85-110 S7 Software Is Now Available
For the Univerge SV8500
New Hardware
DT770G Cradle Phone - Beginning with 85-110 S7 software, the Cradle Phone DT770G, or ITL2CR- 1(BK) can be used in the SV8500. Hard Phone License (SV8500 IP ENDPOINT LICENSE) is
required for each terminal. Refer to the DT770G Release Notes for further information and details.
* This terminal cannot be used in OAI/ACD system.
UG50 Gateway - The new UG50 Gateway will be available and released in a later version of 85-110 S7
software. The UG50 provides the ability to mix Analog and Digital stations in the same chassis, as well
as supporting Analog C.O. or PRI trunks along with the stations. The UG50 is currently in Beta testing
and will be released upon completion of these tests.
Look for more information and details coming soon.
New Features in 85-110 S7
C-207 – Call Pickup – One Touch Key
This feature allows the station user to pick up an incoming call directed to a Call Pickup Group
that is assigned to a programmable Line/Feature Key on a Desktop Terminal in advance.
Desktop terminal users can pick up the call by pressing the Feature Key corresponding to the Call
Pickup Group.
If multiple Call Pickup Groups are assigned to Feature Keys, users can determine which group
is receiving a call by a flashing indication on the corresponding Feature Key Lamp when an
incoming call arrives to a station.
L-57 – Last Login Info Retention
Last Login Info Retention allows IP terminals and SIP Multiple Line terminals (herein after referred
to “IP/SIP terminal”) users to log in to any phone in free seating arrangements with their station
number as the login User ID, and continue using the station number after an interruption due to
power loss or terminal reset. If the office data is backed up and restored, they can resume using
their phones even after the system initialization while maintaining the last station number used.
For example, after transferring to another section of the company or changing seating
arrangements, if an IP/SIP terminal user logs in to a different phone once, the terminal user
does not have to log in subsequently even after a terminal reset occurs.
M-124 – My Line Number Display – D
For DT700 Series, My Line information (Name and Number) is displayed on the LCD when the terminal
is idle. Four patterns are available: [Name + Number], [Number only], [Name only] and [No display]. In
prior software versions, if the user wanted to confirm the My Line information, they would need to take
the handset off hook or press the SPEAKER key. Beginning with the 85-110 S7 software version, this
additional action is not needed in order to confirm the My Line information.
R-62 – Ring Volume in Headset Mode
Ringer Tone Volume of a DT700 series Desktop Terminal in the Headset Mode is adjusted to -20db
from normal ringer volume on a Desktop Terminal. This feature enables Ringer Tone Volume from the
speaker on the Desktop Terminal in the Headset Mode to be selected either from -20dB (lower than
normal volume) or no change (same as the normal volume without volume down to -20dB) by setting
the SFI 263 parameter.
The term “Headset Mode” refers to the condition that a Headset is activated by pressing the
Handset/Headset key, and a Desktop Terminal user hears tone such as Dial Tone (DT), Ring Back
Tone (RBT) or Reorder Tone (ROT), or line selection is still not performed.
This feature is not supported by DT710/DT770G because they cannot be connected to a Headset.
S-166 – Software Digital Tone Generator (IP-DTG)
Before 85-110 S7 software, various features cannot be applied to Standard SIP terminals
because Standard SIP terminals could not hear the PBX tones.
To solve this issue, 85-110 S7 software supports a Software Tone Generator (hereinafter
referred to as IP-DTG) that is built into the Telephony Server and it enables Standard SIP
terminals to hear PBX tones without using any external devices.
This feature can improve the services of Business/Hotel/OAI systems by using the same call
control method with IP terminals or SIP Multiple Line terminals.
S-167 – SIP Handler – Standard SIP Station
Single Line Mode - Either mode can be selected with the AUACL/AUACN command
The SIP Handler controls the Standard SIP terminal using My Line only. When a Standard SIP terminal
user talks with a destination station and puts the call on hold in the first session, and then the Standard
SIP terminal user places another call in the second session, the second session is processed as an
event of Call Transfer operation.
Multi Line Mode - Either mode can be selected with the AUACL/AUACN command
In Multi Line Mode, a Standard SIP terminal can accommodate lines of other Standard SIP terminals or
Desktop terminals as Sub Line. When a Standard SIP terminal user puts an ongoing line (the first
session) on hold, another call arrival at a different line is treated as the second session which is
different from the ongoing call in the first session.
Multi Line Mode enables a Standard SIP terminal to accept multiple calls but Single Line Mode enables
Standard SIP terminals to use more features than Multiple Line Mode.
S-168 – SP Controlled SIP Terminal
This feature was previously named S-146 – SIP Terminal Accommodation, which has been removed
and replaced with S-168. This feature allows the system to accommodate the SIP terminals mentioned
below as stations by controlling them through the SP part and the Internal PHE inside of the Telephony
Server. The SIP terminals can establish an interconnection with another Telephony Server Station such
as a Desktop terminal and use various Telephony Server services.
Although the SIP Terminal Accommodation feature allows Telephony Servers to accommodate SIP
terminals as Stations, this feature performs call control of standard SIP terminals using the common
method with PHS and there are some service conditions resulting from the conditions of PHS call
control.
Feature Enhancements in 85-110 S7
Burst Access Filter on MG-SIP
In a large-scale Call Center configuration, burst incoming calls may occur. In that case, the SV8500
could become overloaded because all the calls are routed to the SV8500. To avoid this, this
enhancement allows the MG-SIP to reject reception of further calls when the number of incoming calls
reaches a threshold value during a specific time period.
This is accomplished by assigning a period for monitoring incoming calls: t (seconds) and a limit to the
number of incoming calls: n (calls). This feature is in MG-SIP, therefore availability with 85-110 S7
depends on each MG-SIP firmware.
C-
76D – Call Park Group - D
This enhancement eliminates the requirement that a Call Park Group must consist of stations in the
same Module Group (MG) and enables the Call Park Group - D feature between Desktop Terminals
belonging to:
 Different MGs in the same IMG
 Different IMGs
L-54 – Least Cost Routing for MG-PRI
When a Telephony Server in the main office controls MGs (Media Gateways) in multiple remote offices,
adding MA-ID (Message Area ID) to the Subline enables a call to be made through the suitable
network. When a call is made from the Subline by using this feature, a call is made through a Pilot
station line by referring to the seized Subline's MA-ID. The Pilot station number is displayed on a called
station and then called party can call back properly.
MG-SIP Enhancements
A-law/u-law Conversion on MG-SIP
MG-SIP supports a Software DSP function “G.711μ-law/A-law Conversion” with the following latest
firmware.
 MG-128SIPMGK(SP-4032 MGSIP PROG-E Issue 2)
 MG-128SIPMGG(SP-4051 MGSIP PROG-G Issue 2)
International calls between countries that have different types of PCM codec via public SIP network can
converse regardless of the difference.
PAD/EC Control for MG-SIP
PAD/EC (Packet Assembler Disassembler / Echo Canceller) control (volume adjustment of incoming
and outgoing calls) of voice and FAX at MG-SIP is now available. Previously adjustment of volume
levels of SV8500 system was very limited and was made for each terminal. This feature allows
selection of various adjustment levels of the MG-SIP rather than each terminal.
Predictive Dialer Expansion
The limit of simultaneous connections for origination by the Predictive Dialer is expanded from 252 to
460 per LP. Also the value for call types (SCF: FN=128, 129) is expanded from 255 to 65535.
Real-Time Call Information on SoftPhone
This enhancement enables the telephony server to send information about the opposite party and call
records to SoftPhones in real time by using an interface similar to that used for SIP Multiple Line
terminals. A maximum of 24 digits can be sent for each caller ID. Previously, recorded incoming and
missed call history showed only 8 digits of the phone number, and overflow of 8 digits shows “*”. With
this display, the user could not recall using it. Enhancement to the maximum of 24 digits in a record
allows recall using over 8 digits in the incoming and missed call history.
Also, when the user retrieves a parked call, the call is recorded in the incoming call log, not in the
outgoing call log of your SoftPhone.
SR-MGC (E) for OAI
In an OAI system with using SR-MGC (E), the OAI facilities and Infolink messages can continue to be
used if a Telephony Server is down or a failure occurs in a network between nodes. 85-110 S7 software
supports the following OAI facilities and Infolink messages…
OAI Facilities
Infolink Messages
 SR-MGC does not support the inter-office features.
 SR-MGC can control only the OAI terminals belonging to the SR-MGC site office. Therefore, the
UAPs must be connected for each SR-MGC when the OAI terminals belonging to the site office
use the OAI services.
 This feature supports Infolink messages, but limited to the control request or status notification
within the site office.
 When the Telephony Server recovers, IP devices can be switched back from the SR-MGC.
 To clear the status of the UAP that is associated with SR-MGC, reset the UAP after the
completion of switchback.
PCPro User Right Enhancement
In previous versions a user for PCPro had to have Administrator rights for the PC. This enhancement
provides a log-in and use of PCPro with a Standard User right. Also, it manages resources like a
connection account, command execution logs and operation logs.