No User Source Codes

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Standard User Source Numbers

Value Description
AT<string> Strings beginning with AT can be used with a user called DTEDefault to configure the default settings of the control unit’s DTE port.
BST_MESSAGE_FOR_YOU If set, then the BST phone user sees the top line Message for you or Messages for you, indicating that voicemail messages are present. This source number can be used as a NoUser source number to enable the feature for all users.
BST_NO_MESSAGE_FOR_YOU If the source number above has been used as a NoUser source number to enable the feature for all BST users, this individual user source number can be used to disable the feature for selected users. If set, the user does not see a message indication when the NoUser setting BST_MESSAGE_FOR_YOU is set. The user’s phone presents the idle date/time in the normal fashion.
Enable_OTT Enable one touch transfer operation for the user.
H<Group_Name> Allows the user to receive message waiting indication of new group messages. Enter H followed by the group name, for example HMain. The group is added to the user’s Visual Voice menu.

On suitable display extensions, the hunt group name and number of new messages is displayed. Refer to the appropriate telephone user guide.

If the user is not a member of the group, a voicemail code must be set for the group’s mailbox. See the setting Group | Voicemail | Voicemail Code .

P<Telephone Number> This record sets the destination for callback (outbound alert) calls from voicemail. Enter Pfollowed by the telephone number including any necessary external dialing prefix, for example P917325559876. This facility is only available when using Voicemail Pro through which a default Callback or a user specific Callback start point has been configured. Refer to the Voicemail Pro documentation. This feature is separate from voicemail ringback and Voicemail Pro outcalling.
R<Caller’s ICLID> To allow Dial In/RAS call access only from a specified number prefix the number with a “R”, for example R7325551234.
U<User_Name or Extension#> Allows the user to receive message waiting indication of new messages. Enter U followed by the user name or extension number, for example U201. The specified user is added to the user’s Visual Voice menu.

On suitable display extensions, the user name and number of new messages is displayed. Refer to the appropriate telephone user guide.

If the user is not a trusted source and a Voicemail Code exists, the user must enter the Voicemail Code corresponding to the monitored mailbox.

V<Caller’s ICLID> Strings prefixed with a V indicate numbers from which access to the users mailbox is allowed without requiring entry of the mailbox’s voicemail code. This is referred to as “trusted source”.

For Voicemail Pro running in Intuity mode, trusted source is used for calls from programmable buttons set to Voicemail Collect and Visual Voice. Other controls are prompted for the mailbox number and then password.

System Wide Source Numbers

Value Description
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. Refer to IP Office Technical Tip 204.
ALLOW_5410_UPGRADES Previously the only control over the upgrading of 5410 phones was
controlled by the use of the turn_on.bat and turn_off.bat batch files
installed with the Manager application. Now in addition, this option
must be present for 5410 phones to update their firmware. Refer to
the IP Office Installation manual for full details.
B_DISABLE_SIP_IPADDR Disables the blacklisting of SIP device registration based on the
device IP address. Refer to the IP Office Security Guidelines document
for more details.
BST_MESSAGE_FOR_YOU If set, all BST phones display the top line Message for you or Messages for
you
, indicating that voicemail messages are present.
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_OTT Enable one touch transfer for all users.
FORCE_HANDSFREE_TRANSFER If set, when using the handsfree announced transfer process, 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, Conn,
etc, on DS phones. Used in conjunction with the LONGER_NAMES option.
Not supported for 1600 and 9600 Series phones.
LONGER_NAMES Used to increase the length of names sent for display on older
DS phones, i.e. 2400, 4400 and 5400 Series.
MEDIA_NAT_DM_INTERNAL=X Used in conjunction with the setting System
| VoIP | Allow Direct Media Within NAT Location

When Allow Direct Media Within NAT Location is set to on, The default behavior is to allow direct media between all types of devices (H323 and SIP remote workers and IP Office Lines behind a NAT). In the case of routers that have H323 or SIP ALG, it can be desirable to allow direct media only between certain categories of devices. In this case, set this NoUser user source number where X is a hex number defined as a  combination of the following flags:

  • 0x01 (includes H323 phones)
  • 0x02 (includes SIP phones)
  • 0x04 (includes IP Office Lines)

For example, if the router has SIP ALG that can’t be disabled,
you might want to disable direct media for SIP devices. To configure,
set MEDIA_NAT_DM_INTERNAL=5 to include only
H323 phones and IP Office Lines.

NI2_CALLED_PARTY_PLAN=X X = UNKNOWN or ISDN

Forces the NI2 Called Party Numbering plan for ETSI PRI trunks.

NI2_CALLED_PARTY_TYPE=X X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER

Forces the NI2 Called Party Numbering type for ETSI PRI trunks.

NI2_CALLING_PARTY_PLAN=X X = UNKNOWN or ISDN

Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks.

NI2_CALLING_PARTY_PLAN=X X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER

Forces the NI2 Calling Party Numbering type for ETSI PRI trunks.

NO_DIALLED_REF_EXTERNAL On outgoing external calls made using short codes to dial the full
number, only the short code dialed is displayed on the dialing user’s
phone and any directory matching is based on that number dialled.
On systems with this source number added to the configuration, after
dialing a short code the full number dialled by that short code is
shown and directory matching is based on that full number.
onex_l1=X Sets the IP address of the one-X server that can be accessed by
clients registered on the LAN1 interface.
onex_l2=X Sets the IP address of the one-X server that can be accessed by
clients registered on the LAN2 interface.
onex_port_l1=X Sets the port of the one-X server that can be accessed by clients
registered on the LAN1 interface.
onex_port_l2=X Sets the port of the one-X server that can be accessed by clients
registered on the LAN2 interface.
onex_port_r1=X Sets the port of the one-X server that can be accessed by remote
clients registered on the LAN1 interface.
onex_port_r2=X Sets the port of the one-X server that can be accessed by remote
clients registered on the LAN2 interface.
onex_r1=X Sets the IP address of the one-X server that can be accessed by
remote clients registered on the LAN1 interface.
onex_r2=X Sets the IP address of the one-X server that can be accessed by
remote clients registered on the LAN2 interface.
PRESERVED_CONN_DURATION=X X = time in minutes. Range = 1 to 120.

When the setting System
| Telephony | Telephony | Media Connection Preservation
is enabled, preserved calls have a maximum

duration of 120 minutes. After that time, they are hung up. Use this
setting to change the maximum duration value.

PRESERVED_NO_MEDIA_DURATION=X X = time in minutes. Range = 1 to 120.

When the setting System
| Telephony | Telephony | Media Connection Preservation
is enabled, preserved calls have a maximum

duration of 120 minutes. If monitoring RTP or RTCP and no speech is
detected, calls are hung up after 10 minutes. Use this setting to
change the default value of 10 minutes.

ProgressEndsOverlapSend See Line | VoIP.
REPEATING_BEEP_ON_LISTEN By default, if you set Beep on Listen and invoke Call Listen you’ll
hear an entry tone (3 beeps). When this parameter is set, you hear
a beep every 10 seconds when you invoke Call Listen.
RTCP_COLLECTOR_IP=X X = IP address of the IP Office system as configured in the Prognosis
server.
 RW_SBC_REG=<SBC-B1-public-SIP-IPaddr> Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
on 11xx, 12xx, and E129 phones. The IP address is used as a S1/S2
for 11xx and 12xx and for outbound-proxy-server for E129 sets.
RW_SBC_PROV=<SBC-B1-private-HTTP/S-IPaddr> Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
on 11xx, 12xx, and E129 phones. The IP address is used to determine
whether a 11xx, 12xx, or E129 set is registered as an IP
Office SBCE Remote Worker. 
RW_SBC_TLS=<SBC-public-TLS-port> Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
on 11xx, 12xx, and E129 phones. The  port is used as a S1/S2 TLS
port for 11xx and 12xx phones and as outbound-proxy-server TLS port
for E129 phones.
RW_SBC_TCP=<SBC-public-TCP-port> Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
on 11xx, 12xx, and E129 phones. The port is used as a S1/S2 TCP
port for 11xx and 12xx phones and as outbound-proxy-server TCP port
for E129 phones.
RW_SBC_UDP=<SBC-public-UDP-port> Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
on 11xx, 12xx, and E129 phones. The port is used as a S1/S2 UDP
port for 11xx and 12xx phones and as outbound-proxy-server UDP port
for E129 phones. 
SET_46xx_PROCPSWD=X X= New password

When set, the new password is indicated to phones through the auto-generated
settings file.

SET_96xx_SIG=X When set, inserts the line “SET SIG X into the auto-generated
settings files.
SET_HEADSYS_1 If set, alters the operation of the headset button on 96×1 phones
via the auto-generated 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_E129_PREFER_UDP When set, the auto-generated E129 configuration file is altered
to set the transport method as UDP regardless of whether TCP or TLS
are selected on the LAN1/LAN2 VoIP configuration settings.
SIP_ENABLE_HOT_DESK For IP Office Release 10.1, 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=X X = Number of minutes (0 (no limit) to 255).

Sets the unanswered call duration after which unanswered SIP calls
are automatically disconnected. If not set, the normal default is
5 minutes.

SIP_OPTIONS_PERIOD=X X = time in minutes. The system sends SIP options messages periodically
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 (in seconds) set on the Network Topology tab and the SIP_OPTIONS_PERIOD parameter (in minutes). The frequency of sent messages is determined
as follows:

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

To establish a period less than 300 seconds, do not define a SIP_OPTIONS_PERIOD parameter and set the Binding Refresh Time to a value less than 300 seconds. The OPTIONS message period will
be equal to the Binding Refresh Time.

To establish a period greater than 300 seconds, a SIP_OPTIONS_PERIOD parameter must be defined. The Binding Refresh Time must be set to a value greater than 300 seconds. The OPTIONS message
period will be the smaller of the Binding Refresh Time and the SIP_OPTIONS_PERIOD.

SOFTPHONE_RTP_MAX=X X = Maximum port in the range 1024 to 65534.

The maximum usable port indicated to the IP Office Video Softphone
when SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MIN are set.

SOFTPHONE_RTP_MIN=X X = Minimum port in the range 1024 to 65534.

The minimum usable port indicated to the IP Office Video Softphone
when SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MAX are set.

SOFTPHONE_RTP_RANGE_ENABLE When set, the usable ports indicated to the IP Office Video Softphone
are set via the SOFTPHONE_RTP_MIN and SOFTPHONE_RTP_MAX values.
SUPPRESS_ALARM=1 When set, suppresses the NoCallerID alarm otherwise shown in SysMonitor,
SNMP traps, email notifications, SysLog or System Status.
TUI:NAME_SEARCH_MODE=1 The default directory search matching behavior is to simultaneously
match against first and last name characters. This source number sets
the system to match from the start of the name only.
VM_TRUNCATE_TIME=X X= time in seconds. Range = 0 to 7.

On analog trunks, call disconnection can occur though busy tone
detection. When such calls go to voicemail to be recorded or leave
a message, when the call ends the system indicates to the voicemail
server how much to remove from the end of the recording in order to
remove the busy tone segment. This amount varies by system locale,
the defaults being listed below. For some systems it may be necessary
to override the default if analog call recordings are being clipped
or include busy tone. That can be done by adding a VM_TRUNCATE_TIME= setting with the required value in the range 0 to 7 seconds.

  • New Zealand, Australia,
    China, Saudi Arabia and Custom
    : 5 seconds.

  • Korea: 3 seconds.

  • Italy, Mexico, Chile,
    Colombia and Brazil
    : 2 seconds.

  • Argentina, United States,
    Canada and Turkey
    : 0 seconds.

  • All other locales: 7 seconds.

VMAIL_WAIT_DURATION=X The number of milliseconds to wait before cutting through the audio
to Voicemail. Some delay is required to allow for codec negotiation.
VMPRO_OOB_DTMF_OFF When set, disabled the sending of out-of-band digits to the Voicemail
Pro voicemail server.
xmpp_port_l1=X X = The port of the XMPP server that can be accessed by clients
registered on the LAN1 interface.
xmpp_port_l2=X X = The port of the XMPP server that can be accessed by clients
registered on the LAN2 interface.
xmpp_port_r1=X X = The port of the XMPP server that can be accessed by remote
clients registered on the LAN1 interface.
xmpp_port_r2=X X = The port of the XMPP server that can be accessed by remote
clients registered on the LAN2 interface.
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