones.

    • 4 = Include IP Office lines.

    For example, if the router has SIP ALG that cannot be disabled, to disable attempting NAT direct media for SIP devices, set MEDIA_NAT_DM_INTERNAL=5 to include only H323 phones and IP Office Lines.

  • NI2_CALLED.../NI2_CALLING...

    The following NoUser source numbers are applied to calls on ETSI PRI trunks:

    • NI2_CALLED_PARTY_PLAN=X

      Forces the NI2 Called Party Numbering plan for ETSI PRI trunks, where X equals UNKNOWN or ISDN.

    • NI2_CALLED_PARTY_TYPE=X

      Forces the NI2 Called Party Numbering type for ETSI PRI trunks, where X equals UNKNOWN, INT, NATIONAL or SUBSCRIBER.

    • NI2_CALLING_PARTY_PLAN=X

      Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks, where X equals UNKNOWN or ISDN.

    • NI2_CALLING_PARTY_TYPE=X

      Forces the NI2 Calling Party Numbering type for ETSI PRI trunks, where X equals UNKNOWN, INT, NATIONAL or SUBSCRIBER.

  • NO_DIALLED_REF_EXTERNAL

    On outgoing external calls made using short codes, the short code dialed is displayed on the user's phone and any directory matching is based on that number. This source number changes the behavior to display the telephone number output by the short codes and base directory matching on that number.

  • onex_...

    The following NoUser source numbers are used to alter the IP addresses used for Avaya one-X® Portal for IP Office access.

    • onex_l1=<IP Address>

      Sets the IP address of the one-X server that can be accessed by clients registered on the LAN1 interface.

    • onex_l2=<IP Address>

      Sets the IP address of the one-X server that can be accessed by clients registered on the LAN2 interface.

    • onex_port_l1=<IP Address>

      Sets the port of the one-X server that can be accessed by clients registered on the LAN1 interface.

    • onex_port_l2=<IP Address>

      Sets the port of the one-X server that can be accessed by clients registered on the LAN2 interface.

    • onex_port_r1=<IP Address>

      Sets the port of the one-X server that can be accessed by remote clients registered on the LAN1 interface.

    • onex_port_r2=<IP Address>

      Sets the port of the one-X server that can be accessed by remote clients registered on the LAN2 interface.

    • onex_r1=<IP Address>

      Sets the IP address of the one-X server that can be accessed by remote clients registered on the LAN1 interface.

    • onex_r2=<IP Address>

      Sets the IP address of the one-X server that can be accessed by remote clients registered on the LAN2 interface.

  • PHONE_LANGUAGES

    Cause an IP Office system to output a set of language files that can then be used to customize the text used on some phones. Refer to the Avaya IP Office Locale Settings manual.

  • PRESERVED_CONN_DURATION=<Minutes (1 to 120)>

    When System | Telephony | Telephony | Media Connection Preservation is enabled, active calls are preserved for up to 120 minutes before being disconnected.. This NoUser source number can be used to adjust the duration in the range 1 to 120 minutes.

  • PRESERVED_NO_MEDIA_DURATION=<Minutes (1 to 120)>

    When System | Telephony | Telephony | Media Connection Preservation is enabled, calls on which no RTP, RTCP or speech is detected are disconnected after 10 minutes. This NoUser source number can be used to adjust the duration in the range 1 to 120 minutes.

  • PUBLIC_HTTP=<File server address>

    If the IP Office is using the HTTP Redirection settings, this source number can be used to set a separate redirection address to be given to remote phones.

  • REPEATING_BEEP_ON_LISTEN

    By default, if you set Beep on Listen, when a user invokes Call Listen they hear an entry tone (3 beeps) only at the start of the call. When this parameter is set, they also hear a beep every 10 seconds.

  • RTCP_COLLECTOR_IP=<IP Address>

    When using a Prognosis server for call quality monitoring, set the IP address of the IP Office system as configured in the Prognosis server.

  • RW_SBC_...

    Set the IP addresses that remote SIP extensions should use to connect to the IP Office via an ASBCE. For R11.1.2.4 and higher, these have been replaced with settings on the System | LAN | Network Topology menus.

  • SET_46xx_PROCPSWD=<NNNNN>

    Set the new password indicated to phones through the auto-generated 46xxsettings.txt file.

  • SET_96xx_SIG=<X>

    When set, inserts the line SET SIG X into the auto-generated 46xxsettings.txt settings files.

  • SET_ADMINNPSWD=<NNNNN>

    Set the new admin password indicated to K100 Series phones through the auto-generated 46xxsettings.txt file.

  • SET_B199_FW_VER=<NNNN>

    If set, overrides the default B199 firmware version the IP Office system inserts into its auto-generated avayab199_fw_version.xml file. with firmware-NNNN-release.kt. Supported for IP Office R11.1.2.4 and higher.

  • SET_CDNL

    This source number can be used to add cellular direct dialing numbers to the auto-generated 46xxsettings file. For Avaya Workplace Client clients on mobile iOS and Android devices, this specifies numbers that should be dialed using the device's native dialer rather than using by the client application. For details, refer to the IP Office Avaya Workplace Client Installation Notes manual.

  • SET_HEADSYS_1

    If set, alters the operation of the headset button on 9600 Series phones via the auto-generated 46xxsettings.txt settings file. Normally the headset goes off-hook when the far end disconnects. When this option is set, the headset remains on-hook when the far end disconnects.

  • SIP_ENABLE_HOT_DESK

    By default, the use of hot-desking on J129 and H175 phones is blocked. This source numbers overrides that behavior.

  • SIP_EXTN_CALL_Q_TIMEOUT=<Minutes>

    Sets the unanswered call duration after which unanswered SIP calls are automatically disconnected. If not set, the normal default is 5 minutes. This NoUser source number can be used to adjust the duration in the range 0 (unlimited) to 255 minutes.

  • SIP_OPTIONS_PERIOD=<Minutes>

    On SIP trunks, the system periodically sends OPTIONS messages to determine if the SIP connection is active. The rate at which the messages are sent is determined by the combination of the Binding Refresh Time (seconds) set on the Network Topology tab and the SIP_OPTIONS_PERIOD parameter (in minutes). The frequency of sent messages is determined as follows:

    Target

    Method

    300 seconds

    If no SIP_OPTIONS_PERIOD parameter is defined and the Binding Refresh Time (seconds) is 0, then the default value of 300 seconds is used.

    Less than 300 seconds

    Do not define a SIP_OPTIONS_PERIOD parameter and set the Binding Refresh Time (seconds) to a value less than 300 seconds.

    More than 300 seconds

    Set both the SIP_OPTIONS_PERIOD and Binding Refresh Time (seconds) to a value greater than 300 seconds.

    The OPTIONS message period used is the smaller of the Binding Refresh Time (seconds) and the SIP_OPTIONS_PERIOD.

  • SET_STIMULUS_SBC_REG_INTERVAL=<seconds>

    Set the registration interval used for remote J100 Series phones. Reducing this is necessary if the SBC fails to send TCP RST end-to-end. The recommend value is 180 seconds. If not specified, the default is 1 hour (3600 seconds). Range 180 to 3600 seconds.

  • SUPPRESS_ALARM=1

    When set, the NoCallerID alarm is not shown in system alarms, SysMonitor and System Status Application .

  • TUI:J139_REDUCED_FEATURE_SET

    For R11.1.2.4 and higher, reinstate the pre-R11.1.2.4 feature restrictions applied to J139 phones.

  • TUI:NAME_SEARCH_MODE=<n>

    The default directory search matching used on feature phones is to simultaneously show matches against all parts of names. This source number can be used to change the name matching behavior.

    • 1 = Match starting from start of name.

    • 2 - Match starting from last word in name.

    • 3 = Match simultaneously from both 1 & 2.

    • 4 = Match from the penultimate word in name.

    • 7 = Match simultaneously from first, last and penultimate words in name.

  • TUI:NO_TOVM_SK_WHEN_VMOFF

    On feature phones, suppress the display of the To VM softkey when the user's VoiceMail setting is off.

  • VM_TRUNCATE_TIME=<Seconds: 0 to 7>

    Analog trunks can use busy tone detection to end calls. On calls that go to voicemail, to be recorded or to leave a message, when busy tone detection occurs, the IP Office indicates to the voicemail server how much to remove from the end of the recording in order to remove the busy tone segment. By default, the amount varies to match the system locale (refer to the Avaya IP Office Locale Settings manual).

    For some systems, it may be necessary to override the default if the end of analog call recordings is either being clipped or includes busy tone. This NoUser source number can be used to adjust the amount removed in the range 0 to 7 seconds.

  • VMAIL_WAIT_DURATION=<Milliseconds>

    Sets the number of milliseconds to system waits before passing call audio to Voicemail. On some systems, a delay may be required to allow completion of codec negotiation.

  • VMPRO_OOB_DTMF_OFF

    Disable the sending of out-of-band digits to the Voicemail Pro voicemail server. This may be necessary on some systems if digit presses are being recorded on calls.

  • WEBRTC_...

    These source numbers are used for WebRTC support when the User Portal user connects to the remotely using either STUN and/or TURN. For R11.1.2.4 and higher, these have been replaced with settings on the System | LAN | Network Topology menus.

  • xmpp_port...

  • These NoUser source numbers can be used Avaya one-X® Portal for IP Office to alter the ports used for XMPP connections.

    • xmpp_port_l1=<Port>

      Set the port of the XMPP server used by clients registered on the LAN1 interface.

    • xmpp_port_l2=<Port>

      Set the port of the XMPP server used by clients registered on the LAN2 interface.

    • xmpp_port_r1=<Port>

      Set the port of the XMPP server used by remote clients registered on the LAN1 interface.

    • xmpp_port_r2=<Port>

      Set the port of the XMPP server used by remote clients registered on the LAN2 interface.