SIP Phone who has 3-way conference capability

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. System Conference Feature
1. Conference Room : refer to 2.23.1 Conference Room – SLT Operation
2. Conference Group : refer to 2.23.4 Conference Group – Operation
3. System need to be equipped MCIM module to mix voice

TWO-WAY RECORD

TWO-WAY RECORD
Description
An iPECS Phone user can record any active conversation to the station user’s mailbox or to hard disk drive of an iPECS Phontage or UCS Client. All calls including incoming, outgoing, internal, external, conference, conference rooms and conference group calls can be recorded. A {RECORD} button must be assigned to access this feature.
Operation iPECS Phone To assign a flexible button as a {RECORD} button [PGM] + {FLEX} + [PGM] + 80 + recording destination (optional) + [SAVE]
To activate Two-Way Record while on a CO/IP call
1. Press the {RECORD} button, record warning tone heard, and recording starts.
To stop Two-Way Record while on an CO/IP call
1. Press the {RECORD} again. Or,
1. Hang-up, return to idle.

TRANSFER TO SLT CLI

TRANSFER TO SLT CLI
Description
A SLT phone can be received CLI from internal caller instead of station number by programming when it’s transferring.
Operation
SLT
To transfer to SLT when it’s CO call
1. An User answers a call from CO with CLI.
2. The User is transferring to another SLT.
3. The SLT can see CLI of CO instead of station number.

OFF-HOOK SIGNALING

OFF-HOOK SIGNALING
Description
When a station, which is off-hook, receives a call or a CO/IP call rings into the system for the off-hook station, the station will receive the assigned Off-hook ring signal or, for ICM calls, a Camp-On, Voice-Over Announcement or Off-hook ring signal may be received.
The Off-hook ring Signal may be either a muted normal ring signal or a single tone burst based on the system assignment. This signal is delivered to the iPECS Phone speaker.
Operation
System
Operation of Off-hook ring signals is automatically controlled.
Conditions
1. While using the speakerphone, a Camp-On tone is provided over the speaker in place of the assigned Off-hook ring Signal.
2. Activating the DND or One-Time DND places the station in DND, terminating any Off-hook signaling.
3. Off-hook ring signals terminate when the call is answered, forwarded, or abandoned.
4. A station, which is receiving Off-hook ring signals, will receive normal ring upon return to idle status.

iPECS NMS

iPECS NMS is a Web-based application for monitoring and management of multiple iPECS systems using standard SNMP (Simple Network Management Protocol). iPECS NMS is an efficient convenient tool employing standards based protocols and a Web based architecture to permit administrators remote access to iPECS systems using any common Web browser. iPECS NMS monitors multiple systems displaying real-time detailed status information for the system devices and channels. iPECS NMS maintains a log of alarm and fault events defined by the administrator and can alert administrators of potential service affecting faults. In addition, call statistics are maintained and can be reported with various tables and graphs.

Distributed Control Network

Distributed Control Network
Description
In the Distributed Control Network, each iPECS system maintains control over the devices registered to it. The networked systems communicate allowing other networked systems access to resources over the network. In addition, other features and functions as detailed in the following sections of this manual are available to users in a distributed network environment. The iPECS permits remote access to various resources through registered gateway Modules and terminals.
In addition, iPECS will request access to resources of remote systems. The user-dialed number is analyzed and the call routed according to the Net numbering table. Should the main path fail to respond, the iPECS routes the call employing the alternative Speed Dial route assigned.
iPECS supports two (2) standard protocols, QSIG over ISDN and H.450 over IP, for the basic networking functions and the proprietary iPECS protocol for the advanced networking features. QSIG employs ISDN PRI channels only with support for ESTI standards ETS 300- 237/238/256/257/260/261/361/362/363/364.

What is a VPS

What is a VPS? Introduction to VPS Features
Description
The VPS handles incoming and outgoing calls. When a call comes in, it answers, forwards to appropriate extensions, takes and stores messages, and notifies subscribers when messages are left. Subscribers may send and transfer messages to other subscribers within the VPS. The VPS is easy to use, providing voice guidance (referred to as "system prompts") to callers, directing them to press certain buttons to access desired features.
Unlike handwritten messages or those left with answering services, VPS messages are confidential; they are stored in a mailbox and retrieved only with the subscriber's password. Other advantages of the VPS are clarity and accuracy, which are commonly lacking with written messages. The messages come directly from the caller, in the caller's own voice. To further ensure accuracy, the VPS allows the sender to correct or change messages before saving them. Messages can be erased or transferred by the recipient.
1.1.2 Basic Features
Description
Greeting Callers
The VPS greets callers with a prerecorded message that includes directions for leaving and editing messages. The VPS can list single-digit numbers for each available extension or mailbox. Callers who know the extension of the person they wish to reach may dial the extension number at any time. Callers with rotary phones are transferred to a preprogrammed destination, which is normally an operator or the General Delivery Mailbox.
Sending Messages
Callers can review and edit messages before leaving them in a mailbox. Subscribers can send messages to an individual or to several mailboxes at once. Extension users can also receive verification when messages they send to other subscribers have been received.
Receiving Messages
Subscribers can receive messages from outside callers and other subscribers. The total amount of recording time for all messages, as well as the maximum length of each message may be limited by the subscriber's Class of Service (COS). The VPS can notify subscribers when they have new messages by sending a beeper page, an e-mail, and even by calling subscribers' home or mobile phones. For subscribers who are on premises, the VPS can also light the message waiting lamp on their extension telephones. Subscribers can choose their preferred notification method. If the VPS is connected to the PBX using APT/DPT Integration, subscribers can press a preprogrammed button to record their telephone conversations into their own mailboxes or into other subscribers' mailboxes.

Rotation of jack number

Rotation of jack number
Each jack of the Digital Super Hybrid System supports the connection of a digital proprietary telephone and a single line device with different extension numbers (eXtra Device Port: XDP function). To program this function it is necessary to assign two parts for each jack. The first part of jack one is 01-1. The second part of jack one is 01-2. The first part of jack two is 02-1 and so on. The NEXT and PREV buttons can be used to move from jack to jack as require

ISDN Data Link Mod

ISDN Data Link Mode
Description
Assigns the data link mode on an ISDN port basis.

Calls-In-Queue routing

Calls-In-Queue routing
Description
When a caller is queued to an ACD Group, various announcements may be played and music on hold may be sent to the caller. The caller may dial a digit at any time while queued to exit the queue, except during a Guaranteed Announcement. The dialed digit is compared to digits defined in the ACD Group CIQ Route Table. If a match is found, the call is routed to the defined destination (Station, Station Group, etc.). If a match is not found, external callers receive an error message and are placed back in queue; internal callers are simply placed back in queue

System Components

System Components
System Components Table
Model Description Shelves KX-TDA100 Basic Shelf KX-TDA200 Basic Shelf Main Processing Card Main Processing Card (MPR) MPR Option Card KX-TDA0105 Memory Expansion Card (MEC) KX-TDA0196 Remote Card (RMT) CO Line Cards KX-TDA0180 8-Port Analog Trunk Card (LCOT8) KX-TDA0181 16-Port Analog Trunk Card (LCOT16) KX-TDA0187 T-1 Trunk Card (T1) KX-TDA0193 8-Port Caller ID Card (CID8) KX-TDA0290 PRI Card (PRI23) KX-TDA0480 4-Channel VoIP Gateway Card (IP-GW4) KX-TDA0484 4-Channel VoIP Gateway Card (IP-GW4E) KX-TDA0490 16-Channel VoIP Gateway Card (IP-GW16) Extension Cards KX-TDA0143 4 Cell Station Interface Card (CSIF4) KX-TDA0144 8 Cell Station Interface Card (CSIF8) KX-TDA0170 8-Port Digital Hybrid Extension Card (DHLC8) KX-TDA0171 8-Port Digital Extension Card (DLC8) KX-TDA0172 16-Port Digital Extension Card (DLC16) KX-TDA0173 8-Port Single Line Telephone Extension Card (SLC8) KX-TDA0174 16-Port Single Line Telephone Extension Card (SLC16) KX-TDA0175 16-Port Single Line Telephone Extension with Message Lamp Card (MSLC16) KX-TDA0470 16-Channel VoIP Extension Card (IP-EXT16) Other Cards KX-TDA0161 4-Port Doorphone Card (DPH4) KX-TDA0164 4-Port External Input/Output Card (EIO4) KX-TDA0166 16-Channel Echo Canceller Card (ECHO16) KX-TDA0168 Extension Caller ID Card (EXT-CID) KX-TDA0190 Optional 3-Slot Base Card (OPB3) KX-TDA0191 4-Channel Message Card (MSG4) KX-TDA0410 CTI Link Card (CTI-LINK)

TERMINAL

TERMINAL CONNECTI0N Requirements for Connecting Programming Terminal
The programming terminal must be connected with a serial cable with an RS-232C connector at the RS-232C port. This must be a null modem cable. This enables system administration (system setup, mailbox setup, and system diagnosis) to be performed.
Communication parameters of the VPS have been set to the following values at the factory:

Station Programming

Station Programming (Personal Programming)

Caller ID Indication Button (Assignment)
Allows you to assign a Flexible CO button as the Caller ID Indication button.
— Be sure that you are in the Station Programming mode. Set the MEMORY switch to “PROGRAM”.

Station Speed Dialing

Station Speed Dialing
Description
Allows an extension user to store frequently dialed numbers in order to place a call with abbreviated dialing. It is performed by dialing the feature number and a speed dial number from 0 through 9. Up to 10 numbers can be stored in each telephone.
Conditions
• Station Speed Dialing can be followed by manual dialing to supplement the dialed digits.
• You may make a call with One-Touch Dialing button, instead of Station Speed Dialing.
• The single line telephone may be replaced with a proprietary telephone (PT) temporarily to store one-touch dialing into memory. The Function Buttons F1 through F10 correspond to speed dial numbers as follows:

DIGITAL INTEGRATION

DIGITAL INTEGRATION
General
There are 2 types of Digital Integration: APT Integration and DPT Integration.
APT Integration is available when the VPS is connected to a KX-TA624. DPT Integration is available when the VPS is connected to a KX-TD or KX-TA1232 digital PBX.
APT Integration
To the Panasonic KX-TA624, the VPS ports look like proprietary telephones. The PBX thinks that the VPS is a proprietary telephone, and the VPS mimics all actions of a proprietary telephone. Communication between the VPS and the PBX through digital integration requires the proper software level in the PBX and 4-wire connections for each port. To communicate between the VPS and the PBX through APT Integration, the PBX and VPS must be programmed to work together.

Line , Direct Access

Line , Direct Access
Description
Allows the proprietary telephone user to select an outside line by pressing an idle CO button, which automatically establishes the handsfree operation mode and allows the user to perform On-Hook Dialing. The user need not press the SP-PHONE button, MONITOR button nor lift the handset.
Conditions
• There are three types of CO buttons which can be programmed on an extension: Single-CO button, Group-CO button, and Loop-CO button.
• Each extension is subject to System Programming items for outside lines available to access.

Automatic Configuration

Automatic Configuration*1
Description
The system sends the Voice Processing System (VPS) data which contains the extension number configuration information. The VPS automatically creates mailboxes with this data (Quick Setup).
Conditions
• The data is transmitted to the VPS via the lowest jack port.
• Please see the Note in the Programming Guide, [003] Extension Number Set. [003] should be done before Automatic Configuration is performed.

Logging in to Your Mailbox

Logging in to Your Mailbox
In order to access Subscriber Services (to play messages, change mailbox settings, etc.), you must first log in to your mailbox. There are 2 ways to log in to your mailbox:
a) Automatic Log-in:
When logging in to your mailbox from your own extension, you do not need to enter any special commands or your mailbox number. This feature is not available with certain PBXs and may be disabled by the System Administrator or System Manager for your mailbox.
b) Manual Log-in:
You need to enter the Voice Mail Service Command [#6] followed by [ ] and your mailbox number. Manual Log-in is necessary when logging in to your mailbox using someone else's extension, when Automatic Log-in is not available with your PBX, or when Automatic Log-in is not enabled for your mailbox.
You will know that you have logged in successfully because the VPS will announce either the Main Menu of Subscriber Services or the number of new messages you have, or the VM Menu will be displayed.

Station programming

Station Programming (Personal Programming)

Log-In/Log-Out Button (Assignment)
Allows you to assign a Flexible CO button as the Log-In/Log-Out button.
— Be sure that you are in the Station Programming mode. Set the MEMORY switch to “PROGRAM”.
PT
CO
Press the desired Flexible CO button you wish to assign as the Log-In/Log-Out button.
8
Dial 8.
<PT Display Example> Group Log In/Out
AUTO DIAL STORE Press STORE.
• The STORE indicator lights.
• The display shows the initial programming mode.

Automatic Station Release

Automatic Station Release
Description
After going off-hook, if an extension user fails to dial any digits within a specified time period, the user will be disconnected from the line after reorder tone is sent. To get a line again, the user must go back on-hook and then off-hook.
Conditions
This function works in the following cases: When making a call
a) The first digit has not been dialed within 10 seconds.
b) After a digit is dialed, the next one is not dialed within five seconds (Intercom call only).

Admin & Maintenance

Admin & Maintenance

To change the Country Code:
set the MFIM switch 3 and 4 to the On position,
follow the procedure below to modify the Country code initialize the MFIM as outlined in the Initialization section.
After initialization, reset switches as needed, switch 4 initializes database on reset and switch 3 enables automatic registrations. Generally, switch 4 is set to Off and switch 3 is left On until after initial installation of all Modules and terminals.
A twenty-three (23) character SITE NAME and the local Area Code are also defined in this program. The SITE NAME is primarily useful for the installer/programmer as a reference to the customer.
In addition, under this program the system can be programmed to select one of eight (8) Flexible Number Plans, refer to Appendix B. Individual items from the selected Numbering Plan can be changed under Flexible Numbering Plan part A to D –

Planning your ISDN network

Planning your ISDN network
Consult ISDN hardware on page 64 and ISDN programming on page 76 to determine a configuration of ISDN trunks and terminal equipment (TE) for the Modular ICS, then order the appropriate ISDN capability package from your ISDN service provider.
For ISDN BRI service, your service provider supplies service profile identifiers (SPIDs), network directory numbers (Network DNs), terminal endpoint identifiers (TEIs), and other information, as required, to program your Modular ICS, TE, and other ISDN equipment.
Modular ICS does not support any package with EKTS (Electronic Key Telephone System) or CACH (Call Appearance Call Handling). EKTS is a package of features provided by the service provider and may include features such as Call Forwarding, Link, Three-Way Calling, and Calling Party Identification.
Ordering ISDN PRI
When you order ISDN PRI, order two-way DID because it simplifies provisioning and provides efficient use of the PRI bandwidth.
Ordering ISDN PRI service in Canada
In Canada, order Megalink™ service, the trade name for standard PRI service and set the Norstar equipment to the supported protocol that is identified by your service provider, either DMS-100 or NI-2.

Call Duration Timer

Call Duration Timer
Call Duration Timer lets a Multiline Terminal with an LCD time their trunk calls on the telephone display. This helps users that must keep track of their time on the telephone. For incoming trunk calls, the Call Time begins as soon as the user answers the call.
Conditions
• The Call Timer starts over each time the call is retrieved from Hold or Park.
• The Call Duration Timer (Program 20-13-36) is not displayed for inbound ACD calls. (V1.5 or higher)

H.323 v4 Service

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

Entry Points with Keyless Entry

Provide 4 Door Entry Points with Keyless Entry (or Optional Card Readers) for up to 250 Apartments or Offices

The C-4000 converts any four touch tone phones into multi-number auto dialers that will store up to 250 tele- phone numbers in non-volatile memory. Use with Viking’s K-1700-3 or K-1900-8 phones to provide vandal resistant handsfree or handset communication from entry points to apartments or offices.
When a call initiated by the C-4000 is answered by an apartment or business tenant, a built-in contact closure may be activated to control an electric gate or door strike.
Up to 250 entry codes may also be programmed provid- ing tenants with keyless entry or optional PROX-1 Card Readers may be added for Proxy card entry.
The C-4000 can be programmed locally or remotely using a standard Touch Tone phone. The C-4000 has built-in user dialing restriction to help prevent unauthorized calls and toll fraud.

SA-25 paging control

The most versatile paging system ever!
Manufacturers usually refer to their product as a „system‰ just so they can lock you into buying all of their accessory components. Not VIKING! We designed the SA-25 paging control unit to install as a new unit, a direct retrofit replacement for your existing 4-wire legacy system, or to expand your present system. To replace your existing amp, just connect all the 4-wire speakers directly to the terminals on the SA-25. To expand, just add the SA-X12 and VIKING’S 2-wire SA- 1S self-powered speakers (or SA-1H self powered speakers horns).
The SA-25 provides paging, loud ringing, and background music when connected to electronic 1A2 key systems, PBXÊs, No-KSU phones, multi-line phones, as well as IP phone systems with an analog FXS of FXO port.

Connect the SA-25 to a paging port, an unused telephone line input (trunk port), a ringing C.O. line, Centrex line, analog PABX/KSU station port, or the IP FXS or FXO port.
For loud ringing, connect the SA-25 to a ringing analog line or a dry contact closure. Select an electronic warble or one of the three soft chime tones built into the control unit. Then connect an external „night transfer‰ switch to control loud ringing in night bell applications.
The SA-25 powers up to 25 self-amplified SA-1S ceiling speakers or SA-1H self-amplified horns. The SA-IR remote control allows users to set the volume on a speaker-by-speaker basis. If you need more speakers, simply use the SA- X12 to add up to 12 VIKING self amplified speakers at a time.

Executive/Secretary

Executive/Secretary
Description
9 STATION PAIRS
Stations in the system can be assigned as an Executive/Secretary pair. By activating DND, the Executive also activates Unconditional Call Forward to the Secretary, which will forward Executive calls to the Secretary. In addition, Intercom and/or CO calls to the Executive can be assigned to forward to the Secretary regardless of the Executive’s station status by assigning ICM Call to Secretary and CO Call to Secretary options. Also in the above case,, if the Secretary is in DND, Executive calls sent to the Secretary route back to the Executive if the “Call Exec If Sec in DND” option is enabled.
Each Executive can be assigned a “Grade” (01, highest ~12, lowest). Executives with the same or higher grade can call lower grade Executives overriding the ICM Call to Secretary assignment.
Operation
To activate Executive/Secretary Forward from the Executive's DKT,
1. Press the [DND/FOR] button.
Consideration
 An Executive may have multiple Secretaries and a Secretary may have multiple
Executives. Each forms a separate Executive/Secretary pair.
 If the Secretary is busy when a call is received for the Executive, the caller will receive
busy tone.
 The Secretary may override the DND (refer to Ref. A) status of the Executive to Camp-On
(refer to Ref. B) and transfer calls to the Executive.
 A chain can be constructed by assigning the Secretary of one pair as an Executive of
another. Although a chain may be constructed, a loop back is not allowed.
 If an Executive has multiple Secretaries, calls will automatically route to the Executive’s
first idle Secretary.
 The Executive may use Call Forward (refer to Ref. D) to send calls to stations other than
the Secretary.
 When both Executive and Secretary are busy, camp-on, transfers and messages are sent
to the last Secretary Station in the chain.

Call Forwarding, Off-Premise

Call Forwarding, Off-Premise
Off-Premise Call Forwarding allows an extension user to forward their calls to an off-site location. By enabling Call Forward, Off-Premise, the user can stay in touch by having the system forward their calls while they are away from the office. The forwarding destination can be any telephone number the user enters, such as a mobile phone, home office, and hotel or meeting room. Off-Premise Call Forwarding can route the off-site telephone number over a specific trunk or through a trunk group, Automatic Route Selection or Trunk Group Routing.
Off-Premise Call Forwarding reroutes the following types of incoming calls:
• Ringing intercom calls from co-worker’s extensions
• Calls routed from the VRS or Voice Mail 1
• Direct Inward Lines 1
• DISA and DID calls to the forwarded extension 1
• Transferred calls 1
1 Off-Premise Call Forwarding can reroute an incoming trunk call only if the outgoing trunk has disconnect supervision enabled (refer to the Programming section).
Off-Premise Call Forwarding does not reroute Call Arrival (CAR) Keys, Virtual Extension keys or Ring Group calls (i.e., trunk ringing according to Ring Group assignments made in PRG 22-04 and PRG 22-05).
Conditions
• If a call that forwards Off-Premise goes out on a trunk assigned as TIE or DID, and the called party does not answer before the timer in PRG 34-07-05, the call recalls to the station that performed the transfer.
• Call Forwarding Off-Premise requires loop start trunks with disconnect supervision.
• The trunk access code and the outside telephone number combined cannot exceed 24 digits.
• Call Forwarding an extension in a Department Group prevents that extension from receiving Department Pilot Calls.
• If a Programmable Function key is not defined for Call Forwarding (10 ~ 17), the DND key flashes to indicate that the extension is call forwarded.
• DID calls to an extension with Off-Premise Call Forwarding set do not recall if there is no answer.
• Door Boxes must be programmed for the calls to be transferred Off-Premise.
• The outside number Call Forwarding dials can only be a number normally allowed by the forwarded extension Toll Restriction.
• In systems with a PZ-VM21, callers to an extension forwarded off-premise hear, “Please hold on, your call is being rerouted.” This option can be disabled in PRG 40-10-01 by setting it to disable.
• When a station is in DND and any Call Forwarding Off Premise is set, the call forwards immediately.
• Call Forwarding, Off-Premise is not supported when using Alternate Trunk Group Routing.