8100 features

Alarm Reports Alarm Reports Alphanumeric Display Alphanumeric Display Analog Communications Interface (ACI) Analog Communications Interface (ACI) Ancillary Device Connection Ancillary Device Connection Answer Hold Answer Hold Answer Key Answer Key Attendant Call Queuing Attendant Call Queuing Automatic Call Distribution (ACD) Automatic Call Distribution (ACD) Automatic Release Automatic Release Automatic Route Selection Automatic Route Selection Background Music Background Music Barge-In Barge-In Battery Backup – System MemoryBattery Backup – System Memory Battery Backup – System PowerBattery Backup – System Power Call Appearance (CAP) Keys Call Appearance (CAP) Keys Call Arrival (CAR) KeysCall Arrival (CAR) Keys Call Duration Timer Call Duration Timer Call Forwarding – Centrex Call Forwarding – Centrex Call Forwarding – Park and PageVoice Response System (VRS) – Call Forwarding – Park and Page Call Forwarding Call Forwarding Call Forwarding with Follow MeCall Forwarding with Follow Me Call Forwarding, Off-Premise Call Forwarding, Off-Premise Call Forwarding/Do Not Disturb Override Call Forwarding/Do Not Disturb Override Call Monitoring Call Monitoring Call Redirect Call Redirect Call Waiting/Camp-On Call Waiting/Camp-On Callback Callback Caller ID Call ReturnCaller ID Call Return

Automatic Access to VM by Caller ID

Automatic Access to VM by Caller ID
SL1100
Description
Before, when a user outside the system accessed their InMail mailbox, they dialed voice mail, then entered an access code followed by their mailbox number and password (if enabled). InMail mailbox can be associated with a specific caller ID (CID) number. When the CID number is presented to the InMail it will automatically log the user into their mailbox. This enhancement improves VM accessibility for outside callers, allowing them to simply dial the main voice mail number and be automatically logged into their mailbox.

External transfer

InMail: External Transfer Available
The software allows the InMail to perform an external transfer. This allows the InMail to route an incoming Automated Attendant call out of the SL1100 system on a new trunk based on an Speed Dial number stored in a Dial Action Table.
InMail: Softkey With Security Code Programming
InMail provides softkeys when programming the security code. These softkeys allow a user to select OK, CLEAR or EXIT following an entry of a new code.
InMail: Internal Message Notification Timer
When Message Notification places a call out, the system waits up to 30 seconds for ringback, reorder, or busy tone from the trunk. If detected, notification call out processing begins normally. If not detected, the system abandons the call and decrements the Ring No Answer (RNA) count.
InMail: Directory Dialing
Directory Dialing allows an Automated Attendant caller to reach an extension by dialing the first few letters in the extension user’s name. With Directory Dialing, the caller does not have to remember the extension number of the person they wish to reach – just the name.
The following steps describe Directory Dialing:
1.When the Automated Attendant answers, it sends the call to the Main Greeting box. The caller
must dial a digit to access Directory Dialing.
2.The Directory Dialing Mailbox plays the Directory Dialing Message which asks the caller to dial
letters for the name of the person they wish to reach.
3.The caller dials the letters for the person’s name plus #. They can dial by first name or last name, depending on how the Directory Dialing Message was recorded and the Directory Dialing Mailbox was set up.

SIP EXTENSION

SIP EXTENSION 8.1 REGISTRATION
Description
iPECS-LIK supports standard protocol equiped SIP Phone including series of LG-Ericsson SIP Phone Extension.
Operation
SIP Phone Self Programming Network Configuration
1. IP mode : Static(Fixed) / DHCP
2. Subnet Mask
3. Default gateway IP address
4. IP address
5. DNS IP address
6. Prifiling (for Wireless)
SIP Server Configuration
1. Proxy IP address : MFIM IP address
2. Proxy IP port : 5060
3. Domain : MFIM IP address
4. Registration : ON
5. Registration Timer : 30 ~ 3600 second (more than 10 minute recommended)
6. Local UDP/TCP/TLS port : 5060 or other value
7. Signaling/Transport Mode : UDP (or TCP or TLS)
Line(User) Configuration
1. SIP Account :
- Display Name (Optional) : Station Name (this will be applied to MFIM – Station Name).
- User Name (Mandatory) : Station Number (this should be same as MFIM – Device Login / Station User Login (443) / ‘Desired Number’)
- Authorization Name (Mandatory) : Login ID (this should be same as MFIM – Device Login / Station User Login (443) / ‘ID’)
- Authorization Password (Optional) : Login Password Login ID (this should be same as MFIM – Device Login / Station User Login (443) / ‘Password’)
Call Preferences
1. Call Wait : ON / OFF (When on BUSY, accept other call setup or not)
2. Call Forward
3. DTMF Type (Mandatory) : one of INFO type (After registration to system, SIP Data / SIP Phone Attributes(211) / ‘DTMF Type’ – set the same type as SIP Phone) c.f) only support INFO type
8-1
4. CODEC
5. Call Blocking … and so

HOWLER TONE

HOWLER TONE
Description
When an SLT station goes off-hook and does not initiate dialing in the Dial tone timer duration, delays dialing between digits in excess of the inter-digit time or stays off-hook at the completion of activating a feature or program, the station will receive howler tone as an error indication and the call attempt will be abandoned. In order to complete the call, the user must return to the on-hook state and restart the call.
Operation System The system will deliver howler tone automatically, as required
Conditions
1. Howler Tone is sent after a period, of about 30 seconds of error tone.
2. Lock-out occurs when howler tone starts.

Admin

Admin & Maintenance Issue
STATION ADMIN PROGRAMMING
GENERAL LCD & Button Functions
While in the PROGRAM MODE, the Liquid Crystal Display (LCD) and Flex button LEDs of an Admin Station are used to guide and indicate status of the feature. The dial-pad is most often used to enter data after selecting a data item using the Flex buttons. In some cases, pressing a Flex button will toggle the entry with the Flex button LED indicating the status (ON/OFF).
For PROGRAM CODES with multiple Flex button selections, the volume controls ([VOL UP] and [VOL DOWN] buttons) may be used to select the next or previous item. The [SPEED] button is generally employed as a delete button to erase existing entries however, where noted, it may be used to confirm a range input. Pressing the [CONF] button returns to the 1st step of the data entry procedure for the PROGRAM CODE without storing unsaved entries.
The [SAVE] button is used to store data after entry. If there are no conflicts in the entered data, confirmation tone will be received and the data stored. If a conflict exists, error tone is provided and newly entered data are not saved. Generally, corrected data may be entered and stored without restarting the entry procedure from the 1st step.

CALL PARK Ericsson-LG

CALL PARK
Description
A user may place an active CO/IP call in a special holding location (Park Orbit) for easy access from any station in the system.
Operation iPECS Phone To park an active external call
1. Press the [TRANS] button.
2. Dial the Park Orbit.
3. Return to idle.
To retrieve a parked call
1. Lift the handset or press the [SPEAKER] button,
2-45
2. Dial the Park Orbit.
SLT
To park an active external call
1. Momentarily press the hook-switch.

Upgrading the System

Upgrading the System

You can upgrade your system software to a new release of the PARTNER ACS by using a PCMCIA card. You also can add or replace modules and add lines, pools, and extensions.
Both old and new (introduced in system Release 4.0) 5-slot carriers are compatible with PARTNER ACS R7.0.
Battery Replacement 10
The processor module uses two user-replaceable AAA alkaline batteries. These batteries provide enough power to retain the system programming settings during a power failure for 45 days to six months, depending on the freshness of the batteries. When battery power is getting low, the system displays a ChgBat W/PowerOn or ReplaceSysBat W/Power On message on the top line of display telephones at extensions 10 and 11 in place of the default day/date/time message. Users at these extensions should notify the System Administrator when they see this message. You should replace the batteries within 45 days of seeing the message.
The message may flicker on and off as the batteries approach the low-power threshold.
CAUTION:
Do not turn off the power or remove the processor module before replacing the batteries! If you do, all settings for system and telephone programming revert to the factory settings. If you have a Backup/Restore PC Card, do a backup before changing the batteries.
PARTNER ACS Release 7.0 supports the PARTNER Remote Access PC Card, which allows you to program the system remotely and perform backup and restore functions.

Voicemail Email Integration

Voicemail Email Integration
Voicemail Email features with Embedded Voicemail is supported. This uses the IP Office system's SMTP settings to send messages to the customer's email server. That server then forwards those messages into the user email mailboxes.
Once enabled, users can select to have a email alert about each new voicemail message or to have the voicemail message forwarded to their email mailbox. In addition when listening to a message in their voicemail mailbox they can forward it to their email mailbox.
· Warning
The sending of .wav files across a network creates a high loading on the network and networks servers. A one- minute message requires a 1MB .wav file.
1.Note that changing the IP Office's SMTP settings requires the system to be restarted.
2.Obtain details of the customer's SMTP email server. You can configure a user account on that server in order for it to accept and relay emails from the IP Office.
3.Using IP Office Manager, receive the IP Office system configuration. 4.Select System and then the SMTP tab.
5.Enter the details to match the customer SMTP server.
· IP Address
The IP address of the customer's SMTP server. If not on the same subnet as the IP Office LAN, an IP route must also be added.
· Port
The SMTP listening port of the server. The default is 25.
· Email From Address
This is the address that will be used by the IP Office. Some servers will only relay messages from recognized full addresses or addresses in the same domain.
· Server Requires Authentication
If the server requires a user account to receive and send emails, enter the details of an account configured on that server for use by the IP Office.
6.For each user, select User | Voicemail.
7.In the Voicemail Email field enter the user's email address.
8.Using the radio button, select the type of Voicemail Email function alerts for the user.
· Off
Don't send email alerts for new messages. Users can select this themselves by dialing *03.
· Copy
Send a copy of each new message received to the user's email address. User's cannot select this mode themselves.
· Forward
Forward each new message received to the user's email address, deleting the message from their mailbox. Users can select this themselves by dialing *01.
· Alert
Send an email alert for each new message received. Users can select this themselves by dialing *02.

Amanda

Amanda
Amanda is an automated telephone attendant and voice messaging system designed especially for ease of use and flexibility. To you and the people who call you, Amanda is also a voice on the telephone guiding you to people, services, and messages.
Each user of the Amanda system has both a mailbox and a telephone extension number. These are always the same number. The extension number is what Amanda dials to reach you when you have a call. The mailbox identifies a record in Amanda’s database. The record contains fields that define how Amanda processes your calls. You can change the contents of these fields from a touch-tone telephone using a series of menus. For example, you may turn Do Not Disturb on and off during a hectic day.
In addition to these fields, each mailbox has greetings that you record. For example, when you cannot answer the telephone, callers hear a greeting that asks them to leave a message.
Your mailbox is configured to ring a telephone extension and record messages from callers. You periodically check your mailbox for messages, or you may be notified that a message exists in a variety of ways.

System Start Up

System Start Up 3
SECTION 1 SYSTEM START UP
1.1 Before Starting Up the System
Before starting up the system, make sure:
• KSU(s) are installed correctly.
• All extensions are cabled correctly.
• All earth ground and PSTN Trunks are cabled correctly.
• All PCBs are configured, equipped, and secured correctly.
• AC power cord is cabled correctly.
• At least one display type MultilineTelephone is connected to the system. (for Programming)
• Pull out the Lithium battery protection sheet, before starting up the system.
Lithium battery protection sheet
Figure 3-1 Lithium Battery Protection Sheet
• If Expansion KSU(s) are installed, turn the power on/off in the order of Expansion 2 KSU, Expansion 1 KSU and then Main KS

Ancillary Device Connection

Ancillary Device Connection
Description
Ancillary Device Connection allows installation of selected peripheral (ancillary) devices to a Multiline Terminal. This feature enhances peripheral device objectives.
Allows a SL1100 Multiline Terminal user to connect ancillary devices such as Wireless Headset.

Soft Key

Soft Key
Following is an alphabetical index of the InMail soft keys (available for SL1100 Multiline terminals and IP Multiline terminal only). Also see the Operation section of each feature.
Soft Key Feature Definition
AM Message Notification When programming a Message Notification time, press to indicate that the entered time is AM.
Annc Announcement Mailbox
System Administrator Press to access the Announce-
ment Mailbox message options.
ATime Auto Time Stamp Press to select the Auto Time Stamp feature.
Back Answer Schedule Override Announcement Mailbox Auto Attendant Direct to Voice Mail Auto Time Stamp Call Routing Mailbox Erasing All Messages Exiting a Mailbox Greeting Instruction Menu
Listening to Messages Mailbox Name Mailbox Security Code Delete Main Menu Message Forward Message Notification Message Reply Record and Send a Message Security Code
Press to go back to the previous menu level or exit your mailbox.
CallH Auto Attendant Direct to Voice Mail Press to access Call Handling options, Auto Attendant Direct to Voice Mail and Paging.
Cancel Announcement Mailbox
Auto Attendant Direct to Voice Mail Call Routing Mailbox Conversation Record Greeting
Instruction Menu Mailbox Name Message Forward Message Reply Record and Send a Message
Press to erase the current message, name, or greeting.
Cncl
Chnge Message Notification Change the Message Notifica- tion setup.
Code Security Code Press to access the Security Code options.
Cont Message Forward
Message Reply Record and Send a Message Press to begin recording.
Del Announcement Mailbox
Auto Attendant Direct to Voice Mail Call Routing Mailbox Greeting Instruction Menu
Listening to Messages Mailbox Name Message Delete Security Code
Press to delete the currently accessed message, name, greeting, or Security Code.
DList Distribution List
System Administrator Press to access Distribution List
setup.
Done Announcement Mailbox
Auto Attendant Direct to Voice Mail Call Routing Mailbox Conversation Record Greeting
Instruction Menu Mailbox Name Message Forward Message Reply Record and Send a Message
Press to exit the recording mode while recording a message, name, or greeting.
DVM Auto Attendant Direct to Voice Mail Press to enable or disable the Auto Att. Do Not Disturb option.

Alarm Reports

Alarm Reports
Version 2.0 or higher software provides;
•Enable to send the Alarm Report by E-Mail to SMTP client
•Enable to send the DIMLAST data, DIMDUMP data from the system automatically
Description
The system logs various errors and reports information about the operation that can be used to determine the cause of a problem. The system can indicate several errors on the Multiline Terminal display, output to a Maintenance CF card on the CPU, or be downloaded in PCPro. The report data also can be sent via e-mail.
DSP Resource Full
When attempting a call requiring an IP to TDM conversion and no DSP resource is available, the system displays a message on Multiline Terminal and can generate an alarm via the Alarm Report.
IP Collision
System is able to detect another device on the same subnet having an IP address that conflicts with those assigned to the CPU, VoIPDB and DSP resource to make troubleshooting easy when IP packets are not sent.
Alarm Report
The Alarm Reports indicate:
•System start-up/upgrade date and time.
•Unit communication error with date and time and the restoration date and time.
•Date and time a unit was removed from the system.
•Date and time an extension was disconnected from the system.
•Date and time of any system data change

Door Box

Door Box
Description
The Door Box is a self-contained Intercom unit typically used to monitor an entrance door. A visitor at the door can press the Door Box call button (like a door bell). The Door Box then sends chime tones to all extensions programmed to receive chimes. To answer the chime, the called extension user just lifts the handset. This lets the extension user talk to the visitor at the Door Box. The Door Box is convenient to have at a delivery entrance, for example. It is not necessary to have company personnel monitor the delivery entrance; they answer the Door Box chimes instead. Any number of system extensions can receive Door Box chime tones.
Each Door Box has a pair of normally open relay contacts that can connect to an electric door strike. Use these contacts to remotely control the entrance door. After answering the Door Box chimes, a Multiline Terminal user can press Flash key to activate the Door Box contacts. This in turn releases the electric strike on the entrance door. The device connected to the Door Box contacts cannot exceed the contact ratings shown in the following table:

ISDN Compatibility

ISDN Compatibility
Description
ISDN-PRI
•DID Line Service
ISDN-PRI (Integrated Service Digital Network - Primary Rate Interface) is a Public Switched Telephone Network (PSTN) service that provides 23 B channels and a single D channel (23B+1D) for trunking. Caller ID indication displays the calling party telephone number on the LCD of the Multiline Terminal for CO incoming calls. This interface provides voice communication path only. I ISDN - PRI Features
When configured for DID Line Service, the trunks emulate Loop Start or Ground Start trunks for outgoing calls and DID trunks for incoming calls.
•Calling Line Identification Presentation (CLIP)
PRG 10-03-05: ETU Configuration - CLIP Information Announcement, will allow the Calling Party Number IE in the Setup Message for a call when placed out an ISDN Trunk.
•Calling Party Number (CPN) Presentation from Station
Calling Party Number (CPN) Presentation from Station allows each unique station or virtual extension 10-digit number (representing the DID number of the originating station) to be sent out over the ISDN Network, if it is programmed. If there is no Extension Calling Number assigned, the system will send the calling number for the ISDN trunk. If both the extension and trunk information is programmed, the extension information is sent as it takes priority.
•Calling Party Name:
If programmed, Calling Party Name allows the station name to be sent out over the ISDN network. A system wide name can be programmed to be sent over the network or the name can be defined on a per station basis. If both are programmed, the system wide name takes priority over the station name.
•SMDR Includes Dialed Number
The SMDR report can optionally print the trunk name (entered in system programming) or the number the incoming caller dialed (i.e., the dialed ISDN digits). This gives you the option of analyzing the SMDR report based on the number your callers dial. (This option also applies to a DID trunk as well.)
•Display Shows Why Caller ID is Not Available
With Caller ID enabled, the system provides information for ISDN calls that do not contain the Caller ID information. If the Caller ID information is restricted, the telephone display shows PRIVATE. If the system is not able to provide Caller ID information because the Telco information is not available, then the display shows OUT OF AREA.
Conditions
•Primary Rate Interface (PRI):
The system is compatible with ISDN Primary Rate Interface (PRI) services. PRI services currently supported include:
-Basic PRI Call Control (BCC)
-Display of incoming caller’s name and number when allowed by Telco
-Routing in the system based on the number the caller dialed
-ISDN maintenance functions (such as In Service/Out of Service Messaging)
-Speech and 3.1 KHz audio
PRI capability requires the installation of 1PRIU-C1. Each PRI circuit provides 24 PRI channels (23B + D) 4 with 64K Clear Channel response. The T1/PRI Interface uses a single slot. When installed, the T1/PRI Interface uses the first block of 24 consecutiv

ACD (Automatic Call Distribution) Group & Supervisor

ACD (Automatic Call Distribution) Group & Supervisor Description
A station can be assigned as the ACD Group Supervisor. A station can be the ACD Supervisor for one or multiple ACD groups. The ACD Supervisor receives status information for the group indicating queued calls and time. When a call is queued to a group for longer than a queue time or when a predefined number of calls are queued, the supervisor's LCD will indicate the number of calls in queue, and the queued time for the longest queue.
The ACD Supervisor can control and can change overflow destination and timing as well as the On/Off duty status of ACD group members. In addition, he ACD Supervisor can monitor an agent’s call and, if desired, the ACD Supervisor can record an agent’s call to the VMIB using Two-way record (refer to Ref. C).
Operation
ACD Group Agents
ACD Group Agents are logged into each ACD group as a Group Member in the system database. Agents use the Agent Duty code or a Flexible button assigned as a UCD-DND button to go ‘On’ and ‘Off’ duty for a specified ACD Group.
To toggle Agent On/Off Duty,
1. Press the [TRANS/PGM] button.
2. Dial ‘8 7’, the Agent Duty (UCD-DND) code.
3. Dial the ACD Group Number.
4. Press the [HOLD/SAVE] button.
Once ‘On-duty’, the LCD display of the DKT will display the queued call count for the ACD Group. Also, if the station has a DSS button for the ACD Group, the LED will flash to indicate the number of calls queued.
 Off 0 calls in queue
 60 ipm 1 to 3 calls in queue  120ipm 4 to 6 calls in queue  240 ipm 7 or more calls in queue
Supervisor
ACD Group Supervisors are logged into an ACD Group as a Group Member and separately assigned as the ACD Group Supervisor. The ACD Supervisor goes ‘On’ and ‘Off’ duty as with any ACD Group Agent. In addition, the ACD Group Supervisor must have a Flexible button assigned as an [ACD Group] button for each supervised ACD Group.
An ACD Supervisor can monitor calls of ACD Agents in the supervised ACD Group. The Supervisor enters the conversation in a ‘Listen-only’ mode (Mic is muted) but can join the conference with the Mute button. A warning tone can be configured to ‘beep’ when the ACD Supervisor monitors a call.
105

Do Not Disturb Override

Do Not Disturb Override
Easily override a co-worker’s Do Not Disturb.
Description
DSX | Features |
Do Not Disturb Override lets an extension user override another extension’s Do Not Disturb. This allows a priority employee (such as a supervisor or executive) to get through to a co-worker right away while the co-worker’s phone is in Do Not Disturb. DND Override is available to all extensions that have DND Override set in their Class of Service. It is also available to any extension that has a Hotline key for a co-worker, even without the Class of Service option enabled.

FAX over IP

FAX over IP
General Description
This feature allows the system to transmit facsimile communications over IP network, via Local Area Networks (LAN) and corporate Wide Area Network (WAN).
Since PBX regards facsimile equipment as one of ordinary telephones, VoIPDB are required for facsimile uses over IP network same as legacy stations. The facsimile transmission procedure (G.711/G.726 pass-through or T.38 (UDPTL)) is supported with VoIPDB/SIP.
The following figure shows a typical configuration of facsimile use on Peer-to-Peer CCIS network.

SL1100 hardware

Expansion Slot 2 4 6
System Maximum Port 58 116 156 1KSU: 084M+PRI+080E+4COIDBx2 2KSU: (084M+PRI+080E)x2+4COIDBx4 3KSU: 084Mx3+PRIx2+080Ex4+4COIDBx7
Trunk Port Max. 38 76 88 1KSU: PRIx1+4COIDB/BRIx2 2KSU: PRIx2+4COIDB/BRIx4 3KSU: PRIx2+4COIDB/BRIx7
Trunk Port Analog Trunks
(COT) 12 24 36 1KSU: 4COIDBx3 on 084M/080E/008E
2KSU: 4COIDBx6 3KSU: 4COIDBx9
BRI (T-Point) 12 24 36 1KSU: 2BRIx3 on 084M/080E/008E 2KSU: 2BRIx6 3KSU: 2BRIx9
PRI (24B/30B) 24/30 48/60 48/60 Max. one PRI/KSU Max. two PRIs/system
IPTrunk
(SIP/H.323) 16 When MEMDB is Installed
Station Port Max. 40 80 120 1KSU: 084M+080Ex2+BRIx3 2KSU: 084Mx2+080Ex4+BRIx6 3KSU: 084Mx3+080Ex6+BRIx9

Programming VoiceMail

Programming VoiceMail
You can program (customize) InMail by using the PCPro and WebPro programming applications. Limited programming can also be done from a System Administrator's mailbox.
See Programming Voice Mail for more information. Also see System Administrator Mailbox.
Maintenance
Quick Message Automated Attendant callers can dial a digit followed by an extension number to leave a message directly in a user's mailbox. See Quick Message for more information.
Automated Attendant
Record and Send a
Message A Subscriber Mailbox user can record and send a message to any other
Subscriber Mailbox. Voice Mail Recorded Help Prerecorded voice prompts guide the user through the InMail features. Voice Mail
Recording a
Conversation See Live Record. Voice Mail
Recording a Message See Record and Send a Message. Voice Mail
Recording Conversation
Beep See Live Record. Voice Mail
Recording Options See Record and Send a Message. Voice Mail Remote Log On See Log On to Voice Mail. Voice Mail Remote Greetings See Greeting. Voice Mail
Remote Message
Notification See Message Notification. Voice Mail
Remote Programming You can remotely program (customize) InMail by using the PCPro and WebPro programming applications.You can also do limited remote programming from
the System Administrator's mailbox. Maintenance
Return Call (with Caller
ID) See Make Call. Voice Mail
Rotary Dial Telephones See Automatic Routing for Rotary Dial Callers. Automated Attendant
Routing Mailbox Routing Mailboxes are used to route Automated Attendant calls. A Routing
Mailbox can be either an Announcement or Call Routing Mailbox. Automated
Attendant
Screened Transfer
Similar to telephone system screened transfers when the transferring party controls the transfer. After an Automated Attendant caller dials an extension, InMail calls (screens) the destination extension to see if the transfer can go through.
• If the destination is busy or in DND, the Automated Attendant does not extend the call and immediately provides the caller with additional options.
• If the destination is available, the Automated Attendant rings it. If the destina- tion answers, the call goes through. If the destination does not answer in a programmed time, the Automated Attendant does not extend the call and provides the caller with additional options. Also see Unscreened Transfer.
Automated Attendant
Security Code An extension user's mailbox can have a security code to protect the mailbox from unauthorized access.
See Mailbox Security Code Delete. Voice Mail
Select Listen Mode See Message Listen Mode. Voice Mail Single Digit Dialing An Automated Attendant caller can press a single key to route to an extension,
route to another destination, or use an InMail feature. Automated
Attendant Subscriber Mailbox The mailbox type normally used for telephone system extensions. Voice Mail System Administrator The extension user that has InMail system administration abilities. Maintenance System Administrator
Mailbox A Subscriber Mailbox option that enables the system administration abilities.
Also see System Administrator. Voice Mail System Re-initialization Re-initializing InMail returns all programmed options to default value. Maintenance
Time and Date After listening to a message, an extension user can dial a code to hear what time the message was sent.
Also see Auto Time Stamp.

Voice Mail

Greeting
A Subscriber Mailbox user can record a personalized greeting for their mailbox. With Multiple Greetings, the mailbox subscriber can record up to three separate greetings and make one of the three active. Callers to the user's mailbox hear the active greeting.
With Remote Greeting, an extension user can call into the Automated Atten- dant, access their mailbox, and remotely record their mailbox greeting. See Auto Attendant Direct to Voice Mail.
Voice Mail
Group (Shared) Mailbox An extension user can share a Group Mailbox with co-workers for receiving
and sending messages. Voice Mail
Guest Mailbox An outside party can have their own mailbox for receiving and sending
messages. Voice Mail
Hang Up An Automated Attendant option that immediately hangs up the outside call. Automated Attendant
Help See Getting Recorded Help. Voice Mail
Individual Trunk
Greetings See Multiple Company Greetings. Automated
Attendant
InMail Upload Download
Audio Allows the user to upload/download audio messages for greetings, etc. Voice Mail
Instruction Menu The Instruction Menu is the announcement that plays to Automated Attendant callers. Normally, the Instruction Menu provides callers with the Automated Attendant dialing options.
Automated Attendant
Language Setting This feature allows the telephone display language and the InMail mailbox
language to be changed from the telephone. Voice Mail
Leaving a Message An extension user can leave a voice message in a co-worker's mailbox if that
extension is busy, unanswered, or in Do Not Disturb. Voice Mail
Leaving a Message at a
Busy/DND Extension See Leaving a Message. Voice Mail
Leave a Quick Message See Quick Message. Automated Attendant
Listening Options See Listening to Messages. Voice Mail
Listening to Messages While or after listening to a message, an extension user has many message
handling options from which to choose. Voice Mail
Live Monitor
Live Monitor lets Voice Mail screen calls, just like an answering machine at home. When activated, the extension's incoming calls route to the user's subscriber mailbox. The Live Monitor feature is supported for External and Internal calls. Once the mailbox answers, the user hears the caller's incoming message.
Voice Mail
Live Record Allows an extension user to record an active call as a message in their mailbox. InMail broadcasts a beep and a voice prompt to the caller as Live
Record begins. Voice Mail
Local Notification See Message Notification. Voice Mail
Log On to Voice Mail
An extension user can press a key to log on to access their InMail mailbox. With Remote Logon, an employee calling through the Automated Attendant can dial a single digit followed by their own mailbox number to remotely log on to their mailbox.
Voice Mail
Mailbox Announcement Message
The Mailbox Announcement Message is recorded by the System Adminis- trator, and plays to each subscriber when they log on to their mailbox. The message plays at each log on until it expires, is deleted, or made inactive by the System Administrator.
Administration
Mailbox Greeting See Greeting. Voice Mail Mailbox Logon See Log On to Voice Mail. Voice Mail
Mailbox Name A mailbox caller can hear the extension user's recorded name instead of their
mailbox number. Voice Mail
Mailbox Options Menu Sub-menu of a subscriber's Main Menu that provides access to the Auto Time
Stamp, Mailbox Security Code Delete, and Message Notification features. Voice Mail

InMail

Administrator Security
Code Control See Security Code. Voice Mail
Alternate Next Call
Routing Mailbox See Next Call Routing Mailbox. Voice Mail
Announcement Mailbox Mailbox that allows a recorded greeting to play to callers. Automated Attendant
Announcement Message The message that the System Administrator records for a specific Announce-
ment Mailbox. Automated
Attendant
Answer Table Determines how the Automated Attendant answers outside calls on each trunk, according to the time of the day and day of the week that the call is ringing.
Automated Attendant
Answering Schedule
Override Enables an alternate greeting and alternate dialing options for callers. Automated
Attendant
Auto Attendant Direct to Voice Mail
Auto Attendant Direct to Voice Mail sends Automated Attendant calls directly to Automated an extension user's mailbox. Their phone does not ring for calls from the Auto- mated Attendant.
See Greeting in this glossary.
Attendant
Auto Erase or Save When a mailbox user completely listens to a new message and then exits their
mailbox, InMail either automatically archives or deletes the message. Voice Mail
Auto Time Stamp After a user listens to a message, InMail can optionally announce the time and
date the message was left. Voice Mail
Automated Attendant The Automated Attendant can automatically answer the telephone system incoming calls, play an Instruction Menu message, and provide dialing options to callers.
Automated Attendant
Automated Attendant Transfer
While on a trunk call, an extension user can transfer the trunk call to the Auto- mated Attendant so the caller can use the Automated Attendant dialing
options. Voice Mail
Automatic Access to VM by Caller ID
InMail mailbox can be associated with a specific caller ID (CID) number. When the CID number is presented to the InMail it will automatically log the user into
their mailbox. Voice Mail
Automatic Call Routing
to a Mailbox See Go to a Mailbox in this glossary. Automated
Attendant
Automatic Message
Erase/Save See Auto Erase or Save in this glossary. Voice Mail
Automatic Routing for
Rotary Dial Callers If an Automated Attendant caller does not dial any digits, InMail automatically
routes them to a specified option (such as the operator or a mailbox). Automated

Warning tones

The system can broadcast warning tones to a trunk caller, warning the caller that he has been on the call too long. If he chooses, the caller can disregard the tones and continue talking. The outside caller does not hear the warning tones. Warning tones do not occur for Intercom calls and most incoming trunk calls. DISA trunks can also have warning tones. Warning tones are not available to analog Single Line Terminal (SLT) users.
W There are two types of warning tones: Alarm Tone 1 and Alarm Tone 2. Alarm Tone 1 is the first set of tones that occur after the user initially places a trunk call. Alarm Tone 2 broadcasts periodically after Alarm Tone 1 as a continued reminder. Each alarm tone consists of three short beeps.
If programmed, DISA calls are disconnected unless the continue code is entered by the user. With the Long Conversation Cutoff feature, incoming or outgoing central office calls can also be disconnected.
Warning Tone for DISA Callers
For DISA callers, with this feature enabled, the warning tone timer begins when an incoming DISA call places an outgoing call and either the inter-digit timer expires or the outgoing call is answered.
If an outside call is transferred to forwarded off-premise using an outside trunk, the warning tone timer

Toshiba

access codes CO line, 33 feature, 21 speed dial, 37 access DISA, 4 account codes revision, 29 verified, 31 auto attendant, 1, 4 announcement recording
recommendations, 4 CO line assignments, 2 dialing plan, 3 menu, 3 system dialing plan, 3 automatic station relocation physically, 13
swap buttons/features, 13
C
CAMA trunk calls, 6 CO lines, 4
access codes, 33
auto attendant assignments, 2 codesfeature access, 21
paging group, 35 speed dial access, 37 toll restriction override, 29 traveling class, 29 verified account, 31
D
day 2 mode, 8 day mode, 8 day/day 2/night modes, 2, 4, 8 default DNs, 3
dial pad key equivalents, 16 directories, 39
speed dial memo, 40 user name/number, 39

DISAaccess, 4
outgoing calls, 3 security code, 1, 4 cancel, 5 enter/change, 5, 6
E
emergency 911 calls, 6 equipment notes, iii
F
feature access codes, 21 flash button, 24
LLCD name/number information
clear, 18 enter, 17 erase, 18 other stations/devices enter name/number, 18 erase name/number, 20 linked speed dial example, 26
Mmemos, 15
messages, 15
multiple announcement machine, 4
Nnames, 15
night mode, 8
night transfer, 1, 2, 8 enable/disable, 8 lock mode, 9 lock/unlock password, 8
assign/change, 9 routing (ringing) patterns, 8 non-LCD telephones, 17
P
page button, 19, 28, 29 paging group codes, 35 primary announcement, 2, 4
R
records, 39
telephone location, 41 relocation by special dial, 14
S
secondary announcement, 2, 4 setting date/time/day, 1, 10, 11 soft keys, 1, 12, 18, 27 to turn off, 12 to turn on, 12 speed dial, 20, 26 access codes, 37 accessing stored number, 21 advanced features, 21 checking a number, 29 clearing an entry, 21 dialing, 28 flash, 24 long pause, 24

number linking, 24 example, 26 store CO line access code, 26 use new link, 26 pause and flash storage, 24 storing a feature, 20 storing system speed dial number in a
system speed dial code, 20 system numbers, 20 view, 28
station (directory) number, 3 station relocation, 1, 12 automatically relocate, 13 relocation by special dial, 14 systemauto attendant dialing plan, 3
messages, names and memos, 1, 15 setting parameters, 1
speed dial memo numbers, 1 speed dial numbers, 20
Ttelephone location record, 27
tenant service, 8
toll restriction override, 1, 29 add/delete/change, 29 by system speed dial, 26 delete, 30
traveling class codes, 1, 29 add/delete/change, 29 delete, 30
Uuser LCD name/number display, 17
clear, 18 erase, 18 other stations/devices enter name/number display, 18 erase name/number display, 20 write, 15
V
verified account codes, 1, 31 add/change, 31 delete, 32
W
writeLCD memos, 15
LCD messages, 15
LCD name/number displays, 15

NEC systems

PBX SYSTEM DESIGNATION
PBX system is usually designated as “PBX” or “system”.
When we must draw a clear line between the PBX systems, they are designated as follows.
SV8300 : UNIVERGE SV8300 SV8500 : UNIVERGE SV8500 SV7000 : UNIVERGE SV7000 2000 IPS: NEAX 2000 IPS INTERNET PROTOCOL SERVER 2400 IPX: NEAX 2400 IPX Internet Protocol eXchange
ATTENDANT CONSOLE NAME
Attendant Console is usually designated as “Attendant Console”.

NEC’s UNIVERGE Desktop IP

The right phone for every work situation
With the speed of business today, the importance of the desktop phone has never been greater. Today’s employee requires an accessible communication tool at any location in order to be as effi cient and productive as possible. But, many businesses and employees have not taken advantage of the enhanced capabilities offered by today’s next-generation phones. NEC’s UNIVERGE Desktop IP and Digital terminals and handsets are the answer.
With a wide range of customizable features and a modular design, this terminal can help meet the communications needs of any workplace.
Freedom of choice
UNIVERGE Terminals give you the freedom to tailor your platform and telephony applications to meet your business’s evolving needs. Whether your business is just getting started or is already rapidly growing, NEC provides the right solutions.
Personalized terminals to meet your specifi c requirements
NEC’s innovative terminal design delivers maximum deployment fl exibility. Modularity allows for multiple combinations to fi t any business niche or personalization requirement, from the front desk, to the conference room, to knowledge workers, to remote workers, to executives.
Easy-to-use, intuitive interfaces
NEC’s terminal interfaces are designed to improve the overall user experience. The NEC terminal interfaces are designed
to be intuitive, no extensive training is needed. Global icons indicate status at a glance including notifi cation of new voice or instant messages, missed calls, the telephone user’s current presence status, and the device’s current data protection mode.
Personal, system and corporate directories
Quick access to directories; each entry in the directory is searchable, and a call can be placed from the searched entry. Name display on incoming calls, if the Caller-ID matches the registered phone number with the entry in the directory.
Good reasons to choose UNIVERGE® terminals
• Modular construction - the interchangeable design provides easy and cost-effective upgrades, helping to future-proof this businesses investment
• Customizable design - choose from a range of add-on line key modules, faceplates, LCDs, keypads and even printable side panels
• Customizable function keys - can be adapted to the exact individual requirements of a business
• User-friendly interface - little or no staff training required

The UNIVERGE®

The UNIVERGE® SV8100 is a unique communication solution for up to 500 extensions. It improves business performance signifi cantly by making an entire workforce more reachable wherever they are based.

Part of the UNIVERGE®360 portfolio, the SV8100 creates ‘360-degree communication’ encompassing fi xed, mobile and converged communication such as e-mail, Presence and instant messaging. Executives have real-time access to a full circle view of their business; managers easily communicate with team members and supervisors; sales people have immediate access to the data and resources they need to do their jobs anywhere they are.
In short, it makes Unifi ed Communications a reality.
Why NEC?
• A leading global enterprise telephony solution provider
• Empowering our customers through over 100 years of experience in IT and Networking
• Spanning the full spectrum of ICT products and solutions
• Investing over 2,7 billion Euro in research and development every year
• Employing more than 150,000 people worldwide
• The only global company in the world’s top 5 in both computers and communications
• Unsurpassed technical support and logistics
• A reliable, stable partner with the mission to realize an information society friendly to humans and the earth

T1 PRI parameters

Trunk Parameters
· Switch Type: Default = NI2
Options 4ESS, 5ESS, DMS100 and NI2.
· Provider: Default = Local Telco
Select the PSTN service provider (AT&T, Sprint, WorldCom or Local Telco). When set to AT&T, an additional AT
& T Provider Setup menu can be accessed from the menu. 90
· Test Number:
Used to remember the external telephone number of this line to assist with loop-back testing. For information only.
· Send Redirecting Number: Default = Off
· Clock Quality: Default = Network
Leave as Network unless advised otherwise by Avaya.
· Framing: Default = ESF
Selects the type of signal framing used (ESF or D4).
· CRC Checking: Default = On Turns CRC on or off.
· Zero Suppression: Default = B8ZS
Selects the method of zero suppression used (B8ZS or AMI ZCS).
· CSU Operation:
Tick this field to enable the T1 line to respond to loop-back requests from the line.
· Line Signaling: Default = CPE
The field can be set to either CPE (Customer Premises Equipment) or CO (Central Office). This field should normally be left at its default of CPE. The setting CO is normally only used in lab back-to-back testing.
· Haul Length: Default = 0-115 feet
Sets the line length to a specific distance.
· Channel Unit: Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided by the Central Office. The options are Foreign Exchange, Special Access or Normal

Electra Elite IPK Multiline Terminals Hands free

Full Duplex Handsfree
___________________________________________________________________________________
FEATURE DESCRIPTION
The HF-R Unit is an add-on device to the Electra Elite IPK Multiline Terminals that provides a full duplex speakerphone for small conference rooms. An external microphone is also provided that has a locking push-to-mute control button to turn the microphone off.
SYSTEM AVAILABILITY
Terminal Type:
All Electra Elite IPK Multiline Terminals with an HF-R Unit installed
Required Software: R1500 or higher
Required Components: HF-R Unit
AC-R Unit (AC Adapter)
OPERATING PROCEDURES
To use the HF-R using an Electra Elite IPK Multiline Terminal:
1.Press E , and make an internal or external call.
2.When muting is desired, press the Mute key on the external microphone. To cancel muting, press the Mute button again.

Attenfant Station

Attendant Station Outgoing Lockout allows an Attendant Position with an Attendant Add-On Console to be used to set a predetermined Code Restriction Class Assignment at any station assigned on the Attendant Add-On Console. This allows an Attendant to set/reset a restriction to allow/deny an outgoing call.

ISDN (PRI)

ISDN (PRI), Answering Calls
Primary Rate Interface (PRI), Answering Calls
Description
The system provides flexible routing of incoming PRI calls to help meet the exact site requirements. This allows PRI calls to ring and be answered at any combination of system extensions. Many of the options available to incoming analog trunk calls are also available to incoming PRI calls.
Delayed Ringing
Extensions in a Ring Group can have delayed ringing for PRI trunks - just like other types of trunks. If the PRI trunk is not answered at its original destination, it rings the DIL No Answer Ring Group. This could, for example, help a secretary that covers calls for their boss. If the boss doesn’t answer the call, it rings the sec- retary’s phone after a programmable time.
Calling Name Delivery
If provided by the telco, and depending on the version of your system software, the system can support call- ing name delivery in the Facility Information Element. With this information available, display telephone users can see the name of the calling party.
Caller ID
With Caller ID enabled, the system will provide information for ISDN calls that do not contain the Caller ID information. If the Caller ID information is restricted, the telephone display will show “PRIVATE”. If the system is not able to provide Caller ID information because the telco information is not available, then the display will show “OUT OF AREA”.
SMDR Includes Dialed Number
The SMDR report can optionally print the trunk’s name (entered in system programming) or the number the incoming caller dialed (i.e., the dialed ISDN digits). This gives you the option of analyzing the SMDR report based on the number your callers dial. (This option also applies to a DID trunk as well.)
Calling Party Number Notification
The system can provide calling party number notification for outgoing ISDN calls. When a call is made on an ISDN line by an extension, the system will send the identification for the extension placing the call, if it’s programmed. If there is no Extension Calling Number assigned, the system will send the calling number for the ISDN trunk. If both the extension and trunk information is programmed, the extension information will be sent as it takes priority.
When the option for calling party subaddress is on, the extension number will be sent as the subaddress infor- mation. Both the calling party number and calling party subaddress are sent in a SETUP message as the calling party information element and a calling party subaddress information element. Allow the system to send the subaddress by setting the following programs: 10-03-05=1, 15-01-04=1, 20-08-13=1, 21-13-01=enter number to be sent.
Calling Line Identification Presentation
CLIP display available with software 1.02+.
A Class of Service option has been added which can be used to allow the Calling Party Number IE in the Setup Message.
Calling Party Allowed or Prevented for Extension
Calling Party allowed for extension with software 1.04+.
The system allows the Calling Party Number for outgoing ISDN calls based on the extension’s set up in Pro- gram 15-01-04 : Basic Extension Data Setup - ISDN Caller ID. If this option is to be enabled, then

PcPro WebPro

In addition to the keyset programming, the Aspire system provides the ability to use a PC to access system programming. The Windows-based PCPro and the HTML-based WebPro allows you to:
● Edit the telephone system programming options from a remote location.
The PCPro application requires changes to be uploaded to the system before they take effect. The WebPro application applies the changes as soon as the APPLY or OK icon is clicked. ● Access system maintenance functions (like reports and tests).
In addition, PCPro allows you to:
● Save your programming to your PC’s hard disk - then upload it via a LAN (Local Area Network), PPP, serial or USB connection.
PPP is a protocol that allows a computer to use a regular telephone line and a serial interface (modem) to make TCP/IP connections.
Local programming is possible using the LAN, USB or serial connection (the USB connection is not avail- able on the Aspire S) (CTA and CTU adapters cannot be used). Remote programming is possible using the dial- up serial port connection or LAN connection provided there is internet/network access to the IP address of the CPU.
● Download the existing programming in the telephone system via a LAN, PPP, serial or USB connection - and save it to your PC’s hard disk.
● Set up a default database with the settings you use most often.
● Create a unique database for each phone system you have installed. Since you save the site-specific data to your PC’s hard disk, you can easily retrieve a customer’s programming if something goes wrong.
Aspire S/Aspire System Requirements
● Aspire S system software 2.50 or higher ● PCPro: NTCPU LAN, Serial or USB Connection or any version of Aspire system software to PC (USB on Aspire M/L/XL Only)
● Aspire S: ENTU for LAN Connection WebPro: TCP/IP via LAN Connection to PC
PCPro PC Requirements
● CPU: Pentium II 500 MHz equivalent or higher ● 128Mb of RAM
● 20Mb of Hard Drive Space ● Monitor Resolution: 800 x 600 pixels or higher ● Mouse ● Microsoft Windows 2000/XP
● Internet Browser: Internet Explorer 6.0 or higher or ● Network Interface Card (NIC) Netscape 6.0 or higher (depending on connection type)
WebPro PC Requirements
● CPU: Pentium III 600 MHz equivalent or higher ● Monitor Resolution: 800 x 600 pixels or higher ● Mouse ● Microsoft Windows 2000/XP
● Internet Browser: Internet Explorer 6.0 or higher or ● Network Interface Card (NIC) Netscape 6.0 or higher (depending on connection type)

Hold

Hold lets an extension user put a call in a temporary waiting state. The caller on Hold hears silence or Music on Hold, not conversation in the extension user’s work area. While the call waits on Hold, the extension user may process calls or use a system feature. Calls left on Hold too long recall the extension that placed them on Hold. There are four types of Hold:
● System Hold
An outside call a user places on Hold flashes the line key (if programmed) at all other keysets. Any keyset user with the flashing line key can pick up the call.
● Exclusive Hold
When a user places a call on Exclusive Hold, only that user can pick up the call from Hold. The trunk appears busy to all other keysets that have a key for the trunk. Exclusive hold is important if a user doesn’t want a co-worker picking up their call on Hold.
● Group Hold
If a user places a call on Group Hold, another user in the Department Group can dial a code to pick up the call. This lets members of a department easily pick up each other’s calls.
● Intercom Hold
A user can place an Intercom call on Hold. The Intercom call on Hold does not indicate at any other extension.
With Automatic Hold enabled (Program 15-02-07), when the user is on an outside call using the handset, the user can press a flashing line/loop key to answer an incoming call without disconnect- ing their first call. The first caller is automatically placed on hold. This feature does not work using handsfree or when the user is on an ICM and presses a flashing line/loop key (the ICM call is disconnected).
Hold Recall to Operator
Hold Recall to Operator enhances how the system handles calls that have been left on hold too long. With Hold Recall to Operator:
● A trunk call recalls the extension that placed it on Hold after the Hold/Exclusive Hold Recall time.
● The recalling trunk will ring the extension that placed it on Hold for the Hold/Exclusive Hold Recall Callback Time.
● After the Hold/Exclusive Hold Recall Callback Time, the trunk call will ring the operator.
Hold Recall to Operator applies to trunk calls placed on System Hold, Exclusive Hold and Group Hold. It does not apply to Intercom calls.
Conditions
The called extension must lift the handset or press the SPK key before the call can be placed on hold.
Default Set

Computer Telephony Integration (CTI)

Computer Telephony Integration (CTI). It uses the Telephony Application Programming Interface (TAPI) 2.1 protocol. To allow CTI an Ethernet inter- face is present on the NTCPU.
TAPI 2.1 based CTI realises third party call control features such as ACD, Predictive Dialing, and Call Routing.

General Description
The following section assumes you have installed the Aspire TSP. This section contains further infor- mation on configuring TAPI2.1 on Windows NT Server.
The following information is provided:
• Aspire TSP Configuration
• Enabling TAPI Server
• TAPI Server User Administration

Aspire WebPro

WebPro
The WebPro software is installed on the Aspire NTCPU PCB - there is no separate software installation. This means that when the Aspire system software is updated, the WebPro software is updated as well.
Make sure the SW3 switch on the Aspire NTCPU (switch just above the serial port connector) is set to OFF (down).
The Aspire system (Program 10-21-02) and the Dial Up setting must be set to use the same baud rate (19200 by default).
Make sure the PC is connected to the serial or USB port on the Aspire NTCPU with a null modem (cross- over) cable. For LAN connections, use a straight-through cable if connected through a hub. If connected directly to the Aspire S/Aspire LAN connector, use a cross-connect cable.
The WebPro software provides a Help system if you experience difficulty in using the program. Simply press the ‘F1’ key.
1.Make sure the required cable (USB, serial, LAN) is connected from the PC to the Aspire S/Aspire system. 2.When using a LAN connection, skip to Step 5.
When using a connection other than a LAN, you must first connect to the Aspire using the dial-up connection. If not already connected, click on START  SETTINGS  NETWORK CONNECTIONS  select the dial up connection to be used to connect to the Aspire system.
3.In the window that appears, if the user name and password are acceptable, click CONNECT or DIAL depending on your connection type.
Only one person is allowed in programming mode at a time. An error message will be received if trying to log in while another user is already in programming mode.
4.If the PRE-DIAL TERMINAL SCREEN option was selected in the dial-up setting, when it appears, left click on the black area of that screen. Type AT and press ENTER. Once OK has been displayed, click on CONTINUE and wait for the computer to connect to the Aspire system.
If an OK does not appear on the screen, continue to type AT and then press enter until you get an OK on the screen.
5.Once the connection has been established, with either Internet Explorer or Netscape Navigator installed, open the internet
browser application.
6.Enter the IP address of the Aspire system (example: http://192.78.0.1). This address is selected based on the type of con- nection to the Aspire.
N 47

WebPro

WebPro
Using WebPro
The WebPro software is installed on the Aspire NTCPU PCB - there is no separate software installation. This means that when the Aspire system software is updated, the WebPro software is updated as well.
Make sure the SW3 switch on the Aspire NTCPU (switch just above the serial port connector) is set to OFF (down).
The Aspire system (Program 10-21-02) and the Dial Up setting must be set to use the same baud rate (19200 by default).
Make sure the PC is connected to the serial or USB port on the Aspire NTCPU with a null modem (cross- over) cable. For LAN connections, use a straight-through cable if connected through a hub. If connected directly to the Aspire S/Aspire LAN connector, use a cross-connect cable.
The WebPro software provides a Help system if you experience difficulty in using the program. Simply press the ‘F1’ key.
1.Make sure the required cable (USB, serial, LAN) is connected from the PC to the Aspire S/Aspire system. 2.When using a LAN connection, skip to Step 5.
When using a connection other than a LAN, you must first connect to the Aspire using the dial-up connection. If not already connected, click on START  SETTINGS  NETWORK CONNECTIONS  select the dial up connection to be used to connect to the Aspire system.
3.In the window that appears, if the user name and password are acceptable, click CONNECT or DIAL depending on your connection type.
Only one person is allowed in programming mode at a time. An error message will be received if trying to log in while another user is already in programming mode.
4.If the PRE-DIAL TERMINAL SCREEN option was selected in the dial-up setting, when it appears, left click on the black area of that screen. Type AT and press ENTER. Once OK has been displayed, click on CONTINUE and wait for the computer to connect to the Aspire system.
If an OK does not appear on the screen, continue to type AT and then press enter until you get an OK on the screen.
5.Once the connection has been established, with either Internet Explorer or Netscape Navigator installed, open the internet
browser application.
6.Enter the IP address of the Aspire system (example: http://192.78.0.1). This address is selected based on the type of con- nection to the Aspire.

Station Message Detail Recording

Station Message Detail Recording
SMDR provides a printed record of your calls.
Description
DSX | Features | 527
Station Message Detail Recording (SMDR) provides a record of the system’s outside calls. Typically, the record outputs to a customer-provided printer, terminal or SMDR data collection device. SMDR allows you to monitor the usage at each extension and line. This makes charge-back and traffc management easier. SMDR includes both incoming and outgoing calls, and can be turned off system-wide or selectively for each line.
The SMDR call record outputs when the call completes. The system assigns the SMDR record to the last extension on the call. For example, if extension 306 answers the call, talks for 20 minutes, and then transfers the call to extension 302, extension 302 “owns” the entire call record as soon as they hang up.
SMDR requires a customer-provided data collection device connected to the system’s RS-232 port. The default baud rate is 38,400. The data format is fxed at 8 data bits, no parity, with one stop bit (8N1). Connection requires:
• Adaptor P/N 1091014 to connect to the 9-pin RS-232 port on the data collection device.
• A standard 6-conductor line cord to connect the adaptor to the system’s RS-232 port.
SMDR does not buffer records when the data collection device is disconnected.
Call Duration Independent of System Clock
The duration of a call on the SMDR report is calculated independently of the system clock. This prevents changes made to the system Time and Date from inaccurately reporting the call duration after the Time and Date change. The automatic Daylight Savings Time adjustment also will not affect the call duration.

Battery Backup

Battery Backup
The system provides permanent backup of system memory.
Description
DSX
In the event of commercial AC power failure, the NAND Flash memory on the CPU PCB permanently maintains the site database. Additionally, an internal battery on the CPU provides short-term backup of the system date and time (Real Time Clock) and certain station parameters (such as the Caller ID log). The battery will hold the Real Time Clock and station parameters for up to 10-14 days. When commercial AC power is restored, the system restarts with all programming and the time and date intact.
Additional Battery Backup capability can be provided by a customer-supplied Uninterruptable Power Supply (UPS). The length of time the UPS will power the system when power fails depends on the capacity of the UPS unit. Consult with the UPS manufacturer for the specifcs. Refer to the Hardware Manual for additional details.

Voice recorder

A voltmeter is helpful in determining if your inputs are “wet loops” or dry lines. Wet loops typically carry -48V DC when measured off-hook. Wet loops are used to connect to your telecom provider, to your PBX, or to analog phone extensions. Dry lines such as headsets, radios, microphones and speakers carry audio, but no DC signaling voltage.
After you’ve determined a line type (either wet-loop or dry line), select a recording trigger: either Loop Start or VOX. Outside wet loops and inside analog extensions normally use Loop Start signaling. All dry lines use VOX signaling to trigger recording.
VOX Recording for Dry Lines
To bridge a dry line using Voice Operated Switch (VOX):
• Note: VOX mode is used for dry lines (lines that do not carry signaling voltage). Examples are handset connections, headsets, radios, microphones and speakers.
• Use a punch-down block, handset Y connector, breakout box, or RJ- 11 octopus cable to connect each line (two twisted wires) to the logger. Each of the logger inputs connects in “bridge” fashion across the existing wiring.
• Verify that the line audio level is within telecom specifications. Peak level should not exceed +3dBm. Gain controls should be set so that recording levels do not exceed 0dBm. Use Level Boost only if the input level is below -20dBm. If the VU meter peaks, the line level is too high. Reduce it using the manual gain controls, turn off Level Boost, and/or switch Auto Level on.
• Use Level Boost for handsets, microphones, and low-level signals only.
• Rename the line label by clicking on the line name. Type the new name and press Enter. For example, a handset tap or analog extension connected to LINE 01 might be renamed “Support Desk.”
• Set the line configuration mode to VOX by clicking on the line mode.

PAGING Internal,External & All Call Page

PAGING Internal,External & All Call Page
Description
A station, which is permitted to access page facilities, can connect and transmit voice announcements to any or all of the systems Internal,External Page zones. Stations are grouped into “zones” to receive pages to the zone. Stations not assigned to any zone will not receive a page including All Call pages.
A page warning tone, if assigned, will be provided to the Page Zone(s) prior to the audio connection. The user is allowed to continue the page for the specified Page Time-out timer after which the user is disconnected and the Page Zone(s) is returned to idle

Do Not Disturb Override

Do Not Disturb Override
Easily override a co-worker’s Do Not Disturb.
Description

Do Not Disturb Override lets an extension user override another extension’s Do Not Disturb. This allows a priority employee (such as a supervisor or executive) to get through to a co-worker right away while the co-worker’s phone is in Do Not Disturb. DND Override is available to all extensions that have DND Override set in their Class of Service. It is also available to any extension that has a Hotline key for a co-worker, even without the Class of Service option enabled.

Automatic System Answer Button

Automatic System Answer Button
Use this feature to program a button to turn Automatic System Answer (ASA) on and off. This feature helps the operator answer calls during busy periods.
Considerations 7
■ This feature is available only on the system telephone at extension 10.
■ Program an Automatic System Answer Button on a button with lights on the system display telephone at extension 10. (This feature is not supported on a button without lights.)
■ The Automatic System Answer Button returns to the status (on/off) it was in before a power failure occurred or System Reset was used

Door phone

Communication - In Noisier Locations

E-30-PT/E-30-PT-EWP Brushed 316 Stainless Steel (shown in optional VE-5x5)
The E-30-PT is designed to provide quick and reliable communication in noisier areas. The mic sensitivity is set to a low level until the TALK button is pressed then it is raised to a normal level. In this way, the E- 30-PT assures that the called party’s voice will be broadcast over the speaker. In applications where the background noise can be loud- er than the person calling, a handset type phone is recommended.
The E-30-PT features non-volatile memory, a built in dialer, and intelli- gent call progress detection for automatic hang-up when the call is completed. The E-30-PT can be programmed to dial up to 5 different numbers on ring no answer or busy and can be configured to dial these numbers until answered.
The E-30-PT-EWP shares all of the features of the E-30-PT in addition to Enhanced Weather Protection (EWP) for outdoor installations where the unit is exposed to precipita- tion or condensation. EWP products feature foam rubber gaskets and boots, silicon sealed connections, gel-filled butt connectors, as well as urethane or thermal plastic potted circuit boards with internally sealed, field-adjustable trim pots and DIP switches for easy on-site programming.
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Non-ADA Hot-Line Phones for:
• Terminals• Stadiums
• Parking lots/ramps• Convention centers
• ATM machines
Gate and Door Entry Phones for:
• Business lobbies
• Vehicular and pedestrian gates
• Residences
CAUTION - When installing on an analog extension of a phone system: Some phone systems do not conform to analog telecom standards and might not be compatible with the E-30-PT phones. For a detailed description of the telephone line specifications required for any of the E-30-PT phones, see DOD# 869.
SSppeecciiffiiccaattiioonnss
Power: Telephone line powered. Minimum 24V DC talk battery voltage, with a minimum loop current of 20mA loop (room temp), or 25mA (extended cold temp range). Loop current may be boosted on low current lines with a Viking Model TBB-1B talk battery booster (DOD# 632). Minimum Ring Voltage: 90VAC RMS Dimensions: Overall-127mm x 127mm x 57mm (5” x 5” x 2.25”), Plastic Electrical Box-102mm x 102mm x 54mm (4” x 4” x 2.12”) Shipping Weight: 1 Kg (2.2 lbs)
Operating Temperature: -26°C to 54°C (-15°F to 130°F) Humidity - E-30-PT: 5% to 95% non-condensing Humidity - E-30-PT-EWP: Up to 100% condensing Connections - E-30-PT: RJ11 jack
Connections - E-30-PT-EWP: Gel-filled butt connectors
• Vandal Resistant Features: 14 gauge louvered 316 stainless steel faceplate with permanent laser etched graphics, speaker/ mic screen, heavy duty metal keypad and “CALL” button and hex drive mounting screws
• Weather Resistant Features: Marine grade 316 stainless steel faceplate, screws and and push button switch. Switch internally sealed per IP67. Mylar speaker. Self-draining mic mount. Faceplate, mic and speaker gaskets. Weather resistant powder paint on optional VE-5x5 (DOD# 424).
• E-30-PT-EWP is designed to meet IP66 Ingress Protection Rating (see DOD# 859 for more information)
• Push to talk button
• Telephone line powered
• Non-volatile E2 memory (no batteries required)
• Programmable to dial up to 5 numbers on busy or ring no answer
• Red off-hook LED indicator
• Volume adjustments for microphone and speaker
• Advanced call progress detection: disconnects on busy signal, return to dial tone, CPC, reorder tone, maximum call time out and programmable silence time out
• Selectable auto-answer feature for monitoring
• Selectable push button disconnect
• Extended temperature range (-15°F to 130°F)
• Flush mountable using included plastic rough-in box
• Optional VE-5x5 surface mount back box (DOD# 4

Door entry Telephone

Add a Viking Entry Phone to an
Existing Phone Line
The C-200 allows single line telephones or a telephone system to share a phone line with a single Viking entry phone. Tenants may answer an entry phone call and converse with the visitor.
The C-200 provides a “Call Waiting” tone when the phone line is in use.
Tenants may call out to the entry phone for mon- itoring purposes. Auxiliary contacts are provided to operate a doorbell, or activate a camera, lights, etc.

If additional features (such as doorstrike, multiple entry phones with caller ID, and key- less entry) are needed, use the Viking model C-200

AUTOMATIC PRIVACY

AUTOMATIC PRIVACY
Description
Privacy is insured on all communications in the system. If desired, the customer may elect to disable the Automatic Privacy feature, allowing another station to join in an existing external conversation uninvited. In such a case, a conference is established.
Operation iPECS Phone To intrude into a call when Privacy is disabled
1. Press a busy (lit steady) individual {CO}/{IP} access button, user connected to the call with existing internal station user.
Conditions
1. With Automatic Privacy disabled, privacy is still assured on all intercom and conference calls.
2. To override privacy, Privacy must be disabled and the intruding station must have Override enabled as well as a direct appearance for the desired {CO}/{IP} line.
3. Only one station can intrude on an active call.
4. An intrusion tone can be provided to the call indicating another station has accessed the line.
5. If either internal party presses another {CO}/{IP}, a {DSS}, {PAGE}, [CONF] or other conflicting button, the party is removed from the “Conference” and must press the {CO}/{IP} button again to reenter the conversation.

Barge-in

Barge-In
Description
Barge-In permits an extension user to break into another extension user’s established call, including Conference calls. This sets up a Conference-type conversation between the intruding extension and the parties on the initial call. With Barge-In, an extension user can get a message through to a busy co-worker right away.
There are two Barge-In modes: Monitor Mode (Silent Monitor) and Speech Mode. With Monitor Mode, B the caller barging in can listen to another user’s conversation but cannot participate. With Speech Mode, the caller barging in can listen and join another user’s conversation.
The use of monitoring, recording, or listening devices to eavesdrop, monitor, retrieve, or record telephone conversation or other sound activities, whether or not contemporaneous with transmission, may be illegal in certain circumstances under federal or state laws. Legal advice should be sought prior to implementing any practice that monitors or records any telephone conversation. Some federal and state laws require some form of notification to all parties to a telephone conversation, such as using a beep tone or other notification methods or requiring the consent of all parties to the telephone conversation, prior to monitoring or recording the telephone conversation. Some of these laws incorporate strict penalties.
Conditions
•An extension user can barge-in on a conference.
•An extension user cannot barge-in on an Intercom call if one of the intercom callers is using Handsfree Answerback. Both Intercom parties must lift the handset or press Speaker key.
•barged into call can be placed on hold by the originator of the outside call. Both the outside caller and the extension that barged into the call are placed on hold.
•A call which is barged into can be placed on Park by the originator of the outside call, but only the outside caller is placed in Park. The extension which barged into the call is dropped.
•Privacy blocks Barge-In attempts.
•Function keys simplify the Barge-In operation.
•When Silent Monitor Mode is used, Mute key can be used to activate speech path to the internal and external parties.

T1 Lines In DSX-80/160

T1 Lines
In DSX-80/160, provides for connection to advanced digital lines and simplifes installation.
Description
T1 lines require a unique T1 PCB (P/N 80061) and give the system a maximum of 24 lines in a single PCB slot. The available T1 line types include:
• Loop Start (DTMF and Dial Pulse)
• Ground Start (DTMF and Dial Pulse)
• Direct Inward Dial (DID) Wink Start (DTMF and Dial Pulse)
• Direct Inward Dial (DID) Immediate Start (DTMF and Dial Pulse)
• E&M Tie Line Wink Start (DTMF and Dial Pulse)
• E&M Tie Line Immediate Start (DTMF and Dial Pulse)
T1 gives the system the advantages of advanced digital calling as well as conserving PCB slots. For example, you can set up a system with 12 loop start lines, six tie lines, and six DID lines and use only a single PCB slot. Additionally, the T1 PCB has its own on-board processor and DSP so it minimally impacts other system resources.
Note: Although the T1 PCB can connect directly to the telco’s T1 smart jack, your telco may require that you purchase and install a separate Channel Service Unit (CSU). This unit installs between the smart jack and the T1 PCB.
ANI/DNIS Support
The system is compatible with telco's T1 Automatic Number Identifcation (ANI) and Dialed Number Information Service (DNIS) services. ANI/DNIS services can be provided on T1 loop start, ground start, and DID lines (but not E&M). ANI/DNIS Compatibility provides:
• Selectable Receive Format
• You can set up the system for compatibility with any combination of ANI, DNIS and Dialed Number (Address) data provided by the telco.
• Flexible Routing for DID Lines
• For DID lines, the system can route the incoming call based on the received DNIS data and the entries stored in the DID Translation Table. See Direct Inward Line on page 185 for more.
• Caller ID
• The system can use the received ANI data to display the caller’s number on the called extension’s display. The ANI data can be up to 10 digits long.
FSK Caller ID
The T1 PCB can also receive FSK-based Caller ID (if provided by the telco), the same as the COIU (analog) line cards. To receive this type of Caller ID, you must enable DSP Caller ID for the T1 line circuits in programming.

Create an Auto Attendant

Create an Auto Attendant
The following process shows by example the setup for an auto attendant for Embedded Voicemail. In this example the auto-attendant should give callers the option to press 0 for reception (hunt group 200) or 1 for sales (hunt group 301).
38
· For details of routing calls to the auto attendant, see Routing Incoming Calls to an Auto Attendant .
To create an auto attendant:
1.Start IP Office Manager and load the required configuration.
2.Note that if time profiles are going to be used in an auto attendant, the time profile has to be created before creating the auto attendant. For more information, see the IP Office Manager help.
3.Click Auto Attendant. Any existing Auto Attendants are listed.
4.Click Create a New Record in the Group Pane. Select Auto Attendant .In the Name field enter the name for the auto Attendant.

ACD

Automatic Call Distribution (ACD)

Description
You can put any agent in any group. An agent can be in more than one group. This allows, for example, a Technical Service representative to answer customer service calls at lunch when many of the Customer Service representatives are unavailable.
Automatic Call Distribution (ACD) uniformly distributes calls among agents of a programmed ACD Group. When a call rings into an ACD Group, the system automatically routes the call to the agent that has been idle the longest. Automatic Call Distribution is much more sophisticated and comprehensive than Department Calling and other group services – it can accurately judge the work load at each A agent and distribute calls accordingly. The system allows up to 2 ACD Groups and 16 ACD agents.
The ACD Master Number is the extension number of the whole group. Calls directly ringing or transferred to the ACD Master number enter the group and are routed accordingly. Although the master number can be any valid extension number, you should choose a number that is out of the normal extension range.
Automatic Call Distribution operation is further enhanced by: ACD Call Queuing
When all agents in an ACD Group are unavailable, an incoming call queues and causes the Queue Status Display to occur on the ACD agent's display. The display helps the agent keep track of the traffic load in their group.
The Queue Status Displays shows:
•The number of calls queued for an available agent in the group.
•The trunk that has been waiting the longest, and how long it has been waiting.
For each ACD Group, you can set the following conditions:
•The number of trunks that can wait in queue before the Queue Status Display occurs.
•How often the time in queue portion of the display reoccurs.
ACD Overflow (With Announcements)
ACD offers extensive overflow options for another ACD Group. For example, a caller ringing in when all agents are unavailable can hear an initial announcement (called the 1st Announcement). This announcement can be a general greeting like, “Thank you for calling. All of our agents are currently busy helping other customers. Please stay on the line and we will help you shortly.” If the caller continues to wait, you can have them hear another announcement (called the 2nd Announcement) such as, “Your business is important to us. Your call will be automatically answered by the first available agent. Please stay on the line.” If all the ACD Group agents still are unavailable, the call can automatically overflow to another ACD Group or the Voice Mail. If all agents in the overflow ACD Group are busy, Lookback Routing automatically ensures that the waiting call rings into the first agent in either group that becomes free.
You can assign an ACD Group with any combination of 1st Announcement, 2nd Announcement and overflow methods. You can have, for example, a Technical Service group that plays only the 2nd Announcement to callers and then immediately overflows to Voice Mail. At the same time, you can have a Customer Service group that plays both announcements and does not overflow.

Virtual Extensions

Virtual Extensions
Version 2.0 or higher software provides Distinctive ringing (Intercom / Outside) on Virtual Extension.
Version 2.0 or higher software, a special ring tone is provided when a pre-assigned extension places an Intercom call.
Version 3.0 or higher software, number of Ring Tone pattern is increased to 8 from 4.
With version 3.0 or higher if tone pattern 5 ~ 8 is assigned and the system is downgraded to version 2.X or lower incoming ringing will not be provided. To restore ringing, assign the tone pattern to pattern 1 ~
4.
V Version 3.0 or higher software provides the following enhancements to a virtual extension:
•A virtual extension can now display the caller ID of an internal caller (Callers station name is displayed, if station name is not available the extension number is displayed). Also, a virtual extension can now display the caller ID of an internal or external caller when the virtual is not set to ring (Previously the virtual extension must be set to ring or CID is not displayed).
•A virtual extension now has ‘One shot’ ringing which enhances the feature by allowing, either, a single burst of ringing, or normal ringing tone.
Description
Virtual Extensions are available software extensions in the SL1100. A Virtual Extension assigned to a line key, can appear and ring on an individual station or multiple stations and be used for outbound access.
Up to 50 VE keys are provided.
Conditions
•The 84 available ports/Extensions are assigned on a per extension basis for Virtual Extension key mode.
•The 84 available ports/Extensions are assigned per extension for CAR key mode or Virtual Extension key mode.
•More than one extension can share a Virtual Extension key.
•An extension can have more than one Virtual Extension key assigned.
Assigning a Virtual Extension key of the extension the key is assigned on is not supported.
•Up to 32 incoming calls can be queued to busy Virtual Extension key.
•You cannot have a CAR key and Virtual Extension on the same telephone.
•Virtual Extensions do not support the following features:
-Barge-In
-Conference
-Conference, Voice Call/Privacy Release
-Reverse Voice Over
-Tone Override
-Voice Over
•When a valid system station calls a Virtual Extension appearing on another station, Voice and MW softkeys appear in the display of the calling station, but they do not operate.
•When talking on a Virtual Extension you cannot mute the handset.
•Incoming calls to a virtual extension that appear on stations that are used with the CTI applications, PC Assistant, or PC Attendant, do not show up as a second call in the CTI application.

DHCP

DHCP Client
Description
WARNING: When the VoIPDB is installed on the CPU, the built in LAN port on the CPU becomes disabled. Only the LAN Port on the VoIPDB will be operational.
DHCP Client will access an external DHCP server every time the LAN cable is connected to the CPU/ VoIPDB or when the System is powered up. The System can receive the following information from the DHCP server:
IP Address, Subnet Mask, and Default Gateway.
Conditions
•The DHCP Server should be configured to provide the system the same IP address every time. For example in the DHCP server extend the lease time to infinite or setup the server to provide the same IP address based on the systems MAC Address.
•When changing PRG 10-63-01 (DHCP Client Enable/Disable) a system reset is required for this change to become effective.
•DHCP client can set following programs automatically; however other IP related programs (such as PRG 84-26) have to set manually as required.
-IP Addresses: PRG 10-12-01 (CPU), PRG 10-12-09 (VOIPDB)
-Subnet Masks: PRG 10-12-02 (CPU), PRG 10-12-10 (VOIPDB)
-Default Gateway: PRG 10-12-03
•DHCP Client (PRG 10-63) and existing DHCP Server feature (PRG 10-13) can not be used at the same time.
•While the System accesses the DHCP Server, to receive IP Addressing information, the CPU RUN LED flashes as follows. If the System fails to receive an IP Address from the DHCP server the system will use the IP Address assigned in PRG 10-12.

300 ms
On Off
300
ms 300
ms
200
ms 200
ms 700
ms
•Once after IP Address and Subnet Mask are set, if different IP Addresses or Subnet Mask is delivered during normal operation mode, both LED2 (Red lit) and RUN LED (flash as above) indicate system requires reset.

Tone Service

Howler Tone Service
Description
Howler Tone Service provides a Howler Tone when a station remains off-hook after a call is completed or when a station is off-hook and digits are not dialed in a programmed time.
Conditions
Howler tone is generated 30 seconds after a call is disconnected and the telephone is left off-hook or the telephone is left off-hook without dialin

IPEKCS programming

LCD & Button Functions
While in the PROGRAM MODE, the Liquid Crystal Display (LCD) and Flex button LEDs of an Admin Station are used to guide and indicate status of the feature. The dial-pad is most often used to enter data after selecting a data item using the Flex buttons. In some cases, pressing a Flex button will toggle the entry with the Flex button LED indicating the status (ON/OFF).
For PROGRAM CODES with multiple Flex button selections, the volume controls ([VOL UP] and [VOL DOWN] buttons) may be used to select the next or previous item. The [SPEED] button is generally employed as a delete button to erase existing entries however, where noted, it may be used to confirm a range input. Pressing the [CONF] button returns to the 1st step of the data entry procedure for the PROGRAM CODE without storing unsaved entries.
The [SAVE] button is used to store data after entry. If there are no conflicts in the entered data, confirmation tone will be received and the data stored. If a conflict exists, error tone is provided and newly entered data are not saved. Generally, corrected data may be entered and stored without restarting the entry procedure from the 1st step.

Conference

CONFERENCE
Conference Room
Description
In addition to ad-hoc conferencing, users may establish a Conference Room. Other internal and external parties are invited to the conference and can join the conference without further action by the user who established the Conference Room. A user can transfer an active call to a Conference Room. A Conference Room can be password protected so that only parties that enter the password are allowed to join the Room.
Up to 9 Conference Rooms can be set-up and each can support a maximum of 32 parties with the g.711 or g.729 codec or 24 parties with the g.723 codec. Conference Rooms employ channels from
an MCIM (Multi-party Conference Interface Module). Each MCIM supports up to 32 parties and multiple MCIM units may be installed as shown in the chart below. iPECS-Micro 1 MCIM unit maximum iPECS-50 & 100 2 MCIM units maximum iPECS-300 4 MCIM units maximum iPECS-600 8 MCIM units maximum iPECS-1200 30 MCIM units maximum

IP Office

Essential Edition – Norstar Version still available? Yes – they have been renamed “Basic Edition – PARTNER® Mode” and “Basic Edition – Norstar Mode”. Functionality is similar to Basic Edition. Can I upgrade Basic Edition? Yes. Basic Edition may be upgraded to Essential Edition by purchasing and applying the new Essential Edition License. During this upgrade, the configuration of the system will be retained. Can we still forward/copy voicemail to an email address with Basic Edition? Yes. What is the maximum capacity of Basic Edition? • Up to 100 digital phones (if using only Nortel T series and M series phones) or up to 98 digital phones (any combination of 1400, 9500 and Nortel T series and M-series) supported on DS8, TCM8, DS16/16A and DS30/30A modules • Up to 100 analog phones • Up to 18 PARTNER ETR phones (North America only, additional non ETR phones can be added total to 100) • Up to 32 analog CO lines • 1 PRI / E1 (30 lines, EMEA/MENA only) • 1 PRI / T1 (24 lines, North America only) • Up to 12 BRI channels • Up to 20 SIP lines How many ports and hours of voicemail storage come with Basic Edition? Basic Edition Voice Messaging comes with 2 ports and 15 hours as standard. Can I add more messaging ports or storage time to Basic Edition? Yes - Additional ports can be added while simultaneously increasing storage capacity. With 2 additional ports, there are 4 ports in total, 20 hours of storage; with 4 additional ports, there are 6 ports in total, 25 hours of storage. How many Auto Attendants are available with Basic Edition? Basic Edition supports up to 9 Auto Attendants. Can IP500v2 systems with Basic Edition be networked? No.

Pulse to Tone conversion

An extension can use Pulse to Tone Conversion on trunk calls. Pulse to Tone Conversion lets a user change their extension dialing mode while placing a call. For systems in a Dial Pulse area, this permits users to access dial-up OCCs (Other Common Carriers - such as MCI) from their DP area. The user can, for example:
•Place a call to an OCC over a DP trunk.
•Depending on programming:
Manually implement Pulse to Tone Conversion
- OR -
Wait 10 seconds.
•Dial the OCC security code and desired number. The system dials the digits after the conversion as DTMF.

Automatic Release

Automatic Release
Description
Automatic Release drops the line circuit when an outside party abandons the call. For this feature to work with Loop Start Trunks, the CO/PBX providing the outside line must provide a timed disconnect signal. Automatic Release is normally provided on Ground start trunks are not supported please remove, DID and ISDN trunks.
Conditions
•Automatic Release on ISDN trunks is provided by the protocol.
•When an outside line is accessed using a dedicated line key, the LED associated with the line key goes off when Automatic Release occurs.
•On Loop Start trunks Automatic Release is only available on incoming calls.
•This feature functions while a call is in progress, on hold, or in a conference.
•This feature applies to all ICM type calls in progress, holding or parked.
•When Automatic Release occurs and the telephone is in handsfree mode, Speaker automatically turns off. If using the handset, the station is set to idle when the handset goes on-hook.

Auto attendant

Code Control See Security Code. Voice Mail
Alternate Next Call
Routing Mailbox See Next Call Routing Mailbox. Voice Mail
Announcement Mailbox Mailbox that allows a recorded greeting to play to callers. Automated Attendant
Announcement Message The message that the System Administrator records for a specific Announce-
ment Mailbox. Automated
Attendant
Answer Table Determines how the Automated Attendant answers outside calls on each trunk, according to the time of the day and day of the week that the call is ringing.
Automated Attendant
Answering Schedule
Override Enables an alternate greeting and alternate dialing options for callers. Automated
Attendant
Auto Attendant Direct to Voice Mail
Auto Attendant Direct to Voice Mail sends Automated Attendant calls directly to Automated an extension user's mailbox. Their phone does not ring for calls from the Auto- mated Attendant.
See Greeting in this glossary.
Attendant
Auto Erase or Save When a mailbox user completely listens to a new message and then exits their
mailbox, InMail either automatically archives or deletes the message. Voice Mail
Auto Time Stamp After a user listens to a message, InMail can optionally announce the time and
date the message was left. Voice Mail
Automated Attendant The Automated Attendant can automatically answer the telephone system incoming calls, play an Instruction Menu message, and provide dialing options to callers.
Automated Attendant
Automated Attendant Transfer
While on a trunk call, an extension user can transfer the trunk call to the Auto- mated Attendant so the caller can use the Automated Attendant dialing
options. Voice Mail
Automatic Access to VM by Caller ID
InMail mailbox can be associated with a specific caller ID (CID) number. When the CID number is presented to the InMail it will automatically log the user into
their mailbox. Voice Mail
Automatic Call Routing
to a Mailbox See Go to a Mailbox in this glossary. Automated
Attendant
Automatic Message
Erase/Save See Auto Erase or Save in this glossary. Voice Mail
Automatic Routing for
Rotary Dial Callers If an Automated Attendant caller does not dial any digits, InMail automatically
routes them to a specified option (such as the operator or a mailbox). Automated
Attendant

Automatic extension privacy

Automatic Extension Privacy
Use this feature to prevent a user from joining an active call at an extension. When Automatic Extension Privacy is Assigned for an extension, other users cannot join active calls at that extension. This feature is typically used for single-line telephones and extensions connected to fax machines, modems, and credit card scanners, which make and receive data calls that should not be interrupted. This feature can provide the Privacy (F07) function for single-line telephones.
Considerations 4
■ If you want to be able to intercept calls routed to an auxiliary device—such as an answering machine, a voice messaging system, or an auto attendant—make sure Automatic Extension Privacy is Not Assigned for the auxiliary equipment extension.
■ Single-line telephones and system telephones without a programmed Privacy button cannot override this feature once it is assigned to an extension.
■ If Automatic Extension Privacy is Assigned at an extension, the green light is lit automatically after programming the Privacy button to indicate that Privacy is currently active.
■ Automatic Extension Privacy applies only to active calls. Any user can retrieve a held call unless Exclusive Hold is used.

Tip/Ring Device

Tip/Ring Device Requirements 9
A tip/ring device must meet the following conditions:
■ It must be nonproprietary; that is, it cannot be made specifically for use on a particular telephone system.
■ Its Ringer Equivalence Number (REN) cannot be greater than 2.0. (REN) is a measure of the power it takes to ring a telephone. Each extension jack in your system handles up to 2.0 RENs.) The REN is shown on a label on the device, usually on the bottom.
You can connect a tip/ring two-line device to the system, but it should be installed and used as if it were a single-line device.

AVAYA Partner Abbreviated ringing

Abbreviated Ringing (#305) 4
Use this feature to activate or deactivate Abbreviated Ringing at the system telephone at a specific extension. When you are on a call and Abbreviated Ringing is Active, any incoming call rings only once. The green light next to the line or pool button flashes until the call is answered or the caller hangs up (or for a transferred call, until the call returns to the transfer return extension). This feature prevents incoming calls from distracting you when you are busy on another call. To allow calls to ring repeatedly, set Abbreviated Ringing to Not Active.
Considerations 4 ■ Abbreviated Ringing applies only to system telephones.
■ Abbreviated Ringing is typically set to Not Active for operators and others who handle many calls quickly so they have an audible reminder of incoming calls.
■ Abbreviated Ringing applies to outside, transferred, and intercom calls. ■ The volume of an abbreviated ring is lower than a normal ring.
■ Calls to a busy extension ring at a lower volume than normal even if Abbreviated Ringing is set to Not Active.
Programming 4
To change the Abbreviated Ringing setting for an extension:
1. Press f00ss#305 at extension 10 or 11.
2.Enter the number of the extension to be programmed.
3.To set Abbreviated Ringing, press D until the appropriate value appears.
■ 1 = Active (incoming calls ring once; the factory setting)  ■ 2 = Not Active (incoming calls ring repeatedly)
4.To program another extension, press n or p until the extension number appears on the display. Then repeat step 3.
5. Select another procedure, or exit programming mode.