iPECS eMG80/100& eMG800 & UCP & vUCP
Administration and Programming Manual
Issue 2.3
344
Table 4.4.5.1-1 SYSTEM ATTRIBUTES
ATTRIBUTE
DESCRIPTION
RANGE
DEFAULT
Auth Retry Count
When an Authorization code is required, the user may attempt to
enter a Valid code up to the maximum value defined in this field.
1-9
3
Simple Auth
Code Usage
System Authorization codes are entered by the user as “*” and
the code (ON) or
“*”+ the Auth code index and the code (OFF).
OFF
ON
ON
COS 7 when
Auth Fail
If a user fails to enter a valid Authorization code in the number of
attempts assigned in Auth Retry Count above, the station is
disconnected or the Station COS is changed to COS 7. In the
latter case, the user must employ COS Restore in Station User
PGM 2 to return the station to the normal COS.
OFF
ON
OFF
Unified Message
Format
System Integration Messages are sent out the defined serial or
TCP channel.
OFF
ON
OFF
Conference
Room CO Tel
Number
ISDN DID number an external party must dial to enter a
Conference room.
Max. 15
digits
Record warning
tone
When call recording is active, a tone can be sent to all
connected parties to indicate the conversation is being recorded.
OFF
ON
ON
UCP (MPB) DIFF
SERVE
Diff-Serv Code Point applied to packets fromLAN port of the
UCP (iPECS eMG LAN port of the MPB).
TOS:Route(0),Priority(8),Immediate(16),Flash(24),Flash
Override(32),Critic(40),Internetwork Control(48),Network
Control(56)
00-63
4
Device Upgrade
Mode
Transfer mode for upgrades from MPB to an iPECS device.
FTP
TFTP
FTP
CO Transfer
Tone
When a CO call is transferred to a busy extension, Ring Back
Tone or Music On Hold will be played to the CO Line.
MOH, Ring-Back Tone
Refer to
description
Ring-Back
Tone
Conference
Warning Tone
When a new member joins a conference room, the system
provides warning tone to conference members.
OFF
ON
ON
Dummy Dial
Tone
When a CO line does not provide dial tone, the system can
provide dummy dial tone.
Unused/
Use
Unused
SIP Station
Mode
SIP phones may set-up a point-to-point RTP connection (PTP)
or to assure a controlled connection, RTP can be routed via a
VoIP channel (RTD).
RTD/
PTP
Routed
SMS Center
Number
When the PSTN will be used to send SMS, the phone number of
the Short Message Service Center must be entered.
Max. 23
digits
SMS Center CLI
When the CO/IP Line will be used to receive SMS, the Caller Id
expected from the Short MSG Service Center must be defined.
Max. 23
digits
SMS Protocol
The Short Message Service Protocol must be selected to
support SMS.
None, ETSI-P1, ETSI-P2, KT-LivingNet, SIP-Text, SIP-XML,
KT IP-PBX, SKN IP-PBX, KT XML
Refer to
description
NONE