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.
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.


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.
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.


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.


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}.
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


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.
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


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.
1. Ring On and Ring Off are assigned on a system basis.
2. The CO Ring Detect is applied to analog CO Lines only.


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.
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

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


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.
Operation System Operation of Linked pairs is automatic when defined
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).


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 ‘#’.
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.
• 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
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.