Data Line Security

Data Line Security
Data Line Security protects any station port from receiving audible tones (such as Camp-On or Override) and denies a station from barging in while busy to prevent disruption of data transmission when using a modem or facsimile machine.
Conditions
• When a Multiline Terminal and a Single Line Terminal are assigned for Data Line Security, Tone Override/Voice Override and Call Alert notification tone are not heard over the handset speaker.
• Data Line Security protects a station from Barge-in, even when Barge-In is allowed in Class of Service.
• When any Multiline Terminal or Single Line Terminal calls a station with Data Line Security, a constant busy tone is heard.

Retrieve VM Messages

Retrieve VM Messages
To retrieve VM messages from outside of office: Trunk 1: 03-1234-5678 (DIL) Outside party number: 09087654321 PRG 22-02-01: Trunk 1 DIL
PRG 22-07-01: VM Pilot number, 300
PRG 13-04-01: Speed Dial area No.0 -> 09087654321 PRG 13-04-11: Speed Dial area No.0 -> 100 (VM BOX)
1. Call DIL number.
2. After the VM is answered, user can enter VM Box 100 directly.
3. Announce the zone.
- OR -
Trunk 1: 03-1234-5678 (DID) Outside party number: 09087654321 PRG 22-02-01: Trunk 1 DID
PRG 22-11-05: Set transfer destination, 102 InMail PRG 13-04-01: Speed Dial area No.0 -> 09087654321 PRG 13-04-11: Speed Dial area No.0 -> 100 (VM BOX)
1. Call the DID number to set the transfer to InMail from outside party.
2. After the VM is answered, user can enter VM Box 100 directly.
To retrieve VM messages from Mobile extension: Trunk 1: 03-1234-5678 (DIL) Mobile extension number: 4321 PRG 22-02-01: Trunk 1 DIL PRG 22-07-01: VM Pilot number, 300

Call Coverage Keys

A keyset can have Call Coverage Keys for a co-worker’s extensions, Ring Group master numbers and UCD Group master numbers. The Call Coverage Key lights when the co-worker’s extension is busy, flashes slowly when the co-worker has an incoming call, and flashes fast when the co-worker is in Do Not Disturb. The Call Coverage Key can ring immediately when a call comes into the covered extension, ring after a delay or not ring at all. In addition, the keyset user can press the Call Coverage Key to intercept their co-worker’s incoming call. They can also go off hook and press the Call Coverage key to call the covered extension. An extension can have as many Call Coverage Keys as they have available Feature Keys on their telephone.
Call Coverage Key Busy Lamp Indications The covered extension is: When the key is: Idle or not installed Off Busy On Ringing Slow Flash Covered extension is in DND for outside calls (option 1). Medium Flash Covered extension is in DND for Intercom calls (option 2) or All Calls (option 3). Fast Flash
Call Coverage Keys will intercept the following types of calls:
• Key Ring Calls
• Ringing Intercom calls
• Calls to a UCD Group master number
• Calls ringing a Group Ring master number
• Transferred calls
Call Coverage Keys will not intercept a call ringing the attendant’s Operator Call Key.
Call Coverage Guard Timer
The 4 second Call Coverage Guard Timer helps extensions that have the same Call Coverage key assignments. As soon as an extension user presses their Call Coverage key to answer a call, the key becomes unavailable for 4 seconds to all other extensions with that same key. (Users will hear reorder tone if they press their key before the 4 seconds expire.) This helps prevent users from inadvertently placing a call to the covered destination.
Hotline and Call Coverage Key Surfing
Consecutively pressing Call Coverage or Hotline keys, also called “sur ng”, is a convenient way to locate co-workers. The operation of sur ng is an interaction between the Hotline Automatic Transfer and Automatic Hold options. The chart below shows this interaction. For example, when Hotline Automatic Transfer and Automatic Hold are both enabled, the user on an outside call can quickly surf a row of keys to nd a co-worker and then hang up when they nd them. The call transfers to the co-worker without any other steps.

Terminal Type

Terminal Type 0 = Not set
1 = Keyset/DSLT 2 = SLT Adapter 3 = -- Not used -- 4 = -- Not used -- 5 = -- Not used -- 6 = PGD (Paging) 7 = PGD (Tone Ringer) 8 = PGD (Door Box) 9 = PGD (ACI) 10 = DSS Console 11 = -- Not used --
02 Logical Port Number 0 = Not set
1 = Keyset (1-256)
2 = SLT Adapter (1-256) 3 = Not used 4 = Not used 5 = Not used 6 = PGD (Paging) (1-8) 7 = PGD (for Tone Ringer) (1-8) 8 = PGD (for Door Box) (1-8) 9 = PGD (for Analog I/F) (1-96) 10 = DSS (1-32) 11 = Not used

FUNCTION CODE

FUNCTION CODE FUNCTION
10 Enblock Dialing, 600 & 7000 only 11 X Intercom Differential Ring ( X = 1-8 ) 12 X CO Line Differential Ring ( X = 1-8 ) 13 Intercom Answer Mode (1: HF/2: TONE/3: PV) 14X Call Coverage Attribute (1: On/Off, 2: Ring Delay) 15X Station Ring Download ( X = 0-9 ) 19 Ear-Mic Headset, 600 & 7000 only 21 Knock Down Station COS 22 Restore Station COS 23 Walking COS 24 ICR SCENARIO 31 Message Retrieve Method 32 Message Retrieve Example 33 User Authorization Code Registration 34 {DID CALL WAIT} button assignment 35 Message Wait in Executive/Secretary pair 36 Send SMS Message
37 Register Mobile Extension 38 Make Mobile Extension active 39 Register Mobile Extension CLI 41 Set Wake-Up Time
42 Wake-Up Time Disable
51 XX Custom/Pre-select Message Display (XX = 00-20) 52 Register Custom Message (Message 00) 53 Create Conference Room 54 Delete Conference Room 57 {Call Log Display} Button Assignment

Alarm

Alarm
Alarm lets any station extension work like an Alarm clock. An extension user can have Alarm remind them of a meeting or an appointment. There are two types of Alarms:
• Alarm 1 (sounds only once at the preset time)
• Alarm 2 (sounds every day at the preset time)
Conditions
• Single Line Terminals ring and Music on Hold is heard when the Alarm sounds.
• Only a Multiline Terminal user can view what time the Alarm is currently set for.

CLI Message Wait

CLI Message Wait
Description
When a call is received with Caller or Calling Line ID, the Id is displayed in the DKT LCD as well as the LCD of compatible SLTs. If the call is an ISDN line DID call and disconnects before the called station can answer, the system can activate a CLI Message Wait for the called station. The DKT LCD shows the CLI Message and the [Call Back] button LED flashes. The user may view the CLI Messages, return the call, save the CLI number as a Speed Dial or simply delete the CLI Message.
Operation
To view CLI Messages,
1. Press the [Call Back] button to display the first message. 0314504626
DATE TIME CNT:xx
2. Use the Volume Up/Down buttons to scroll through messages.
To manage CLI Messages,
1. Press the Select softkey to select a CLI Message.
2. To call the number, press the Hold/Save button.
To delete the current CLI Message, press Del Cur softkey.
To store the number as a Speed Dial, press the Save softkey or. To delete all CLI messages press the DND/FOR button.

Answer Customer Calls Promptly and Professionally

Answer Customer Calls Promptly and Professionally, Even When You’re on the Other Line
No matter how small your business, projecting the right image is important. Build your company’s image with Viking’s Two Line Call Sequencer.
In the day mode, the TMS-2 answers calls on one line while you are busy or on the opposite line. After a professional greeting has been played, the TMS-2 holds the caller for up to 15 minutes or until you are available to take the call.

call. Music on hold can be provided from a radio or any other audio source. For a truly
professional image, Viking’s DVA-2W “promotion on hold announcer” may also be added (Fax Back Document 110).
In the night mode, the TMS-2 allows your answering machine to answer either line. The LED indicator displays which mode is active (day or night) and which line is on hold. A volume adjustable beep and warble also indicate which line is on hold

To Camp On (wait without hanging up), .

To Camp On (wait without hanging up), .
• (Intercom calls) When you hear ringing,
wait for the called party to answer. If you hear busy/ring instead of busy before camping on, you can optionally dial the Barge In code to send a Voice Over. Check with your Communications Manager for the code.
• (Outside calls) When you hear new dial
tone, place your call again. OR
1. To leave a Callback for a busy line or
extension, and hang up.
• Wait for the system to call you back.
2. or lift handset.
• (Outside calls) Place your call again.
• (Intercom calls) Speak to co-worker.

NAME ENTRY

NAME ENTRY
Description
A SLT user has the capability to program the user name so that a calling user with an LCD can see the name instead of the station number.
Operation SLT
To register the name at the SLT
1. Lift the handset.
2. Dial ‘561’, the SLT Programming code, confirmation tone is heard.
3. Dial ‘74’, the SLT Name Program Code.
4. Enter name, refer to Station Speed Dial, Alphanumeric Chart.
5. Momentarily depress the hook-switch, receive confirmation tone.
To delete the name at the SLT;
1. Lift the handset.
2. Dial ‘561’, the SLT Programming code, confirmation tone is heard.
3. Dial ‘74’, the SLT Name Program Code.
4. Momentarily depress the hook-switch, receive confirmation tone.

Viking entry phones

• Allows 1-4 entry phones to share a single telephone line with a residential or business telephone system
• Compatible with the following Viking entry phones:
- E-10A, E-20B, E-30, E-35, E-40, E-50, E-60, E-65, E-70, and E-75 Phones
- K-1500-7 Stainless Steel Panel Phone
- K-1700-3 and K-1705-3 Phone with Keypad
- K-1900-8 Stainless Steel Panel Phone with Keypad
- Or use with any analog touch tone phone
• Provide entry phone activated chimes and whole house paging when used with an SLP-1 or SLP-4
• Up to 6 different keyless entry codes / “one time use” keyless entry codes per entrance
• Up to 6 master keyless entry codes for use at any entrance
• Analog Station Dial Through Mode allows entry phones to connect to analog PABX station or telephone line
• Optional “Privacy Number” allows only selected visitors to call from an entry phone
• Provides “Caller ID” and “Call Waiting Caller ID”
• Produces “Call Waiting” tones
• Lighted doorbell switch input to activate entry phone call
• One set of normally open and normally closed contacts available for each entry phone
• Programmable contact activation time
• Auxiliary input to activate timed auxiliary contacts

CALL WAIT & BROKER CALL

CALL WAIT & BROKER CALL
Description
When SIP Phone receives second call wait, user can switch talking to each other. There are a few kinds of call wait and according to the type of call wait operation of switching talk is different.
Operation Receive 2nd call
1) In Case of, SIP Phone has no Call Wait feature or Call Wait is disabled
- If caller is from CO line then the call will be rerouted by system setting automatically
- If caller is from a Station then the caller will hear busy tone. During busy tone, the user
can press ‘*’ for Camp-On call or ‘#’ for Voice-Over. The SIP Phone user who receives this 2nd
call will hear notification tone from system.
- The SIP Phone user who receives this 2nd can make hold for 1st call talking and switch to 2nd
call for talking by press ‘HOLD’ twice
2) In Case of, SIP Phone has Call Wait feature and Call Wait is enabled
- The 2nd call is accepted by the SIP Phone and user who receives this 2nd call will hear
notification tone from Phone itself.
- The caller from CO or Station will hear ringback tone ore coloring.
- The SIP Phone user who receives this 2nd can make hold for 1st call talking and switch to 2nd
call for talking by,
a ) 3 rd party SIP Phone : press ‘HOLD’ twice
b) LG-Ericsson WIT400H SIP Phone : press ‘HOLD’ once
c) LG-Ericsson LIP88xx and LIP8002 SIP Phone : press Up or Down of Navigation button
Hold 1st call and Make 2nd call
1) SIP Phone user makes hold with 1st call and makes 2nd call.
2) After 2nd call is connected the SIP Phone user talks with 2nd call party.
3) If he want to return to 1st call party without hangup the 2nd call,
- 3rd party SIP Phone : press ‘HOLD’ once
- LG-Ericsson WIT400H SIP Phone : press ‘HOLD’ once
- LG-Ericsson LIP88xx and LIP8002 SIP Phone : press Up or Down of Navigation button

Directory Dialing

Directory Dialing allows a Multiline Terminal user to select a co-worker or outside caller from a list of names, rather than dialing the telephone number. There are four types of Directory Dialing:
• SPD-Speed Dials
• EXT-co-worker’s Extensions
• STA-Personal Speed Dials
• TELBK-Telephone Book
Conditions
• Directory Dialing sorts and searches directory names in alphabetical order (based on all characters entered of the name) when the system starts up or reboots. The system resorts extension names when:
- You change PRG 15-01-01 (Extension Numbers and Names).
- Any user dials 700 and changes their extension name.
• Directory Dialing follows all the programmed options and conditions for Speed Dial-System/Group/ Station, Intercom Calling and One-Touch Calling.
• Extension Directory only shows a telephones/VEs that are connected and have a name assigned

LCD & Button Functions

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.

Delayed Ringing

Delayed Ringing
Delayed Ringing allows programmed secondary answering positions to ring on incoming calls after a programmed time. This feature applies to CO/PBX lines and Virtual Extension Keys.
Conditions
• An extension user can answer an outside call just by lifting the handset (depending on programming).
• Terminals must have CO line appearance for a trunk call to be answered on the telephone

Flexible Timeouts

Flexible Timeouts
The Flexible Timeouts feature provides a variety of timers in the Resident System Program to allow the system to operate without initial programming. The system timers can be changed to meet customer needs according to the system application requirements.
A Timer Class is used to allow terminals and trunks to have different timers for the same feature. There are 16 timer Classes (0~15). The following table shows the Programs that are used depending on the Timer Class used:

VOICE OVER

VOICE OVER
Description
This feature allows users of iPECS Phones, off-hook on a call (CO, IP or Intercom), to receive a voice announcement through the handset receiver with the existing call. The Voice Over is muted so as not to interfere with the existing conversation. The called station user may then respond to the calling party using Camp-On response or may use Silent Text Messaging to respond.

Howler Tone Service

Howler Tone Service
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 dialing.

Licensing

Licensing
Licenses are used to activate certain features and applications for the SL1100. The SL1100 system provides the following licenses:
System Licenses:
System Capacity
• IP Trunk - Additional SIP Trunk Port License (Initially 4 Ports bundled)
• IP Extension (STD-SIP) - Standard SIP Terminal License
• Mobile Extension - Additional Mobile Extension Port License (Initially 4 Ports bundled)
System Feature License
• Hotel/Motel - This licenses the system to run the Hotel/Motel feature.
• Encryption (VoIP) - Encryption License for Multiline IP Terminal
• 32 Channel VOIP - Addtional 16 Channel VOIP DSP License
• InACD - This license the system to run the Automatic Call Distribution feature. (V1.5 or higher)
• SL Net - This license the number of remote system that can be connected to the main system. (V1.5 or higher)
Voice Mail (Embedded)
• InMail Channel - Additional InMail Channel License
• InMail Advance - InMail Advanced Features License
- E-Mail Notification
- Cascading message notification
- Find-Me/Follow-Me
- Password option
- Hotel/Motel
Applications:
Desktop Application
• Softphone - This licenses the number of Desktop Applications that can be used for Softphone.
• Desktop Client - This licenses the number of Desktop Applications that can be run.
60 Day Free License
The 60 Day Free License comes with the CPU. It allows for all the features to be active for 60 days. The count down starts on the first power on and ends at midnight of the 60th day.
• By default, the 60 Day Free License is set to disabled. The 60 day count down starts when the system is initially powered on and continues if the 60 Day Free License is disabled or enabled.
• The CPU works for 1440 hours from the first time powered on.
• The clock counts down only when the power supply in the KSU is ON - battery is not in effect.
• If the CPU is removed, or the system is powered OFF, the countdown stops.
• Every time the clock is changed, the CPU free license (60 days) loses one hour.
• While the free license is active the user can increase the port size of the system to maximum by

Multiple Trunk Types

Multiple Trunk Types
The SL1100 supports many different Trunks in the system (DID, Loop Start, ISDN PRI). The system supports up to 84 trunks using expanded KSUs.
DID
Refer to the Direct Inward Dialing (DID) on page 1-257 feature for related information. Loop Start Trunks
Loop Start Trunks can be connected to the SL1100 system. Loop Start is assigned per trunk at the associated unit.
ISDN PRI
Refer to the ISDN Compatibility on page 1-564 feature for related information. T1-E1 Trunks
The T1/PRI/E1 Interface gives the system T1/E1 trunking ability. This unit uses a single universal slot and provides up to 24/30 trunk circuits. In additional to providing digital-quality trunking, the T1/PRI Interface allows you to have maximum trunking ability with fewer units. This in turn makes more universal slots available for other functions.

Park

Park
Park places a call in a waiting state (called a Park Orbit) so that an extension user may pick it up. There are two types of Park: System and Personal. Use System Park when you want to have the call wait in a system orbit. Personal Park allows a user to Park a call at their extension so a co-worker can pick it up. After parking a call in orbit, a user can Page the person receiving the call and hang up. The paged party dials a code or presses a programmed Park key to pick up the call. With Park, it is not necessary to locate a person to handle their calls. A call parked for too long recalls the extension that initially parked it, however the call remains in the park orbit until it is answered. There are 64 Park Orbits (1~64) available for use.
Extended Park
An extension Class of Service determines whether it uses the normal Park Orbit Recall time or the Extended Park Orbit Recall time. The timers are set in system programming. When an extension with Extended Park Recall Class of Service option parks a call, it recalls after the Extended Park Orbit Recall time. When an extension with the Normal Park Orbit Recall Class of Service option parks a call, it recalls after the normal Park Orbit Recall time, however the call remains in the park orbit until it is answered.
Programmable Function Key and Service Code Available for Personal Park
The Personal Park feature is enhanced by using a Programmable Function Key or service code (3- digit or 1-digit) to place a call in Personal Park. This option is available for Multiline Terminals and single line sets and can be used for analog or ISDN trunks.

Hotel/Motel - Room Status Printout

Hotel/Motel - Room Status Printout
Use the Room Status Printout to get detailed, up-to-the-minute printouts that show the status of all your rooms. Use the Room Status Printout to get a concise overview of the status of guest rooms at a glance. The printout gives you up to the minute reports showing Check In Status, Room Call Restriction, Do Not Disturb, Message Waiting and Wake Up Calls. This feature requires a connection to the system using an IP port on the CPU. Five separate reports are available
Room Status List (Option 1)
The Room Status List shows the status of each room. This gives you an overview of all rooms in a single report. In the report below:
• Room Clean
Lists all the Checked In rooms (305, 311 and 315).
• Maid Required
Lists all the vacant rooms that need cleaning (309).
• Maid in Room
Lists the rooms in which house cleaning is currently working (317).
• Inspection Required
Lists the rooms that are Checked Out waiting to be cleaned up (313).

Distinctive Ringing, Tones and Flash Patterns

Distinctive Ringing, Tones and Flash
Patterns
Version 3.0 or higher software provides; number of Tone pattern is increased to 8 from 4. After setting new system data (Tone Pattern 5-8), then downgrade from V3.0 to before V2.0 may cause no incoming ring issue.
Description
Distinctive Ringing, Tones and Flash Patterns provide extension users with audible and visual call status signals. This lets users tell the type of calls by listening to the ringing/tones and watching the keys. It also helps users monitor the progress of their calls. In addition, Distinctive Ringing lets Multiline Terminal users customize their Intercom and trunk call ringing. This is helpful for users that work together closely. For example, if several co-workers set their Multiline Terminals to ring at different pitches, each co-worker can always tell which calls are for them. You can also customize the tones the system uses for splash tone, confirmation tone, trunk ring tone, Intercom ring tone and Alarm ring tone. Refer to the chart below and the SL1100 Programming Manual for more details.

Class of Service

Class of Service
Class of Service (COS) sets various features and dialing options (called items) for extensions. The system allows any number of extensions to share the same Class of Service. An extension can have a different Class of Service for each of the Night Service modes. This lets you program a different set of dialing options for daytime operation, nighttime operation and even during lunch breaks. An extension Class of Service can be changed in system programming or via a Service Code (normally 677). There are 15 available Classes of Service.
Conditions
• Before assigning a new COS, make sure the new COS matches the old COS or you may enable options, which the extension should not have or remove options, which it should have.
• An extension can have a different Class of Service for each Service mode. At default, the Mode names are assigned as follows:
- Mode 1 = No setting
- Mode 2 = Night
- Mode 3 = Midnight
- Mode 4 = Rest
- Mode 5 = Day2
- Mode 6 = Night2
- Mode 7 = Midnight2
- Mode 8 = Rest2
• If a user dials a number not programmed in ARS, PRG 26-01-03 determines if the system should route over the trunk group settings defined in PRG 21-02 or play an error tone.
• When using ARS Class of Service, with PRG 26-01-03 set to (1) "Play Warning Tone", any trunk pointed or transferred to a virtual that is Call Forward Off-Premise will not complete. For a virtual to Call Forward Off-Premise, PRG 26-01-03 must be set to "Route to trunk group" and the call will follow the trunk group settings of the trunk, assigned in PRG 21-03.
• When using ARS Class of Service, with PRG 26-01-03 set to (1) "Play Warning Tone" or transferred to a virtual that is Call Forward Off-Premise will always follow ARS Class 1 routing properties.

STATION INDIVIDUAL CALL ROUTING (ICR)

STATION INDIVIDUAL CALL ROUTING (ICR)
Description
Station ICR is an extension of Call Forward where the user establishes a routing scenario. Each of the ten scenarios defines rules to route incoming calls based on Time, Day of week, Date and Caller ID to a destination defined by the User. Each scenario is assigned a priority of 0 to 9. When an incoming call is received at the station, the System will search the ICR scenarios entered by the User, then the call will routed according to the destination in the highest priority matching scenario.
Operation iPECS Phone To create a scenario:
1. Press the [PGM] button.
2. Dial 24, the ICR menu or log on to the Station Web portal.
3. Select the desired Scenario number (0 - 9).
4. Select the type of Caller ID (0 – 5):  Type 0 – Station CID  Type 1 - All Station  Type 2 – CO CID

HEADSET COMPATIBILITY

HEADSET COMPATIBILITY
Description
An industry standard headset can be connected to an iPECS Phone in place of or in addition to the handset. The station is then programmed for Headset operation.
In the Headset mode, pressing the [SPEAKER] button will send audio to the Headset instead of the speakerphone. In addition, when in the Headset mode, ring signals can be delivered to the speaker or the headset as defined in the system database.
2-92

DND - ONE TIME DND

DND - ONE TIME DND
Description
A station, which is ringing or receiving Off-hook muted ring, can press the [DND] button, to reject the call and terminate ringing. The station is placed in DND, ringing terminates and the call receives treatment based on the following precedence:
1. Previous or active Call Forward busy.
2. Preset Call Forward busy.
3. Station Call Coverage.
4. Direct Transfer to Voice Mailbox.
5. Return busy signal and disconnect.
When the station returns to the idle status, DND is cancelled and the [DND] LED is extinguished.
Operation iPECS Phone To activate One Time DND while ringing
1. Press the [DND] button, the [DND] LED lights, station goes to DND state.
System Deactivation
1. When the station returns to idle, DND is disabled and the [DND] LED extinguishes.
Conditions
1. CO/IP recalls will override One Time DND.
2. The Attendant can override stations in One Time DND by using Camp-On or intrusion. The Attendant does not have One Time DND service.
3. One Time DND cancels existing Callback queues.

DISABLE OUTGOING CO/IP ACCESS Description

DISABLE OUTGOING CO/IP ACCESS
Description
The System Attendant can place CO/IP lines out-of service, disabling outgoing calls on the CO/IP path. This is normally done should an undetected fault interrupt service on a CO/IP path. Incoming calls continue to be processed normally.
Operation System Attendant To disable/enable Outgoing CO/IP access (toggle)
1. Press the [PGM] button.
2. Dial ‘072’, the Attendant Station Program code.
3. Press the {CO} button of the line(s) to be disabled, confirmation tone is heard and the status for the selected line(s) is changed.
Conditions
1. If the desired CO/IP line is in use, the System Attendant may still disable the CO/IP line. The feature will take effect after the desired CO/IP line goes to idle.
2. Once the line is disabled, all Attendant appearances for the disabled CO/IP line will flutter at 240 ipm, other stations will indicate the CO/IP line as busy, LED is On.
3. The CO/IP line outgoing access status is stored in battery-protected memory in case of a power failure.
4. Multiple CO/IP lines may be enabled/disabled without redialing the Attendant Station Program code. Confirmation tone is heard after each CO/IP line is enabled/disabled.
5. When the system detects a fault on an analog CO line, the CO line is disabled for outgoing access automatically.
6. Incoming calls on a disabled CO/IP line will continue to operate normally.

CALL FORWARD, ATTENDANT

CALL FORWARD, ATTENDANT
Description
As with other stations, Attendants can forward calls to other stations in the system. Calls may be forwarded unconditionally, on busy or no answer.
Operation Attendant To activate Call Forward
1. Lift the handset or press the [SPEAKER] button to receive intercom dial tone.
2. Press the [FWD] button.
3. Dial code ‘0’ ~ ‘5’, the Call Forward type.
4. Dial the station number to receive forwarded calls.
6-65

ALTERNATE ATTENDANT

ALTERNATE ATTENDANT
Description
This feature allows an Alternate answer point while the Attendant station is in an unavailable mode. When in the unavailable mode, the next available Attendant in the Attendant group will receive Attendant calls and recalls.
Operation Attendant To assign a flexible button to activate {ALT ATD} button [PGM] + {FLEX} + ‘562’ + [SAVE]
To toggle Attendant Unavailable feature
1. Dial ‘562’, the Alternate Attendant code or press {ALT ATD}.
Conditions
1. Alternate Attendant activates the DND feature at the Attendant station and affects all calls to the Attendant station.
2. A Flex button can be assigned to activate Alternate Attendant. The {ALT ATTENDANT} button LED indicates the status of the Alternate Attendant feature, On: Attendant unavailable.
3. A station, which is receiving calls forwarded from the System Attendant, cannot use the Alternate Attendant feature.
4. All except for one attendant can activate Alternate Attendant. When the last Attendant attempts to activate this feature, error tone is received.
5. An Attendant forwarded to an unavailable Attendant is also considered to be in the unavailable Attendant mode.
6. When there is a queued Attendant call, unavailable Attendant stations [HOLD] button will flash but no audible ring is provided and the station cannot retrieve the call. When an Attendant changes from unavailable to available status, any queued Attendant calls will be available to

INTERCOM STEP CALL

INTERCOM STEP CALL
Description
When busy tone is received on a dialed Intercom call, the user may place a call to another station by dialing the last digit of the station number. The system replaces the last digit of the previously dialed busy station with the dialed digit and places an Intercom call to the new station number.
Operation iPECS Phone To activate step call, while receiving busy on a dialed Intercom call;
1. Dial a digit other than the last digit of the busy station’s intercom number.
Conditions
1. If the user dials the last digit of the busy station, Camp-On will be activated.
2. After receiving busy tone, if the user takes no action for

CO RING DETECT

CO RING DETECT
Description
The system incorporates timers for Ring-on and Ring-off durations to assure proper alerting. When the duration of the ring signal exceeds the Ring-on timer, alerting will start. When the ring is not present for a period exceeding the Ring-off timer, alerting will stop. This allows the system Ring cycle detection to be matched to the many and varied PBX systems.
Operation System Operation of Ring detect is automatic.
Conditions
1. Ring On and Ring Off are assigned on a system basis.
2. The CO Ring Detect is applied to analog CO Lines only.

CONFERENCE

CONFERENCE
Description
SIP Phone who has 3-way conference capability can make conference call by Phone itself without utilize system’s conference feature. Also, SIP Phone can utilize system’s conference feature – Conference Room and Conference Group. To serve system conference for SIP Phone, a mixing device - MCIM is required.
Operation
SIP Phone Self Conference
1) Make a call and connected
2) Press ‘Conference’ button
3) Dial 2nd call and connected
4) Press ‘JOIN’ button Or,
1) Make a call and connected
2) Press ‘HOLD’ button
3) Dial 2nd call and connected
4) Press ‘3-way Conference’ button

The System Attendant can cancel features

FEATURE CANCEL
Description
The System Attendant can cancel features such as DND, Call Forwarding and Pre-defined or Custom Messages that are active at other stations.
Operation System Attendant To deactivate DND/Call Forward/Pre-selected Message for other stations
1. Press the [PGM] button.
2. Dial ‘052’, Attendant Station Program code.
3. Dial the desired station range or the same station number twice for a single station.
4. Press the [SAVE] button, confirmation tone is heard and Attendant station returns to idle status.

Feature Numbering Table

Feature Numbering Table (available while a dial tone is heard)]
Feature Numbering Additional Number Absent Message (set/cancel) 75 (1–6 [+ parameter]/0) + #/0 Account Code Entry for an SLT or 49 account code + outside phone no. Built-in Voice Message (BV) (record/play/ erase) 725 ([1 + extn. no.]/2/0) + #/0 Call Forwarding (FWD)—All Calls, Busy/No Answer (set/cancel) 71 ([1 or 2 + extn. no.]/0) + #/0 Call Forwarding (FWD)—All Calls, Busy/No Answer to BV (set/cancel) 71 ([1 or 2 + 725]/0) + #/0 Call Forwarding (FWD)—All Calls, Busy/No Answer to Voice Processing System (VPS) (set/cancel) 71 ([1 or 2 + extn. no.]/0) + #/0 Call Forwarding (FWD)—Follow Me (set/ cancel) 71 (5/8) + extn. no. + #/0 Call Forwarding (FWD)—To Outside (CO) Line (set/cancel) 71 (3 + outside (CO) line access no. + outside phone no. + #)/(0 + #/ 0) Call Hold (Hold Mode 2 or 3) for an SLT 20 Call Hold Retrieve for outside (CO) line calls/ intercom calls 53/5 outside (CO) line no. (1–8)/extn. no. Call Log, Incoming in the Common Area— CLEAR ALL 70 # Call Log, Incoming in the Personal Area— CLEAR ALL 70 # Call Park/Call Park Retrieve 22/52 0–9 Call Log Display Lock, Incoming 77 0000–9999 (2 times/one time) + # Call Pickup, Directed 4 extn. no. Call Pickup, Group 40 Call Pickup Deny (set/cancel) 72 (1/0) + #/0 Call Waiting for intercom calls/doorphone calls (set/cancel) 732 (1/0) + #/0 Call Waiting for outside (CO) line calls (set/ cancel) 731 (1/0) + #/0 Common BV Outgoing Message (OGM) (record/play/erase) 722 01–24 + (1/2/0) + #/0 Data Line Security (set/cancel) 730 (1/0) + #/0 Do Not Disturb (DND) (set/cancel) 71 (4/0) + #/0 Doorphone Call/Door Open 31/55 1–4

LINKED STATION PAIRS

LINKED STATION PAIRS
Description
A station can be logically linked to a primary station so that the two stations function as a single station. When linked, the two stations effectively act as a single station with the station attributes of the primary station. The status of one station is reflected in the status of the other and features activated at one are active at the other. All internal or external calls to a linked pair station will ring both stations.
All features available to the primary station are available and controllable by the secondary station, one station may activate Call Forward and the other may cancel the forward. The displays of the linked stations will display the status of the linked pair. When one is busy, the display of the linked station will be as shown below.
IN USE AT LINK STA
Operation System Operation of Linked pairs is automatic when defined
Conditions
1. Any combination of iPECS Phones and SLTs may be assigned as Linked pairs. However, a DSS Console may not be assigned as a linked pair station.
2. Intercom calls to the Linked stations always signal in the Tone ring mode and cannot be changed using the Caller Controlled ICM Signaling feature.
3. Linked pair stations are treated as having a single station number for all features including LCD displays, station programming, ADMIN access, ACD statistics, SMDR, etc.
4. The station attributes of the Secondary station will follow attributes of the Primary station, i.e. Day/Night COS, CO Warning Tone, CO Auto Hold, CO Call Drop, DID Call Waiting, Speed Access, Alarm, VSF Access, DND, FWD, Paging, CO Line Access, CO Ring Assign, etc.
5. If one station of a Linked pair is busy, the other station of the linked pair is also considered as busy, thus use of the linked station to place a call is not supported.
6. A station can be linked to another station without registration to the system. This allows a station to be linked without affecting the overall capacity of the system. In this case, only an iPECS phone, Phontage or SLT attached to an SLTM2 can be used as the unregistered linked station. In other cases, the linked station must be registered with the system, reducing the system capacity by one.
7. Linked pair stations cannot connect with each other between Master and Slave but ring for indication.
8. Linked pair cannot transfer a call with each other. But if Master(or slave) transfer to Slave(or Master) and on hook, you can recall and hold the call in slave(or Master).

SAVE NUMBER REDIAL (SNR)

SAVE NUMBER REDIAL (SNR)
Description
The last dialed number on a CO/IP call may be stored (up to 48 digits) in a buffer for future redial. This number is saved in memory until the user requests a new number be stored. Numbers dialed for subsequent calls do not affect the Save Number buffer.
Operation iPECS Phone To save a dialed number, while on a CO/IP call
1. After dialing and before hanging up, press the [SAVE] button.
To save a dialed number, while on a CO/IP call using the LIP-8000 menu
1. After dialing and before hanging up, press the [RIGHT NAVIGATION] button.
2. Locate and press the [SAVE] soft button
To dial a saved number
1. Lift the handset or press the [SPEAKER] button.
2. Press the [SPEED] button.
3. Dial #.
To save a dialed number, while on a CO/IP call using the LIP-8000 menu
1. Press the [DIR] soft button.
2. Press the [SPEED] soft button.
3. Dial ‘#’.
Conditions
1. The saved number can be a maximum of 48 digits.
2. Dialing the saved number will automatically seize the CO/IP line that was used for the original call. If the CO/IP line is busy, a CO/IP line from the same group will be selected and the saved number dialed. If all CO/IP lines from the group are busy, the user will receive All Lines busy tone and may queue
3. If user presses the [SAVE] button after seizing a CO/IP line without dialing, the Save Number Redial buffer will be erased.
4. If there is no {CO}/{IP} button, the call will be presented on a {POOL}, or {LOOP} button.
5. Save Number Redial is protected from power failure.
6. Manually dialing a Flash during a CO call will cause only those digits after the Flash to be stored and re-dialed as the Save Number Redial.

Drop Key

Drop Key
The Drop Key abandons a call while retaining the PBX/Centrex line to originate another call. The Drop key is provided by programming a Programmable Function key. This feature allows Flash key to be used to provide a hookflash to the PBX or Central Office. A Single Line Terminal user can use the Drop key function by an access code.

Group Call Pickup

Group Call Pickup
Group Call Pickup allows an extension user to answer a call ringing another extension in a Pickup Group. This permits co-workers in the same work area to easily answer each other’s calls. The user can dial a code or press a programmed Group Call Pickup key to intercept the ringing call. If several extensions within the group are ringing at the same time, Group Call Pickup intercepts the call based on the extension priority in the Pickup Group.
With Group Call Pickup, a user can intercept the following calls:
• A call ringing the user’s own pickup group
• A call ringing another pickup group when the user knows the group number
• A call ringing another pickup group when the user does not know the group number
There are 32 Call Pickup Groups available.
Conditions
• A Call Pickup Group cannot have an associated name.
• Group Call Pickup can be used to answer calls recalling from Hold or Park.
• Group Call Pickup cannot be used to answer calls ringing Call Arrival Keys or Virtual Extensions.
• Virtual Extensions can use Group Call Pickup to answer calls ringing a Multiline Terminal or Single Line Terminal.
• Users can pickup calls regardless of their access map programming.
• Directed Call Pickup provides another way of answering a co-worker’s call.
• Function keys simplify Group Call Pickup operation.

SIP Service

SIP Service
Description
When assigned to support SIP (Session Initiation Protocol), VoIP channels provide protocol conversion between SIP and the iPECS protocol or H.323. This permits the VoIP channel to connect to external SIP networks for call services. In addition, to the IETF RFC-3261 Session Initiation protocol draft standard, iPECS VoIP channels support other SIP related RFCs including:
RFC-2617 HTTP Authentication, Basic & Digest RFC-3515 Refer Method RFC-3264 Offer/Answer Model RFC-3265 SIP Basic Call Flow Examples RFC-3891 SIP “Replaces” Header
Using the SIP database assignments, the system will register and authenticate with the SIP proxy server permitting the system to interoperate employing SIP to establish, manage and terminate real-time voice sessions with external parties.

DID CALL WAIT

DID CALL WAIT
Description
If DID call is incoming to a station that is already call connected, this DID call is wait until the station is answering or the DID/DISA no answer timer is expired. To activate this feature, the ADMIN field of DID call wait must be set enable. If station has a flexible button of DID call wait, then it can set this feature via flexible button.
Operation iPECS Phone To assign a {DID CALL WAIT} button: [PGM] + {FLEX} + [PGM] + ‘34’ + [SAVE]
To activate/deactivate DID call wait from an iPECS Phone:
1. Press the {DID CALL WAIT} button.
2. Dial activate/deactivate code, ‘1’ or ‘0’ respectively.
Conditions
1. The DID call will follow the call routing defined in PGM code 167 after the expiration of the DID/DISA no answer timer expires.
2. The iPECS Phone must have an appearance button for the DID line.
3. Assigning the ICLID Timer, which enables ICLID routing, for a DID line, disables DID Call Wait.

ANSWERING MACHINE EMULATION

IPECS PHONE  ANSWERING MACHINE EMULATION
Description
When a call is sent to a Voice mailbox, the associated station can be assigned to notify the user and allow the user to screen the call. Two methods of notification and call screening are provided, Ring or Speaker mode.
In the Ring mode, the user is notified by flashing of the AME (Answering Machine Emulation) Flex button. The user may press the Flex button to hear the caller as the voice message is stored. In the Speaker mode, when the call is sent to the Voice Mailbox, the caller’s voice is automatically broadcast over the speaker of the user’s iPECS Phone.
The user may terminate the screening leaving the caller in voice mail to record a message, talk with the caller and record the conversation in the mailbox, or answer the call and disconnect the Voice Mail.
The user’s iPECS Phone must be assigned with an AME Flex button for proper operation.

DTMF Type

DTMF Type
1. MFIM currently support only INFO type DTMF for SIP Extension (does not INBAND and
2833).
☞ for SIP Trunk, all type of DTMF is supported.
2. If SIP Extension talking voice path is connected directly each other then they can implement INBAND or 2833 DTMF independently.
3. Synchronization between system and SIP Phone is required.
- SIP Data / SIP Phone Attributes(211) – DTMF Type : One of INFO (default: DTMF RELAY)
- SIP Phone Self Programming - DTMF Type : INFO or a specific INFO type if there is in select list.
4. By Provisioning for LG-Ericsson SIP Phone, DTMF Type of Phone is automatically set to INFO Type. In that case, you do not need to set it by SIP Phone Self Programming.
Network Related Configuration
1. MFIM IP address for WAN is different from original IP address for LAN
- Condition 1 : MFIM has firewall IP address or MFIM is port-forwarded by VOIM WAN-U or other Switches
- Condition 2 : There is a SIP Phone in WAN side of MFIM
- Resolution : SIP Data / SIP Phone Attributes(211) – Same Zone with MFIM - OFF
- Implementation : MFIM will implement signaling with its WAN IP address for the SIP Phone.
2. A SIP Phone is on NAT environment (for example, wireless SIP Phone via AP)
- Condition 1 : The NAT IP address or IP Port of SIP Phone (WAN address of SIP Phone) is frequently updated.
- Condition 2 : Sometimes communication is disconnected, unreachable or mismatched because of so-often updated address by NAT mapping implementation.
- Resolution 1 : Enable the Keep Alive option for the SIP Station(s) that are on NAT environment.
- Implementation 1: MFIM will send ‘OPTIONS’ message so often (in 30 seconds) to assist to maintain the address of SIP Phone’s WAN.
- Resolution 2 : set static NAT address mapping by port-forwarding. For example 1 to 1 static NAT address assignment by port-forwarding in NAT switch.
- Implementation 2: SIP Phone’s WAN address will not be changed.

The System Attendant can place CO/IP lines out-of service,

DISABLE OUTGOING CO/IP ACCESS
Description
The System Attendant can place CO/IP lines out-of service, disabling outgoing calls on the CO/IP path. This is normally done should an undetected fault interrupt service on a CO/IP path. Incoming calls continue to be processed normally.
Operation System Attendant To disable/enable Outgoing CO/IP access (toggle)
1. Press the [PGM] button.
2. Dial ‘072’, the Attendant Station Program code.
3. Press the {CO} button of the line(s) to be disabled, confirmation tone is heard and the status for the selected line(s) is changed.
Conditions
1. If the desired CO/IP line is in use, the System Attendant may still disable the CO/IP line. The feature will take effect after the desired CO/IP line goes to idle.
2. Once the line is disabled, all Attendant appearances for the disabled CO/IP line will flutter at 240 ipm, other stations will indicate the CO/IP line as busy, LED is On.
3. The CO/IP line outgoing access status is stored in battery-protected memory in case of a power failure.
4. Multiple CO/IP lines may be enabled/disabled without redialing the Attendant Station Program code. Confirmation tone is heard after each CO/IP line is enabled/disabled.
5. When the system detects a fault on an analog CO line, the CO line is disabled for outgoing access automatically.
6. Incoming calls on a disabled CO/IP line will continue to operate normally.

CALL PARK

CALL PARK
Description
SIP Phone user can park a talking call to system’s Call Park Location. And the other Station user can retrieve the parked call. The parking operation is ‘Blind Transfer’.
Operation Call Parking
1) SIP Phone user is on a CO talking.
2) SIP Phone user park the talking call and the call will be parked to system’s Call Park Location
- press ‘Blind Transfer’ button
- dial Park Bin number
- [SEND]
3) The SIP Phone user will page to inform somebody to pickup the call.
4) Somebody will retrieve the parked call.

STATION RELOCATION

STATION RELOCATION
Description
The iPECS Phone once registered can be re-located to any LAN port connected to the iPECS system without loss of any data or programming.
Operation
This feature is automatically activated.

Agent Automatic Wrap-Up

Agent Automatic Wrap-Up Description
When an Agent completes an ACD group call, the Agent automatically enters into the Wrap-up state. In this state, an Agent will not receive ACD calls, allowing the Agent to complete paperwork, etc. The Agent remains in this automatic Wrap-Up state for the duration of the ACD group’ Wrap-Up Timer. After the Wrap-Up Timer or by using ‘Wrap-Up-End’ feature, the Agent returns to available status and can receive ACD group calls.
Operation
Agent iPECS Phone
To assign a {WRAP-UP-END} button; [PGM] + {FLEX} + ‘585’ + [SAVE]
Activation;
Automatic when Agent completes an ACD group call
Deactivation;
1. Automatically after Wrap-Up Timer. Or,
1. Dial ‘585’ the Wrap-Up-End code. Or,
1. Press {WRAP-UP-END} flexible button, before expiration of the Wrap-Up Timer.

ALARM SIGNAL/DOOR BELL

ALARM SIGNAL/DOOR BELL
Description
The system can be configured to recognize the status of an external contact (normally open or closed). The system will signal assigned iPECS Phones when the contact activates. This capability is commonly employed to provide remote Alarm or Door Bell signals to the user.
Assigned stations receive the Alarm Signal, either a single tone burst repeated at 1-minute intervals or a continuous tone. The Alarm Signal may be terminated at the user’s phone by dialing the Alarm Stop code or, if assigned, pressing the {ALARM STOP} button. To rearm the Alarm function, the alarm condition must be cleared and the Alarm signal terminated.
When used as a Door Bell, assigned iPECS Phones receive a single tone burst each time the external contact is activated and no reset is required.

SIP EXTENSION REGISTRATION

SIP EXTENSION 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
4. CODEC
5. Call Blocking … and so on

SIP Phone

CALL ANSWER
Description
A SIP Phone will accept a call when it receives call on idle state.
On busy state, it is different according to ‘Call Wait’ supported or not.
Operation
Receives 1st call on Idle State
1) Accept the call and respond with Ringing Pickup Handset to answer the call
Receives 2nd call on Busy State
1) If do not support Call Wait or disabled Reject the call by ‘486’ busy
2) If support Call Wait and enabled
Accept the call and respond with Ringing The call is waited on the SIP Phone
SIP Phone will serve self indication for the second call.
Handsfree Automatic Answer
1) By SIP Phone : self programmable option, always
2) By System : in case of receiving call by Voice Over, Intrusion, Forced Handsfree, Paging
- LG-Ericsson SIP Phone only supported

CO/IP AUTO FAULT DETECTION

CO/IP AUTO FAULT DETECTION AND RECOVERY
Description
If a CO line fault is reported from a PRI/VoIP/SIP gateway, the iPECS places the CO Line in an Out-Of-Service state and places it in the Unused CO/IP Group. Upon receiving the CO Line recovery report, the CO Line is automatically restored. The fault is also reported to the iPECS NMS when configured.
Operation System Operation of Fault Detection and Recovery is automatic.
Conditions
1. The “Unused” CO Group contains CO line numbers that are not used or are temporally blocked.

DID CALL WAIT

iPECS Phone w/Display
DID CALL WAIT
Description
If DID call is incoming to a station that is already call connected, this DID call is wait until the station is answering or the DID/DISA no answer timer is expired. To activate this feature, the ADMIN field of DID call wait must be set enable. If station has a flexible button of DID call wait, then it can set this feature via flexible button.
Operation iPECS Phone To assign a {DID CALL WAIT} button: [PGM] + {FLEX} + [PGM] + ‘34’ + [SAVE]
To activate/deactivate DID call wait from an iPECS Phone:
1. Press the {DID CALL WAIT} button.
2. Dial activate/deactivate code, ‘1’ or ‘0’ respectively.
Conditions
1. The DID call will follow the call routing defined in PGM code 167 after the expiration of the DID/DISA no answer timer expires.
2. The iPECS Phone must have an appearance button for the DID line.
3. Assigning the ICLID Timer, which enables ICLID routing, for a DID line, disables DID Cal

change the ICM Signaling mode

change the ICM Signaling mode;
1. Place intercom call.
2. Dial ‘#’, ICM Signaling mode will change from Voice announce to Tone ring or Tone ring to
SLT
Voice announce.
To change the ICM Signaling mode;
1. Place intercom call as normal.
2. Dial ‘#’, ICM Signaling mode will change from Voice announce to Tone ring or Tone ring to
Voice announce.
Conditions
1. The ICM Signaling mode cannot be changed when an Intercom call is placed to a Linked Pair
station.
2. If the signaling mode is changed, the call is not subject to Call Forward, No Answer.
3. The signaling mode for a specific Intercom call can only be changed once and cannot be
changed back to the original signaling mode.
4. Changing the signaling mode does not affect privacy at the called station.

Auto Attendant

Auto Attendant
This panel gives a summary of the auto attendant services (up to 9) currently configured. For each configured auto attendant, the current service being provided by the auto attendant is shown and the hours for that service. Each auto attendant can be configured with different greeting and options for morning, afternoon, evening and out of hours periods.
The edit icon can be used to access the Auto Attendants menu.
Groups
54
This panel shows a summary of which hunt groups have been configured. The edit icon can be used to access the
52
Groups menu.
· Calling Group 181

This type of group can be set as the Coverage Destination of a trunk or trunk channel. It can be also the destination of calls routed using DID call mapping or call-by-call settings. Calling Group 1 is also used by the Simultaneous Page function (*70).
· Hunt Group 181

This type of group can be set as the Coverage Destination of a trunk or trunk channel. It can also be the destination of calls routed using DID call mapping or call-by-call settings.
· Night Service Group 181

When the system is put into night service, this group overrides the Coverage Destination of all trunks.
· Operator Group 181

This option is only available for systems with their Mode set to PBX. By default the group contains the first extension on the system. For PRI and BRI trunks, it is fixed incoming destination for calls unless DID Mapping is applied to the call. It can also be selected as the destination for incoming SIP calls.
· Pickup Group 181

Users can be pickup calls currently alerting any member of a pickup group. They do not need to be a member of the group.
Calling List
This panel shows a summary of the lists that are used to control which numbers users can dial when making outgoing
calls. The edit icon can be used to access the List Management menu to edit the settings.
· Allowed List
62
Allowed lists are used to enter numbers or types of numbers that users associated with the list can dial even if they are restricted from dialing other numbers. Up to 8 such lists, each containing 10 numbers, can be configured.
· Disallowed List
Disallowed lists are used to enter numbers or types of numbers that users associated with the list cannot dial. Up to 10 such lists can be configured. Up to 8 such lists, each containing 10 numbers, can be configured.
· Emergency Number List
This list is used to enter numbers that all users can dial at any time regardless of any other settings that might restrict them from dialing numbers for outgoing calls. Up to 10 numbers can be configured in this list.
· Account Codes
Up to 99 account codes can be entered. In addition selected users can be configured to have to enter an account code whenever they make an outgoing external call.

DND OVERRIDE

DND OVERRIDE
Description
A station in the DND mode generally cannot receive an incoming call. The Attendant and the Secretary station of an Executive/Secretary pair however may override the DND status to signal the station of an awaiting call.
Operation Attendant To activate DND Override while receiving DND tone
1. Press the {ATD INTRUSION} button, the call signals at DND station.
Conditions
1. An Attendant may use Override to transfer a CO/IP call to a station in DND.

INTERCOM SIGNALING MODE

INTERCOM SIGNALING MODE
Description
Each iPECS Phone can select the signaling mode used for incoming ICM calls while the station is idle. There are three signaling modes available.
H - Call announcing with Handsfree answerback:
When an ICM call is received, the user receives splash tone followed by the ICM caller’s voice. The user may respond to the caller without the need to ‘Lift the handset’ or ‘press the [SPEAKER] button’.
P - Call announcing with Privacy:
When an ICM call is received, the user receives splash tone followed by the ICM caller’s voice. To respond the user must lift the handset or press the [SPEAKER] button.
T - Tone ringing:
An ICM call will cause the iPECS Phone to provide audible ICM ring tone. The user must lift the handset or press [SPEAKER] to answer. An SLT always functions in the Tone ring mode.
Operation iPECS Phone To change ICM Signaling Mode
1. Press [PGM] button, the [SPEAKER] button LED lights steady.
2. Dial Station User Program code ‘13’, confirmation tone is received.
3. Dial the desired ICM Signaling Mode code (‘1’ for H, ‘2’ for T or ‘3’ for P).
4. Press the [SAVE] button.
Conditions
1. Message Wait, Callback, Call Forward and Attendant Override will ring in the tone mode, regardless of ICM Signaling Mode selected by the user.
2. The ICM signaling Mode Selection does not affect Page announcements.
3. The default ICM Signaling mode is Tone ring and the active mode is stored in battery- protected memory.

Alarm

Alarm
Alarm lets any station extension work like an Alarm clock. An extension user can have Alarm remind them of a meeting or an appointment. There are two types of Alarms:
• Alarm 1 (sounds only once at the preset time)
• Alarm 2 (sounds every day at the preset time)
Conditions
• Single Line Terminals ring and Music on Hold is heard when the Alarm sounds.
• Only a Multiline Terminal user can view what time the Alarm is currently set for.

NAME ENTRY

NAME ENTRY

A SLT user has the capability to program the user name so that a calling user with an LCD can see the name instead of the station number.
Operation SLT
To register the name at the SLT
1. Lift the handset.
2. Dial ‘561’, the SLT Programming code, confirmation tone is heard.
3. Dial ‘74’, the SLT Name Program Code.
4. Enter name, refer to Station Speed Dial, Alphanumeric Chart.
5. Momentarily depress the hook-switch, receive confirmation tone.
To delete the name at the SLT;
1. Lift the handset.
2. Dial ‘561’, the SLT Programming code, confirmation tone is heard.
3. Dial ‘74’, the SLT Name Program Code.
4. Momentarily depress the hook-switch, receive confirmation tone.

PULSE SIGNALING

PULSE SIGNALING
Description
An analog CO line will send dial pulse signals to the central office. If programmed as a pulse CO line, the system will send open loop pulses at 10 pps with the assigned break/make ratio.
Operation System Operation of this feature is automatic when programmed.
Conditions
1. The break/make ratio is system programmable as 60/40 or 66/33.
Programming
CO/IP 1. CO Line Signal Type (PGM 141-Btn 5)
SYSTEM 1. Pulse Dial Ratio (PGM 176)
Related Features
2. Pause Time

LINE PRESET FORWARD

LINE PRESET FORWARD
Description
Each CO/IP Line can be assigned a Ring-No-Answer Preset Forward destination. An incoming call on the CO/IP line will be routed to the defined ring destination. At expiration of the CO/IP Line Preset No Answer Forward timer, the call is forward to the defined Preset Forward destination, which is an index to the ICLID Ring Assignment Table.
The destination can be a station or station group including an adjunct Voice Mail. When the call is forwarded to an adjunct Voice Mail group, a predefined Voice Mail Id (VMID) will be sent to the VM system to identify the Mailbox to receive the call.
Operation System Operation of this feature is automatic.
Conditions
1. CO/IP line Preset Forward is disabled for calls initially routed to a station group.
2. CO/IP line Preset Forward will override Call Forward No-Answer at a station.
3. CO/IP line Preset Forward is disabled if the Preset Forward Timer is set to 0.
4. The CO/IP line Preset Forward destination cannot be the VMIM/VSF group.

CALL RELEASE

AUTO CALL RELEASE
Description
CO/IP or intercom calls (except Hands-free Answerback), will be released automatically if the user does not complete dialing or, for intercom calls, the called party does not answer after a pre- determined time.
Operation System Auto Call Release of Intercom calls: If a station places an intercom call and the called station does not answer in the Intercom Call Release Time, the call is terminated and the calling user receives error tone.
Auto Call Release of CO/IP calls:
If a station seizes an idle CO/IP Line and does not dial within the CO/IP Call Automatic Release Time, the call is terminated and the user will receive error tone.
Conditions
1. If the Automatic Call Release Timer is set to ‘0’, Auto Call Release is disabled.
2. When the handset is used to place a call, the user will receive error tone for 30 seconds followed by 30 seconds of Howler tone and the station is placed in a fault mode. If on-hook dialing is used, the station receives error tone for one (1) second and returns to idle automatically

Serial DSS

SERIAL DSS
Description
LIP-8000 keyset supports Serial DSS to be connected through serial cable.
There are 3 types of serial DSS ; 12 button Serial DSS(LIP-8012DSS), 12 button Serial DSS with LCD(LIP-8012LSS), 48 button Serial DSS(LIP-8048DSS).
Operation
Connecting Serial DSS to LIP-8000 keyset
1. Check the station have the authority to use serial DSS in Admin.
2. Check the serial DSS power is off (only for LIP-8048DSS).
3. Power off the LIP-8000 keyset.

ACCOUNT CODE

ACCOUNT CODE
Description
Station users may allow tracking of specific calls by entering a non-verified variable length (up to 12 digits) identifier for a call. The identifier or “Account Code” is output as part of the Station Message Detail Record (SMDR) for the call.
Operation iPECS Phone To assign a Flex button for {ACCOUNT CODE} operation: {ACCOUNT CODE} button:
[PGM] + {FLEX} + [PGM] + ‘84’ + [SAVE] {ONE-TOUCH ACCOUNT CODE} button:
[PGM] + {FLEX} + [PGM] + ‘84’ + Account code (up to 12 digits) + [SAVE]
To enter an Account Code using an {ACCOUNT CODE} button prior to placing a call:
1. Lift the handset.
2. Press the {ACCOUNT CODE} button.
3. Dial the Account Code (1 to 12 digits).
4. Press ‘*’, Intercom dial tone is heard.
5. Place the CO/IP call as normal. Or,
1. Lift the handset or press the [SPEAKER] button.
2. Press the {ONE-TOUCH ACCOUNT CODE} button.
3. Place the CO/IP call as normal.
To enter an Account Code using an {ACCOUNT CODE} button during a call:
1. Press the {ACCOUNT CODE} button.
2. Dial the Account Code (1 to 12 digits).
3. Press ‘*’. Or,
4. Press the {ONE-TOUCH ACCOUNT CODE} button
SLT
To enter an Account Code prior to placing a call:
1. Lift the handset.
2. Dial Flex Numbering Plan code ‘550’.
3. Dial the Account Code (1 to 12 digits).
4. Press ‘*’.
5. Place the CO/IP call as normal.

ALARM SIGNAL/DOOR BELL

ALARM SIGNAL/DOOR BELL
Description
The system can be configured to recognize the status of an external contact (normally open or closed). The system will signal assigned iPECS Phones when the contact activates. This capability is commonly employed to provide remote Alarm or Door Bell signals to the user.
Assigned stations receive the Alarm Signal, either a single tone burst repeated at 1-minute intervals or a continuous tone. The Alarm Signal may be terminated at the user’s phone by dialing the Alarm Stop code or, if assigned, pressing the {ALARM STOP} button. To rearm the Alarm function, the alarm condition must be cleared and the Alarm signal terminated.
When used as a Door Bell, assigned iPECS Phones receive a single tone burst each time the external contact is activated and no reset is required.

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.

INTERCOM LOCK-OUT

INTERCOM LOCK-OUT
Description
If the user takes no action after going off-hook for the Dial Tone timer or fails to dial an additional digit within the Inter-digit timer, the station will receive error tone for 30 seconds and be placed out- of-service (locked-out). The LED of associated {DSS/BLF} buttons as well as the station’s [ICM] button flutter rapidly to indicate the out-of-service state.
For iPECS Phone users, if the [SPEAKER] is used, the station will receive error tone for 30 second and then automatically return to idle.
Operation System Operation of Intercom Lockout is automatic based on the Dial Tone & Inter-digit timers.
3-4

IP TRUNKING

IP TRUNKING 4.13.1 H.323 v4 Service
Description
When assigned to support H.323 protocol, VoIP channels provide protocol conversion between H.323 v4 and the iPECS protocol or SIP. This permits the VoIP channel to connect to external H.323 networks or terminals and to support H.323v4 supplementary services. In addition, H.323 VoIP channels can register with an external H.323 GateKeeper to support Gatekeeper call routing.
Supplementary services are supported employing H.450.1 ~ H.450.12 standards, which define the following supplementary services: H.450.1 H.450.2 H.450.3 H.450.4 H.450.5 H.450.6 H.450.7 H.450.8 H.450.9 H.450.10 H.450.11 H.450.12

INTERCOM TENANCY GROUP

INTERCOM TENANCY GROUP
Description
Stations in the iPECS are assigned to an Intercom Tenancy Group, group 00 ~ 15. Stations in a group are allowed or denied the ability to place intercom calls to stations in other groups on a group-by-group basis.
Each Intercom Tenancy Group is assigned an Attendant station. All dial ‘0’ calls from a station in the group are routed to the assigned Attendant. In addition, the assigned Attendant can control the Day/Night Ring mode for stations in the group switching from Day to Night mode. Each group is assigned a separate Auto Ring Mode Table to change the Ring and COS mode automatically during the day and night service mode. In addition, DID calls to the system can be routed to a specified Intercom Tenancy group. By defining the group as the destination in the Flexible DID Conversion table calls will follow the Tenancy Group Auto Ring Mode table.
Operation System Operation of Intercom Tenancy Groups is automatic when programmed.
Conditions
1. Intercom calls from a station to a denied access Intercom Tenancy Group will return error tone.
2. Intercom Tenancy does not affect the Station Numbering Plan in the system. All stations in the system must have different station numbers even if they are assigned to different Intercom Tenancy groups.
3. The Attendant of an ICM Tenancy Group can be any station in the system and it is not affected by ICM Tenancy Group access.
4. When the Attendant of an ICM Tenancy Group sets Day/Night/Weekend mode, it will affect only the assigned ICM Tenancy Group.
5. Calls to and from CO/IP Lines are not affect by ICM Tenancy however; calls can not be transferred between groups if access is not allowed between the groups.
6. Intercom Tenancy Group 00 is the default or unassigned group. Stations assigned to group 00 are unaffected by Intercom Tenancy and can place and receive calls with stations of all other groups.

INTERCOM TENANCY GROUP

INTERCOM TENANCY GROUP
Description
Stations in the iPECS are assigned to an Intercom Tenancy Group, group 00 ~ 15. Stations in a group are allowed or denied the ability to place intercom calls to stations in other groups on a group-by-group basis.
Each Intercom Tenancy Group is assigned an Attendant station. All dial ‘0’ calls from a station in the group are routed to the assigned Attendant. In addition, the assigned Attendant can control the Day/Night Ring mode for stations in the group switching from Day to Night mode. Each group is assigned a separate Auto Ring Mode Table to change the Ring and COS mode automatically during the day and night service mode. In addition, DID calls to the system can be routed to a specified Intercom Tenancy group. By defining the group as the destination in the Flexible DID Conversion table calls will follow the Tenancy Group Auto Ring Mode table.
Operation System Operation of Intercom Tenancy Groups is automatic when programmed.
Conditions
1. Intercom calls from a station to a denied access Intercom Tenancy Group will return error tone.
2. Intercom Tenancy does not affect the Station Numbering Plan in the system. All stations in the system must have different station numbers even if they are assigned to different Intercom Tenancy groups.
3. The Attendant of an ICM Tenancy Group can be any station in the system and it is not affected by ICM Tenancy Group access.
4. When the Attendant of an ICM Tenancy Group sets Day/Night/Weekend mode, it will affect only the assigned ICM Tenancy Group.
5. Calls to and from CO/IP Lines are not affect by ICM Tenancy however; calls can not be transferred between groups if access is not allowed between the groups.
6. Intercom Tenancy Group 00 is the default or unassigned group. Stations assigned to group 00 are unaffected by Intercom Tenancy and can place and receive calls with stations of all other groups.

CO/IP AUTO FAULT DETECTION AND RECOVERY

CO/IP AUTO FAULT DETECTION AND RECOVERY
Description
If a CO line fault is reported from a PRI/VoIP/SIP gateway, the iPECS places the CO Line in an Out-Of-Service state and places it in the Unused CO/IP Group. Upon receiving the CO Line recovery report, the CO Line is automatically restored. The fault is also reported to the iPECS NMS when configured.
Operation System Operation of Fault Detection and Recovery is automatic.
Conditions
1. The “Unused” CO Group contains CO line numbers that are not used or are temporally blocked.
Programming
CO/IP 1. CO Line Group (PGM 141-Btn 1)
Related Features CO/IP Line Groups

To change the ICM Signaling mode;

iPECS Phone
To change the ICM Signaling mode;
1. Place intercom call.
2. Dial ‘#’, ICM Signaling mode will change from Voice announce to Tone ring or Tone ring to
SLT
Voice announce.
To change the ICM Signaling mode;
1. Place intercom call as normal.
2. Dial ‘#’, ICM Signaling mode will change from Voice announce to Tone ring or Tone ring to
Voice announce.
Conditions
1. The ICM Signaling mode cannot be changed when an Intercom call is placed to a Linked Pair
station.
2. If the signaling mode is changed, the call is not subject to Call Forward, No Answer.
3. The signaling mode for a specific Intercom call can only be changed once and cannot be
changed back to the original signaling mode.
4. Changing the signaling mode does not affect privacy at the called station.
Programming
STATION 1. Caller Controlled ICM Signaling (PGM 111-Btn 15)
Related Features
Intercom Signaling Mode Linked Station Pairs
Hardware
3.5 INTERCOM LOCK-OUT

AUTO CALL RELEASE

AUTO CALL RELEASE
Description
CO/IP or intercom calls (except Hands-free Answerback), will be released automatically if the user does not complete dialing or, for intercom calls, the called party does not answer after a pre- determined time.
Operation System Auto Call Release of Intercom calls: If a station places an intercom call and the called station does not answer in the Intercom Call Release Time, the call is terminated and the calling user receives error tone.
Auto Call Release of CO/IP calls:
If a station seizes an idle CO/IP Line and does not dial within the CO/IP Call Automatic Release Time, the call is terminated and the user will receive error tone.
Conditions
1. If the Automatic Call Release Timer is set to ‘0’, Auto Call Release is disabled.
2. When the handset is used to place a call, the user will receive error tone for 30 seconds followed by 30 seconds of Howler tone and the station is placed in a fault mode. If on-hook dialing is used, the station receives error tone for one (1) second and returns to idle automatically.

VOIP OR VOIM8/24

VOIP OR VOIM8/24 REQUIREMENT FOR SIP PHONE
Description
For SIP Phone, system should be equipped VOIP or VOIM8/24 module to serve below features.
1. CO Line for H323, Networking, SIP Trunking
2. DSP for generation of Busy/ Error/ Confirm/ Ring-Back/ Hold/ Page/ Warning/ OHVA/ Intrusion/ Dial tones from system to SIP Phone
3. Relay of Music On Hold from system to SIP Phone
4. Relay of Paging from/to SIP Phone
5. Voice RTP Packet Relay between private LAN and public WAN, local and remote, NAT resolution

SIP EXTENSION REGISTRATION

Feature Description & Operation

SIP EXTENSION
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
4. CODEC
5. Call Blocking

IP Terminals

IP Terminals 5600 Series
The 5600 series provides the flexibility and future-proof technology of an IP telephony system. These terminals connect to the local area network (LAN).
5610 Terminal

Part Code: 700381965
• Message waiting indicator
G.711, G.729a/B Voice CODECs
• QoS Options of – UDP port selection,
• diffserv and 802.1p/B (VLAN)
• Protocol (SNMP)
Support for simple network management
• Downloadable firm ware for future
Microsoft Netmeeting compatible
• upgrades
• wall mount stand
Wall mountable with included desk/
• Drop, Hold, Redial, Mute, Volume up and
9 F i x e d fe a ture ke y s: Conference, Tr ans fe r,
down, Speaker, Voicemail
• headset jack
Two way speaker phone and built-in
• English, French, Italian, Spanish and
Multiple language support built-in:
KataKana
• 168 x 80 greyscale display with a 5 line
8 personalised ring patterns
• display
• labels providing 12 logical DSS keys
6 physical DSS keys with 13-character soft
• address. Connects to IP Office via the LAN
DHCP client or statically configured IP
• Switched ports for connection of PC
Integrated full duplex 10/100 Base T Ethernet
• from appropriately programmed DSS keys,
IP Office interactive menus can be evoked
and these menus will utilise the 6 DSS keys and the ‘Exit’ fixed function key Auto-negotiation provided separately for
• each port
• • Phone has priority over PC port at all times
802.3 Flow Control
5621 Terminal Speed Dial, Call Log, Web Browser

Features as the 5610 plus:
• cavity for improved sound quality
Two way speaker phone with acoustic
• 24 Programmable Feature Keys Automatically labelled from the system
• (no paper labels)
• : Speaker, Mute, Hold,
6 fixed feature keys
Headset and volume Up/Down
• Large screen backlit 7 line display
• :
5 Fixed feature keys below the display
Conference, Transfer, Hold, Redial and Drop 4 Embedded applications
• :
SIP Terminals Support
10

• (WAP/WML), Options
• jack for use with the EU24, 24 button
Feature key module (FKM) interface
expansion module
• mount stand
7 position adjustable desk stand/wall
• Infrared (IrDA) port

Hotline

Hotline
Hotline gives a Multiline Terminal user one-button calling and Transfer to another extension (the Hotline partner). Hotline helps co-workers that work closely together. The Hotline partners can call or Transfer calls to each other just by pressing a single key.
The Hotline feature has two applications.
• Hotline (Hotline partner)
• Ringdown Extension, Internal/External (Refer to Ringdown Extension (Hotline), Internal/External on page 1-767.)
In addition, the Hotline key shows the status of the partner’s extension.
When the key is . . . The extension is . . . Note Off Idle On Busy or ringing Fast Flash DND - All calls (option 3) or Intercom calls (op- tion 2) Double Wink On ACD Agent logged onto the group (V1.5 Added) Wink Off ACD Agent logged off (V1.5 Added)
There are 84 hotlines available.
Conditions
• An extension user cannot use Hotline to pick up a call ringing their partner’s extension.
• If a station is an ACD agent, the Hotline key blinks to indicate the ACD agent’s status. (V1.5 or higher)
• Hotline keys can be assigned to the DSS consoles.
• Hotline does not override Do Not Disturb.
• Hotline always follows the Handsfree Answerback/Forced Intercom Ringing mode set at the called extension. The Hotline caller can override the setting, if desired.
• External Hotline automatically dials a telephone number or Speed Dial - System/Group/Station number when the handset is lifted.
• If the partner’s extension is busy, Hotline does not automatically activate Off-Hook Signaling.
• A Hotline is a uniquely programmed function key.

InMail Park and Page

InMail Park and Page
InMail Park and Page can automatically Park a call at an extension and Page the user with a recorded Paging Message announcing the parked call. The called extension user can then go to any telephone and implement Personal Park to pick up the call. With InMail Park and Page, InMail tries to locate the person instead of just sending the call to their mailbox. Additionally, there is no need for an operator or receptionist to manually answer the call, park it, and then try to track down the employee.
The Paging Message is usually recorded in the user’s own voice and typically says something like, “Mike Smart, you have a call.” If the Paging Message is not recorded for the extension, a built-in message announces the called party’s name or extension number (if the name is not recorded).
InMail Park and Page is available for all trunk calls that are redirected to voice mail via forwarding or overflow, including transferred calls, Direct Inward Lines, and Direct Inward Dialing. Park and Page is also available for Automated Attendant Screened (STRF) and Unscreened (UTRF) Transfers. Optionally, an extension can have calls from the Automated Attendant immediately Park and Page without trying their extension first.
When InMail Park and Page intercepts the call, it normally offers the caller three options:
1. Dial 1 to leave a message in the called extension’s mailbox. (The caller hears the mailbox Greeting, if recorded.)
2. Dial 2 to Park and Page.
(The caller returns to these options if the Park is not picked up.)
3. Dial 3 for other options.
(Normally, this routes to the extension’s Next Call Routing Mailbox.)
InMail Park and Page is available at Personal and Group Subscriber Mailboxes, and can be enabled through system programming or via the subscriber’s Mailbox Options Menu. InMail Park and Page is not applicable to Intercom calls.
Automated Attendant Direct to Voice Mail (DVM)
When an extension has Automated Attendant Direct to Voice Mail (DVM) enabled, all calls from the Automated Attendant go directly to the subscriber’s mailbox. The extension does not ring for Automated Attendant calls. The caller hears the mailbox greeting and can leave a message, but unlike Park and Page is not normally offered any other routing options. A subscriber typically turns on DVM when they need to work at their desk undisturbed by outside calls from the Automated Attendant.
DVM can be enabled by the installer from system programming or by the extension user from their Mailbox Options Menu.
Keep in mind that DVM does not block Intercom calls from co-workers or any other outside call not routed through the Automated Attendant. For example, with DVM enabled, Direct Inward Lines and transferred outside calls to an extension work normally.

GROUP LISTENING

GROUP LISTENING
Description
All iPECS Phones have a built in speaker. If allowed, users may employ the speaker to monitor a call while using the handset to converse with the outside party. This enables a group of people in the room to listen to both parties in the conversation.
Operation iPECS Phone While on a call using the handset
1. Press the [SPEAKER] button, speaker activates, the speakerphone microphone will be muted while the handset is off-hook.

IP ADDRESS DIALING

IP ADDRESS DIALING Description
If allowed, users may place calls using an IP path. The system accepts user dialed digits as the IP address for the called party. When dialing an IP call, the asterisk, ‘*’, is used as the “dot” between bytes of the IP address.
Operation iPECS Phone To place an IP Call
1. Lift the handset or press the [SPEAKER] button.
2. Press {IP GRP} button or dial IP Group access code.
3. Dial ‘xxx * xxx * xxx * xxx’, use ‘*’ as the dot in the IP address.
4. Press ‘#’ to complete dialing.
SLT
To place an IP Call
1. Lift the handset.
2. Dial IP Group access code.
3. Dial ‘xxx ‘*’ xxx ‘*’ xxx ‘*’ xxx’, use ‘*’ as the dot in the IP address.
4. Press ‘#’ to complete dialing.

DTMF SIGNAL SENDING

DTMF SIGNAL SENDING
Description
Dual Tone Multi-Frequency (DTMF) signals are used with CO lines assigned for DTMF signaling. The duration of the DTMF signal can be adjusted from 40 to 990 milliseconds.
Operation System Operation of this feature is automatic when programmed.
Conditions
1. The system mutes the user’s voice transmission to reduce interference while sending DTMF

ALTERNATE ATTENDANT

ALTERNATE ATTENDANT
Description
This feature allows an Alternate answer point while the Attendant station is in an unavailable mode. When in the unavailable mode, the next available Attendant in the Attendant group will receive Attendant calls and recalls.
Operation Attendant To assign a flexible button to activate {ALT ATD} button [PGM] + {FLEX} + ‘562’ + [SAVE]
To toggle Attendant Unavailable feature
1. Dial ‘562’, the Alternate Attendant code or press {ALT ATD}.
Conditions
1. Alternate Attendant activates the DND feature at the Attendant station and affects all calls to the Attendant station.
2. A Flex button can be assigned to activate Alternate Attendant. The {ALT ATTENDANT} button LED indicates the status of the Alternate Attendant feature, On: Attendant unavailable.
3. A station, which is receiving calls forwarded from the System Attendant, cannot use the Alternate Attendant feature.
4. All except for one attendant can activate Alternate Attendant. When the last Attendant attempts to activate this feature, error tone is received.
5. An Attendant forwarded to an unavailable Attendant is also considered to be in the unavailable Attendant mode.
6. When there is a queued Attendant call, unavailable Attendant stations [HOLD] button will flash but no audible ring is provided and the station cannot retrieve the call. When an Attendant changes from unavailable to available status, any queued Attendant calls will be available to the Attendant.

SLT MESSAGE WAIT INDICATION

SLT MESSAGE WAIT INDICATION
Description
All SLT devices will receive a “Stutter” dial tone as an audible Message Wait Indication. In addition, industry standard Message Waiting telephones may be connected to the system. Software will cause the lamp to flash when a messaging is waiting.
Operation System The system switches the 90 VDC lamp On and Off for assigned SLTs indicating a Message Wait.
Conditions
1. The system switches a 90 VDC supply On and Off to flash the SLT’s neon lamp.
2. Although the SLT Battery Feed is removed during the 90 VDC On cycle, the system will recognize an SLT Off-hook event.
3. The SLT must incorporate a 90 VDC neon lamp that is connected directly across the tip and ring of the voice network

Code Restriction Override/Toll Restriction

Code Restriction Override/Toll Restriction
Description
Override
Code Restriction Override/Toll Restriction Override lets a user temporarily bypass the Code Restriction for an extension. This helps a user that must place an important call that Code Restriction normally prevents. For example, you could set up Code Restriction to block 100 calls and then provide a Code Restriction Override code to your attendant and executives. When the attendant or executive needs to place a 100 call, they just:
• Press Speaker key, dial a service code, and enter their override code.
• Press Speaker key and dial a trunk access code (e.g., 9 or # 9 002).
• Place the 100 call without restriction.
You can assign a different Code Restriction Override code to each extension. Or, extensions can share the same override code.
Code Restriction Override overrides all Code Restriction programming. Walking Code Restriction allows you to assign a Code Restriction level for each user. When a call is placed using Walking Code Restriction, the restriction for the call is based on the Code Restriction level defined in PRG 21-05-xx and PRG 21-06-xx.
Conditions
• Off-Premise notification and external extensions require access to outside lines.
• In the Class heading in the SMDR report, POTA indicates that the call was placed using Temporary Code Restriction Override.
• Code Restriction Override and Walking Code Restriction temporarily overrides an extension Code Restriction.
• Users will hear, “Your call cannot go through. Please call the operator” when they dial a number that Code Restriction prevents.

SLT BROKER CALL

SLT BROKER CALL
Description
Broker Call allows an SLT user to engage in two (2) calls, alternating between the two parties, so that the conversation with each party is private.
There are two types of Broker Call, Transfer and Camped On. Transfer Broker Call: 2nd Call is originated by SLT user. Camped On Broker Call: 2nd Call is delivered to the SLT through a Camp-On.
Operation SLT
To activate a Transfer Broker Call
1. Make or receive an intercom or external call.
2. Momentarily press the hook-switch, intercom dial tone received and active call is placed in Exclusive hold state.
3. Place second call.
4. To alternate between calls momentarily press the hook-switch.
To activate a Camp-On Broker Call
1. Make or receive an intercom or external call.
2. Receive a Call Waiting/Camp-On tone.
3. Momentarily press the hook-Switch, intercom dial tone received and the active call is placed on Exclusive Hold.
4. Dial the Camp-On Answer feature code ‘600’, camped on call is connected
To alternate between the calls
1. Momentarily press the hook-switch.
2. Dial the Camp-On Answer feature code ‘600’.
Conditions
1. After a hook-switch flash, if the call results in an error, busy, no answer or an abnormal state, the SLT user may momentarily press hook-switch to retrieve the held call.
2. During a Transfer Broker Call, if the SLT user goes on-hook, the Broker Call parties are connected completing a Call Transfer.
3. During a Transfer Broker Call, if the active caller disconnects from the SLT user, the held party, if another station, is connected to the SLT. If the held party is an CO/IP call, the SLT user receives error tone and may go on-hook to receive recall and retrieve the held call
4. During a Camp-On Broker Call, if the SLT user goes on-hook, the active call is disconnected and the held call recalls to the SLT.

IPECS PHONE ANSWERING MACHINE EMULATION

IPECS PHONE ANSWERING MACHINE EMULATION
Description
When a call is sent to a Voice mailbox, the associated station can be assigned to notify the user and allow the user to screen the call. Two methods of notification and call screening are provided, Ring or Speaker mode.
In the Ring mode, the user is notified by flashing of the AME (Answering Machine Emulation) Flex button. The user may press the Flex button to hear the caller as the voice message is stored. In the Speaker mode, when the call is sent to the Voice Mailbox, the caller’s voice is automatically broadcast over the speaker of the user’s iPECS Phone.
The user may terminate the screening leaving the caller in voice mail to record a message, talk with the caller and record the conversation in the mailbox, or answer the call and disconnect the Voice Mail.
The user’s iPECS Phone must be assigned with an AME Flex button for proper operation.
Operation iPECS Phone To assign an {AME} button: Ring Mode
[PGM] + {FLEX} + ‘564’ + ‘0’ + [SAVE] Speaker Mode
[PGM] + {FLEX} + ‘564’ + ‘1’ + [SAVE]

Flexible System Numbering

Flexible System Numbering
Flexible System Numbering lets you reassign the system port-to-extension assignments. This allows an employee to retain their extension number if they move to a different office. In addition, factory technicians can make comprehensive changes to your system number plan. You can have factory technicians:
• Set the number of digits in internal (Intercom) functions. For example, extension numbers can have a maximum of 8 digits.
• Change your system Service Code numbers.
• Assign single digit access to selected Service Codes.
Talk to your sales representative to find out if this program is available to you.
You can also use Flexible System Numbering to change the system Trunk Group Routing code. Although the default code of 9 is suitable for most applications, you can alter the code if needed.
The system provides a completely flexible system numbering plan. Refer to the chart below and the SL1100 Programming Manual for more details.
Flexible System

Programmable Function Keys

Programmable Function Keys
Description
Each Multiline Terminal has Programmable Function Keys. Programmable Function Keys simplify placing calls, answering calls and using certain features. You can customize the function of a Multiline Terminal programmable keys from each Multiline Terminal. Depending on your telephone style, you can have up to 24 Programmable Function keys.
Conditions
•When a key is programmed using service code 752, that key cannot be programmed with a function using the 751 code until the key is undefined (000). For example with a Park Key programmed by dialing 752 + *04 must be undefined by dialing 000 before it can be programmed as a Voice Over key by dialing 751 + 48.
•Using PRG 92-01 to copy a Multiline Terminal Programmable Function Keys, copies all the keys whether or not they exist on the telephone to which the programming is being copied. This may cause confusion when trying to define a key which is already defined but which does not exist on the telephone (displays as DUPLICATE DATA). It is recommended to either clear these non-existent keys or to copy only from an extension which has the same or fewer number of keys than the extension to which the programming is being copied.
•Speed Dialing and One-Touch Calling also offer quick access to calls and features.
•Programming a 60-button console requires separate programming.
•Below shows example of function key and LCD display indication (PRG 15-07-01 Function Key Assignment).
FunctionNu
mber FunctionDisplay
00None[All Blank] 01DSS/One-TouchDSS/ONE TOUCH 02Microphone Key (ON/OFF)MIC KEY 03DND KeyDND KEY 04BGM (ON/OFF)BGM 05HeadsetHEADSET 07Conference KeyCONFERENCE 10Call Forward - ImmediateCALL FORWARD 11Call Forward - BusyTRANSFER-BUSY 12Call Forward - No AnswerTRANSFER-NO ANS 13Call Forward - Busy/No AnswerTRANSFER-BUSY/NO ANS 14Call Forward - Both RingCALL FWD-DUAL RING 15Follow MeFOLLOW ME
If a key is programmed as a DSS/One-Touch key for a station that is set for Call Forward All Calls or Do Not Disturb, the DSS/One-Touch key flashes.
Refer to the SL1100 Programming Manual for a complete list of Function Numbers.
•One-Touch keys programmed for Park Hold Service Code cannot be used to park calls without using Hold or Transfer.
•Pauses can be entered in the dial string of a DSS/One Touch key. The pause is entered as P in the dial string and causes the system to wait three seconds before sending the rest of the digits that follow the P (pause). Multiple pauses can be entered.
•The @ can be entered in the dial string of a DSS/One Touch key. The @ only applies to ISDN and Intercom calls. When using the @, the system waits for the destination to answer (answer supervision), and then sends the rest