Automatic Route Selection

Automatic Route Selection (ARS/F-Route) provides call routing and call restriction based on the digits a user dials. ARS gives the system the most cost-effective use of the connected long distance carriers.
ARS is an on-line call routing program that you can customize (like other system options) from a display telephone. ARS accommodates 400 call routing choices - without a custom-ordered rate structure database. With ARS, you can modify the system routing choices quickly and easily. This is often necessary in the telecommunications world of today where the cost structure and service choices frequently change.
The ARS feature can add or delete digits and route calls according to pre-determined levels. ARS Feature Summary ARS provides: Call Routing
ARS can apply up to 36-digit analysis to every number dialed. For programming, ARS provides separate 8-digit and 36-digit tables. Each table can have up to 250 numbers.
• Dialing Translation (Special Dialing Instructions)
ARS can automatically execute stored dialing instructions (called Dial Treatments) when it chooses a route for a call. The system allows up to 15 Dial Treatments. The Dial Treatments can:
- Insert or delete an area code (NPA)
- Add digits (such as a dial-up OCC number), pauses and waits to the dialing sequence
- Require the user to enter an authorization code when placing a call (refer to PRG 44-03)
• Time of Day Selection
For routing purposes, ARS provides 10 different day selections (called Time Schedule Patterns). Each Time Schedule Pattern can provide up to 20 time intervals which are assigned to one of the eight day/night modes. The Time Schedule Patterns are then assigned to a day of the week (Monday~Friday, Saturday, Sunday or Holiday).
• Hierarchical Class of Service Control
ARS allows or denies call route choices based on an extension ARS Class of Service. This allows lower Classes of Service (e.g., 1) to access routes unavailable to higher Classes of Service (e.g.,
16). The system provides up to 16 (0=unrestricted, 1~16) ARS Classes of Service.
• Separate Routing for Selected Call Types
To provide unique control, you can program separate routing instructions for:
- Directory assistance calls
- Emergency calls
Basic ARS Operation
When a user places an outside call, ARS analyzes the digits dialed and assigns one of 400 Selection Numbers to the call. The Selection Number chosen depends on which digits the user dialed. ARS then checks the time of day, the day of week and the extension ARS Class of Service. Based on these call routing options, ARS selects a trunk group for the call and imposes the Dial Treatment instructions (if any).

Trunk or Station Module down

Trunk or Station Module down

1. Run a Maintenance session to ensure that the Trunk Module is not disabled. See Module status on page 537.
2. Disable the module using the Maintenance heading Module status.
3. Enable the module using the Maintenance heading Module status.
4. For Trunk Module
Check the external line by terminating a single-line telephone directly on the distribution block, or equivalent, which connects to the Trunk Module.
5. For Station Module
If the Station Module is still down, power down, then power up the ICS.
If the problem persists
1. If AC power is present and the LED indicator on the Trunk Module is off, replace the Trunk Module.
2. Replace the fiber cable.
3. Replace the Trunk Cartridge.
4. Replace the Expansion Cartridge.
5. Replace the ICS.

Clock/Calendar

The battery on the CPU retains the Clock/Calendar when the CPU encounters a power loss. With a fully charged battery, the settings are retained for approximately three years.
The system programmed memory (Customer Database) is stored in Nonvolatile Memory and can be erased only by performing a First Initialization.
For additional storage time and the database can be copied to the Compact Flash card on the CPU.
Conditions
• The battery on the CPU should be removed during long term storage but must be installed (protection against loss of power) just before ETU installation to provide battery backup for System Memory.
• When fully charged, the battery retains System Memory for approximately three years.
• You should replace the CPU battery every three years.
• During normal operation, the battery is continually recharged using a built-in charging circuit from the CPU.
• Battery backup on the CPU does not protect the following:
- Callback
- Off-line Status (for programming system or station assignments)
- Repeat Redial
- Trunk Queuing/Camp-On

CSU stats

CSU stats

Each DTI is equipped with an internal channel service unit (CSU). When enabled, the internal CSU monitors the quality of the received T1 signal and provides performance statistics and diagnostic information.
DTIs must be individually programmed to establish parameters for collecting and measuring transmission performance statistics by the CSU.
Statistics Installer password required
The CSU provides both performance and alarm statistics. Three performance parameters are accumulated:
• errored seconds (ES-P)
• severely errored seconds (SES-P)
• unavailable seconds (UAS-P)
These parameters are defined as per TIA-547A. Errored seconds are enhanced to include control slip (CS) events.
The parameters are stored for the previous 15-minute interval, the 15-minute intervals in the last 24 hours, and the previous 24-hour interval. Only near-end performance data is recorded.
The internal CSU continuously monitors the received signal and detects four types of transmission defects:
• any active carrier failure alarms (CFA) — loss of signal LOS, out of frame OOF, alarm indication signal AIS, remote alarm indication RAI
• the number of bipolar violations that occurred in the last minute

digital phone

7000 digital phone This telephone has a no display. It does have four memory buttons. (Note this telephone is only supported on systems running Profile 2, 3, or 4).
7100 digital phone: A telephone with a single line display and one programmable memory button without an indicator.
7208 digital phone: A telephone with a single-line display and eight programmable memory buttons with indicators. This telephone has a separate mute key and supports a headset
7316 digital phone: A telephone that has a two-line display, three display buttons, 16 programmable memory buttons with indicators, and 12 memory programmable buttons without indicators. This telephone has a separate mute key and supports a headset.
7316E digital phone: This telephone has the same functionality as the 7316 digital phone, with some additional features, such as a separate handsfree key, special display icons (when running on a MICS 6.1 or newer system), and CAP capability by adding KIMs.
7406 digital phone: The 7406 base station desk set can support three handsets, which function like the other digital phones on the system. It has six programmable memory keys.
7420/7430: These Nortel Networks Digital Mobility phones allow you to set up an extended cell of base stations that allow users to contact the system from a variety of locations within a site.
A
AbsorbLength: A setting that determines how many of the digits in a destination code will not be dialed by the system. AbsorbLength is assigned under Destination codes in Services.
access code: Different sequences of characters used to gain access to these Norstar features: Line pools, Call park, external lines, Direct-Dial telephone, Auto DN, and DISA DN.
alarm code: A number that appears on the alarm telephone display, informing you that the ICS has detected a fault in the system.
alarm telephone: A telephone that is designated to receive reports of Norstar system problems. This function is usually assigned to a prime telephone, but this can be changed under Feature settings in Sys prgrammng.
Analog Terminal Adapter (ATA): A device that permits analog telecommunication devices such as fax machines, answering machines, and single line telephones to be connected to the Norstar system. Programmed defaults for the ATA are automatically assigned by the Norstar system.
ANSI: American National Standards Institute.
Answer button: A telephone button with an indicator that is used to monitor another telephone. The answer button indicates incoming calls destined for the other telephone. Someone working at a telephone with answer buttons (an attendant, for example) can receive all

Programming an Operator’s Extension

Programming an Operator’s Extension 7
One of the initial programming duties in programming extensions is to set up operator extensions. To set up an operator’s extension, you must program call handling options, backup answering options, and buttons used for specific operator features.
Call Handling Options 7
If you set up a centralized telephone answering position at extension 10, use the following settings to customize it:
■ Call Answering. If the operator should answer all calls, use Line Assignment (#301) to assign all lines to extension 10. Set Line Ringing for all lines at extension 10 to the desired number of Rings; set the lines assigned at each user’s extension to Delayed Ring or No Ring.
In Hybrid mode, Immediate Call Answering is the factory setting. (Lines are assigned as individual line buttons on the telephone at extension 10 and all pool buttons assigned to users’ extensions are set to No Ring.)
■ Backup Call Answering. If the operator should answer some lines only when a user does not pick up, set Line Ringing for those lines at extension 10 to Delayed Ring; set the lines or pools assigned at each user’s extension to Immediate Ring.
■ No Answering. If some lines should not be picked up by the operator at all, either set Line Ringing for those lines at extension 10 to No Ring, or simply use Line Assignment (#301) to remove those lines from extension 10. In either case, set Line Access Restriction (#302) to No Access for those lines at extension 10 to prevent the operator from using Direct Line Pickup to access those lines.

Features from 6.1MR addendum

Features from 6.1MR addendum
The following features were introduced with the MICS 6.1MR (Maintenance Release) software in the Modular and Compact ICS 6.1 Maintenance Release (MR) Documentation Update addendum (P0609198 02):
• The enhanced Call log feature allows you to log all calls to a telephone, or to gather logs for specific lines assigned to a telephone. The Call log set feature allows you to determine which assigned lines will collect logs. Refer to Call log set on page 358 and Call log on page 436. (all profiles)
• The second dial tone table allows the user to enter up to 10 one to four-digit numbers that, when dialed, will cause the system to produce a second dial tone, at which time the user can enter the remaining call digits. Refer to Configuring the second dial tone table on page 403. (all profiles)
• Profiles 1 and 4, PRI: The Send Name Display feature allows you to specify if you want the business name and OLI to be transmitted over specific PRI lines.

Programming ISDN BRI resources

Programming ISDN BRI resources
Some steps will not be necessary depending on the service you are providing.
More detailed information is included under the individual headings and settings in the Programming and Maintenance sections.
For complete card and cartridge installation instructions and safety precautions, see Installation on page 227.
1. Collect the information supplied by your service provider to support your ISDN package. This includes network service profile identifiers (SPIDs) and Network DNs. If you are supporting a point-of-sale terminal adapter, you also need one or more terminal endpoint identifiers (TEIs).
2. Make sure a Combination Fiber 6-port Services Cartridge, or a Services Cartridge has been installed in the ICS.
3. Install the BRI card in the ICS, Trunk Module. Refer to Installing the cartridges on page 231 for information about BRI card placement. If you are not using a BRI card, determine which type of card you will preprogram the ICS to use in each slot.

Data Solutions

Data Solutions
Examples of ISDN Scenarios
For information about various ISDN scenarios that may help you to decide on the data solution that is best for you, refer to the following web site: http://www.nortelnetworks.com/ support and perform a Search for TIPS. You will need your user name and access code.
If you do not have a user name and access code, the site provides information about how to get access to this site.
ISDN applications
ISDN terminal equipment delivers a wide range of powerful business applications:
• Video conferencing and video telephony: Video conferencing offers instant visual and audio contact between distant parties using either studio-based or desktop ISDN terminals.
• Desktop conferencing: ISDN allows computer users in distant locations to share and edit any image, data or text file on their own computer screens while they discuss the information.
• File transfer: The ISDN network allows you to transfer files containing data, text, images, data, or audio clips, faster and cheaper than with a conventional modem.
• Telecommuting: Convenient retrieval, processing and storage of files is possible for the employee working at home by using ISDN lines to give high-speed access to information resources at the office.

Working with ISDN

Working with ISDN
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.

ATTENDANT STATION PROGRAM CODES

ATTENDANT STATION PROGRAM CODES
Description
Using the Attendant Station Program Codes, the Attendant can print SMDR and Traffic reports on-demand, assign Authorization Codes, control certain user features, record VMIM/VSF announcements, enable/disable Auto Service Mode Control, etc. Items are available using the Program Code directly or scrolling the multi-level display menu. The following indicates the menu displays, including the digit for selecting the item, the item description and further required entries. The various levels of the display menu are indicated by indentation. For additional information, reference Appendix C of the iPECS Admin & Programming Manual.
Note also, some Program Codes are only available to the System Attendant or stations allowed access to the Attendant Program code “0”.

IP Trunk - (SIP) Session

IP Trunk - (SIP) Session Initiation Protocol
Version 2.0 or higher software provides;
• When + is added to the country code of an incoming SIP trunk call, it is recognized as an
international call, simplifying outgoing calls from the incoming call list.
Description
The SL1100 IP Trunk SIP package sends the real time voice over the corporate LAN or WAN. The voice from the telephone is digitized and then put into frames to be sent over a network using Internet protocol.
Using VoIPDB equipment at a gateway (a network point that acts as an entrance to another network), the packetized voice transmissions from users in the company are received and routed to other parts of the company Intranet (local area or wide area network) or they can be sent over the Internet using CO lines to another gateway.
Depending on the requirements and resource allocation in the LAN/WAN/Internet, the VoIPDB - SIP can be configured to use any of the following voice compressions:
• G.711 μ-law - Highest Bandwidth
• G.729 (a) - Most often used
• The LAN/WAN or Internet connection is provided by a 10 Base-T/100 Base-TX Ethernet.
For a list of vendors that have successfully completed interoperability certification go to http:// www.necntac.com and refer to Technical Documentation.
SIP Trunk E.164 Support (V2.0 or higher)
With the SIP Trunk E.164 Support enabled the PBX is able to support SIP configurations where the number presentation within the SIP messages is formatted using the E.164 international numbering scheme. Specifically the system is able to handle the ‘+’ digit when required as the International Access Code.
For example a normal international SIP call could be dialed and presented from the system as follows; Number dialed = 00441202223344
Request-URI: Invite sip: 00441202223344@172.16.18.100 SIP/2.0
However with SIP Trunk E.164 Support enabled the SIP call could be presented once dialed as below; Request-URI: Invite sip:+441202223344@172.16.18.100 SIP/2.0
This presentation can be a requirement of certain SIP ITSPs (Internet Telephony Service Providers) so it is necessary the PBX can handle these calls and modify any SIP messages to the correct format accordingly.
Below is the full list of SIP header fields used by this feature: Request-URIToFromP-Asserted IdentityP-Preferred Identity
SIP Trunk E.164 CLIP Enhancement (V2.0 or higher)
With the SIP Trunk E.164 CLIP Enhancement enabled, when an incoming SIP call from an external ITSP is presented at the system with a “+” in the From header field as the international access code, it is recognized and displayed as an international call at the terminal display and also logged in the terminals incoming caller history, allowing any outbound calls made from a multiline terminals caller history possible using this numbering scheme.
This presentation can be a requirement of certain SIP ITSPs (Internet Telephony Service Providers) so it is necessary the PBX can handle these calls and modify any SIP messages to the correct format accordingly.

INTERCOM TRANSFER

INTERCOM TRANSFER
Description
Users can transfer an active Intercom call to other stations in the iPECS system. Intercom calls can be transferred after announcing the call (screened) or without announcing the call (unscreened).
The Intercom station is placed on Exclusive Hold. The Transfer Recall Timer is initiated and, if this timer expires before the Intercom call is answered, the call will recall the transferring station until answered or abandoned.
Operation
iPECS Phone
To perform an Screened ICM transfer, while on an ICM call
1. Press [TRANS] button.
2. Dial the station to receive call.
3. At answer or Splash tone, announce call.
4. Hang-up, return to idle. Or,
1. Press {DSS/BLF} button for the desired station.
2. At answer or Splash tone, announce call.
3. Hang-up, return to idle.
While on a Intercom call, Unscreened call transfer
SLT
1. Press [TRANS] button.
2. Dial Station to receive call.
3. Hang-up, return to idle. Or,
1. Press {DSS/BLF} button for the desired station.
2. Hang-up, return to idle.

Confirm that the system is correctly set, perform the following tests

Confirm that the system is correctly set, perform the following tests:
 If you run into an issue with any of these tests, refer to Table 3 Troubleshooting Guide. Test an outgoing call to a local number. Check for ringback, 2-way audio and quality.
1. Test an outgoing call to a long distance number. Check for ringback, 2-way audio and quality.
2. Test an outgoing call to an international number. Check for ringback, 2-way audio and quality.
3. Test a outgoing call lasting more than 15 minutes.
4. Test multiple call concurrences on outgoing calls. Setup multiple calls to PSTN.
5. Test an outgoing call to an Operator ‘0’.
6. Test an outgoing call to directory assistance ‘411’.
7. Test a 911 call.

Identify to the operator that this is a TEST!
8. Test an incoming call to an internal DID. Check for ringback, 2-way audio and quality.
9. Test an incoming call to an auto-attendant. Check DTMF and audio quality.
10. Test transferring calls off-site.
11. Test an outgoing call to an auto-attendant and verify DTMF.

Setting the Time

Setting the Time
Users can set the time on the server using NTP to set the time automatically or users can adjust the time and date manually. NTP is the preferred setting, if the server has internet access.
To set the time:
1. Navigate to Maintenance > Time.
2. Click the modify link in the Action column. The Time page displays.
3. Click the radio button to:
a. Use NTP to set time automatically. In the NTP server field, specify an SNTP server IP address or a domain name. In the Poll Period, specify the number of minutes between polls. OR
b. Set time manually - specify the time in hours, minutes, seconds, and date in the respective fields. .
4. Go to the Time Zone section and select the appropriate time zone from the drop-down list.
0.
a. Check the Automatically adjust clock for Daylight Saving Time box underneath the selected time zone to enable the system to update the time automatically.
b. Click the Get Time button to update the clock to the current time when using an SNTP server to specify the time. Click the Set Time button to save changes. . .

Remote system access

Remote system access
The remote access feature allows callers elsewhere on the private or the public network to access a Norstar system by dialing directly into the system without going through an attendant. Once in the system, the remote user can use some of the system resources. The remote access must be enabled in programming before callers can use it.
Norstar systems support remote system access on the following trunk types, which may require the remote caller to enter a COS password for direct inward system access (DISA):
• auto-answer loop start trunks
• auto-answer E&M trunks
• DID trunks, by means of the DISA DN
• PRI trunks, by means of the DISA DN
The system resources, such as dialing capabilities, line pool access and feature access, that a remote user may access depends on the Class of Service (COS) assigned to the user. See Class of Service on page 103, COS pswds on page 412 and the Modular ICS 7.0 System Coordinator Guide for more details.

Calls cannot be made (but can be received)

Calls cannot be made (but can be received)
1. Press ≤•‚.
2. Press a line button.
3. If an incorrect line number or name appears, or if neither appears, on the display, check the programming settings.
OR
If the correct line number or name appears on the display, make sure the external lines are properly cross-connected.
4. Check external lines by attaching a test telephone directly on the distribution block.
5. Ensure that the 25-pair cable is properly connected to the modules or the ICS.
6. Run a Maintenance session.
7. Use Module status to verify that the module on which the Trunk Cartridge is installed is not disabled or unequipped.
8. Run a Maintenance session and disable the appropriate ports using Port/DN status.
Charts showing external line port number defaults appear under Wiring charts on page 260.
9. Enable the appropriate ports using Port/DN status.

Service Code Setup (for ACD)

Service Code Setup (for ACD)
Use Program 11-13 : Service Code Setup (for ACD) to customize the Service Codes which are used with the Automatic Call Distribution (ACD) feature. You can customize additional Service Codes in Programs 11-10 ~ 11-12 and Programs 11-14 ~ 11-16. The following chart shows:
 The number of each code (01~13).  The function of the Service Code.  The type of telephones that can use the Service Code.  The default entry.
 If you change a Service Code, be sure to record your entry in the New column. Input Data
Item No. Item Terminals Default New 01 ACD Log In / Log Out (for KTS) MLT, SLT  5 02 ACD Log Out (for SLT) SLT 655 03 Set ACD Wrap-Up Time (for SLT) SLT 656 04 Cancel ACD Wrap-Up Time (for SLT) SLT 657 05 Set ACD Off Duty (for SLT) SLT 658 06 Cancel ACD Off Duty (for SLT) SLT 659

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.
SRTP
1. For Voice and Video RTP Data Encryption
2. To implement SRTP, VOIM channel is required for SRTP packet relay. You should equipped LIK VOIP G/W.
3. Usage (for example, LG-Ericsson SIP Phone)
- SIP Data / SIP Phone Attributes(211) – SRTP Usage : ON
- SIP Phone Self Programming(Phone’s Web Programming) – SRTP Usage : ON
4. Encryption Key Method (for example, LG-Ericsson SIP Phone)
- SIP Data / SIP Phone Attributes(211) – 1st CRYPTO : select one of listed

MUSIC SOURCES

MUSIC SOURCES
4.13.1 Music on Hold
The system can provide Music on Hold from either an internally synthesized source on the CCPU or from an external source. The external MOH can be a tuner, tape deck, CD player, etc.
In addition to a connector on the CCPU, the 2PGDAD modules also provide connections for external MOH sources. When using external music sources for external MOH, programming determines the MOH source for each trunk. Refer to Music on Hold in the Software Manual for more details.
4.13.2 Installing External Music on Hold
1. Connecting to the 2PGDAD Module:
Make sure the jumper in the 2PGDAD module for the channel is set correctly. (Refer to Figure 4-4 JUMPER SETTINGS on page 4-5).
2. If a line cord was not previously connected to the 2PGDAD, complete Steps 3-6. Otherwise, skip to Step
7.
3. Install a modular jack for each 2PGDAD module. For each module, run one-pair 24 AWG station cable from the cross-connect block to a modular jack. Ground the unused pair.
4. Terminate the extension leads to GRN/RED of the modular jack. Terminate the unused leads to the jack.
5. Install bridging clips as required.
6. Plug a modular line cord from the mod jack to the CN1 connector on the 2PGDAD module.
7. Connect an RCA jack to the audio output(s) on the back of the 2PGDAD module.
8. The opposite end of this cable is connected to the external music source either directly or by connecting to the cross-connect block where the music source is connected.

2PGDAD

2PGDAD

2PGDAD MODULE
4.2.1 Using a 2PGDAD Module (Figure 4-1 - Figure 4-8)
The 2PGDAD module provides two circuits which allow connection to external terminals such as: Door Box External Speaker External Music Source (external MOH) External Recording System External Ringing
The system allows for 2PGDAD modules to be installed for Door Boxes and Paging. The 2PGDAD module also provides multi-purpose controls. These control relays can be used for controlling the external amplifier, external music source and door lock control with the use of a Door Box. The 2PGDAD module connects to any available digital extension port. The terminal connections made within the PGDAD module and the jumper settings determine what features are used for each circuit.

I DATA LINE SECURITY

DATA LINE SECURITY
Description
Data transmitted over analog CO lines is subject to distortion and errors if system tones such as Camp-On and Override are applied during transmission. To eliminate such errors, stations that use analog data (modems or Fax) can be assigned to block incoming system tones.
Operation System System tones are automatically blocked when Data Line Security is assigned

Set has faulty buttons

Problems with telephones

Set has faulty buttons, display, handset or other hardware problems
Run a Station Set Test (≤°‚fi).
Detailed instructions are contained in the Modular ICS 7.0 System Coordinator Guide.
Unreadable set display
If the trouble is with telephones with two-line displays
1. Press ≤•‡.
2. Press UP or DOWN to adjust the display to the desired level.
3. Press OK.
Tip - The numbe

Call-by-call routing

Call-by-call routing
By default, incoming calls are routed based on the Called Party Number. You program the received number length to include however many digits your system requires. This many digits from the end of the called party number are then used as Received Digits to find a target line.
For example, the incoming calling party number is 800-555-1234. The received digit number length is 4. Therefore, 1234 is used to route the call.
Depending on the protocol and the service, alternate routing maps may be defined.

STATUS Off Bootstrap

STATUS Off Bootstrap mode - The boot code cannot be booted. During bootstrap mode, VOICE, DATA, NET 1-4 LEDs will be red. Green (flashing) Unit is powering up. On power-up the STATUS LED flashes rapidly for 5 seconds, during which time the user may escape to bootstrap mode from the CRAFT port. Green (solid) Power is on and the unit is functioning normally. POWER Off No power. Green AC power is operational. Amber AC power has failed. Battery backup is active. VOICE Off All ports are inactive. Green (solid) At least one port is off hook. Green (flashing) At least one port is ringing. Amber At least one port is in test. Red Fault condition. DATA Off Port is administratively shut down. Green Layer 2 is up on a NET interface. Red Layer 2 is down on a NET interface. NET 1-4 Off Port is administratively shut down. Green Link is up and in normal operation. Amber Port is in test. Red An alarm condition is present. LAN 1-2 Off Link is down or port is administratively shut down. Green (solid) 10BaseT link is up. ETH 0/1-0/2 Green (flashing) 10BaseT link is up and traffic is flowing. (Rear Panel) Amber (solid) 100BaseT link is up. Amber (flashing) 100BaseT link is up and traffic is flowing.

TIME REMINDER

TIME REMINDER
Allows you to be reminded that it is a certain time. You are reminded by hearing tones that are generated from the phone's speaker. If your phone has a display the word Reminder will also appear. The time must be entered in military time.
Operation To set the time:  Press the Speaker button SPKR  Wait for the dial tone
 Press the Feature button FEAT then dial Or
 Press the Time Reminder button TM REMD  Dial the time in military time H H : M M  Wait for a confirmation tone
 Press the Speaker button SPKR
To cancel:
 Press the Speaker button SPKR  Wait for the dial tone
 Press the Feature button FEAT then dial Or
 Press the Time Reminder button TM REMD then dial  Wait for a confirmation tone
 Press the Speaker button SPKR
TONE/VOICE CALLING
The calling mode for intercom calls may be switched between voice and tone signaling. ADIX may be programmed to allow either the calling party or the called party to have control of the signaling mode.
Operation
Calling party control:
 Dial intercom extension number  Dial
Called party control:
 Press the Feature button FEAT then dial Or
 Press the Tone button V/T if assigned as a Multipurpose button and the lamp will be solid red

FLEXIBLE LED FLASH

FLEXIBLE LED FLASH RATES
Description
The flash rates used with the various Flex buttons on the iPECS phone can be adjusted on a system wide basis to meet the customer’s needs. Up to 29 different functions can be assigned any one of 15 different flash rates.
Operation
System
System implements flash rates automatically based on database entries.

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

Feature Setup / Programming SMDR

Feature Setup / Programming SMDR

Digit Counting
With Digit Counting, SMDR can selectively keep track of toll calls. For example, if the digit count is nine, SMDR won’t include toll calls within the home area code. Digit Counting permits SMDR to include only the types of calls you want to monitor.
Digit Masking
Digit Masking lets you “X” out portions of the number dialed on the SMDR report. A digit mask of seven, for example, masks out all exchange codes (NNXs) and local addresses. Digit Masking makes it easier to keep track of calling patterns, without having to interpret each individual number. You can also use Digit Masking to block out access and security codes.
Duration Monitoring
SMDR can include calls of any duration, or only those that last longer than the interval you specify. If you want to keep track of all trunk activity, use a short duration. To keep track of only significant usage, use a longer duration.
Extension Exclusion
You can selectively exclude extensions from the SMDR report. This ensures privacy for high-profile callers. For example, the company attorney negotiating a merger may not want his calls to show up on an in-house report.
PBX Call Reporting
If you system is behind a PBX, you can have SMDR monitor all traffic into the PBX or just calls placed over PBX trunks. The SMDR record can include all PBX calls (including calls to PBX extensions) or just calls that include the PBX trunk access code.
Serial and USB SMDR Communication
The system is compatible with both serial and USB SMDR devices. This gives you many SMDR output options. For example, you can output the SMDR report to a high speed printer or send it to disk through a PC’s serial or USB port (requires the USB driver which can be downloaded from the NEC Technical Support web site - ws1.necii.com).
Trunk Exclusion
Use Trunk Exclusion to exclude certain trunks not subject to per-call charges (like WATS lines) from the SMDR report. This makes call accounting easier, since you review only those calls with variable costs.
Usage Summaries
SMDR can automatically print daily, weekly and monthly call activity summaries. Each summary includes the total number of regular trunk calls and ISDN trunk calls, and the costs for each type. The daily report prints every day at midnight. The weekly report prints every Sunday night at midnight. The monthly report prints at midnight on the last day of the month.
Extension Name or Number
The SMDR report can include an extension’s name or extension number. Choose the method that makes it easier for you to track call usage.
SMDR Enhanced for Caller ID
With software version 4.0E or higher, the SMDR output is enhanced to include up to 16 or 24 characters of the Caller ID name information (depending on the view option selected in Program 35-02-18). You can select to display the Caller ID number or name or the DID number. If you wish to display the Caller Name in the “DIALLED NO./CLI” and “ACCOUNT" area, select "2" in the updated Program 35-02-15 and "1" in Program 35-02-17.

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
1. [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.

ADD/DEL

ADD/DEL : Addition/Deletion ESP : External Speaker AL : All FWDG : Forwarding ALM : Alarm FLSH : Flash ANS : Answer GUARD : Outgoing Guard Time ANSWR : Answer H : High ASSGN : Assignment HR : Hour AUT : Automatic ICM : Intercom (Extension) AUTANS : Autoanswer IMDT : Immediate BLANK : Service Class IN : Incoming BNCE : Bounce INC : Incoming Signal Detection Time Assignment BTN : Button INDV : Individual CAL : Call INTRPT : Interruption CANCLD : Canceled L : Low CKT : Circuit LCD : Liquid Crystal Display CNF : Confirmation LN : Line CL : Class LOOP : Loop Off-Guard Assignment CLD : CO Line Display LNR/SPD : Last Number/Speed Dial CLR : Clear M : Medium CLS : Class MAN : Manual CONN : Connection MF : Dual-Tone Multifrequency (DTMF) DESG : Designation MIN : Minimum DGT : Digit MOH : Music On Hold DISP : Display MSTR : Master DISTM : Disconnection Recognition Time NUM : Number DIVERT : Diversion NT : Night Mode DLY : Delay Signal Time OUT : Outgoing DP : Dial Pulse OG TM : Outgoing Time OUt Assignment DSS : Direct Station Selection OV : Over DY : Day Mode RNGTONE : Ringing Tone

ADIX Digital Key Telephone

Equipment Cable ADIX Digital Key Telephones: 1-pair (twisted) #22/24 AWG ADIX Digital KT’s used with IX-BPAD or ADIX Data Modules: 2-pair (twisted) #22/24 AWG ADIX Direct Station Selection Units: 1-pair (twisted) #22/24 AWG ZT-D Key Telephones: 2-pair (twisted) #22/24 AWG ZT-D KT’s equipped for off-hook voice announce and hands- free answerback: 3-pair (twisted) #22/24 AWG Single-Line Telephones (500/2500 type): 1-pair (twisted or quad) #24 AWG

ACTIVE CPU (MFIM)

ACTIVE CPU (MFIM) DISPLAY ............................................... 349 6.2 ALTERNATE ATTENDANT ..................................................... 350 6.3 ATTENDANT POSITIONS ....................................................... 351 6.4 ATTENDANT RECALL ............................................................ 352 6.5 ATTENDANT STATION PROGRAM CODES ......................... 353 6.6 CABINET ALARM.................................................................... 356 6.7 CALL FORWARD, ATTENDANT ............................................ 357 6.8 CALL FORWARD CO/IP OFF-NET ......................................... 358 6.9 CALL/QUEUING....................................................................... 360 6.10 DAY/NIGHT/TIMED/SCENARIO RING MODE ...................... 361 6.11 DSS CONSOLES ................................................................... 363 6.12 DISABLE OUTGOING CO/IP ACCESS................................. 366 6.13 DND OVERRIDE .................................................................... 367 6.14 EMERGENCY CALL ATTENDANT ALERT .......................... 368 6.15 EZ-ATTENDANT .................................................................... 370 6.16 FEATURE CANCEL............................................................... 371 6.17 INTRUSION ............................................................................ 372 6.18 LCD DISPLAY FORMAT CONTROL..................................... 373 6.19 SYSTEM CLOCK SET ........................................................... 374 6.20 IP-ATTENDANT ..................................................................... 376
7. SLT .......................................................................................377
7.1 BROKER CALL........................................................................ 377 7.2 HOWLER TONE....................................................................... 379 7.3 SLT MESSAGE WAIT INDICATION........................................ 380 7.4 SLT NAME ENTRY .................................................................. 381 7.5 TRANSFER CLI TO SLT....

CO LINE RELEASE GUARD

CO LINE RELEASE GUARD TIME
Description
To assure that the PSTN switching equipment has sufficient time to restore to the idle condition, the system will hold analog CO lines in a busy state to users after release of a CO line by a station. The time between the station disconnect and when the system changes the CO line status from busy to idle is the CO Line Release Guard time.
Operation System Operation of this feature is automatic.
Conditions
Programming
SYSTEM 1. CO Release Guard Timer (PGM 180-Btn 16)
Related Features

Office systems based on the IP500v2

Office systems based on the IP500v2 control unit can run in a number of modes: · IP Office Standard Version
· IP Office Essential Edition - Quick Version
· IP Office Essential Edition - Norstar Version
· IP Office Essential Edition - PARTNER® Version
· Avaya Branch Gateway
This is a special mode used for B5800 control units and is not covered by this documentation. Refer to the separate Avaya Branch Gateway documentation.

Dial 911

BASIC FEATURES
MAKING A 911 SUPPORT CALL
Operation
From an outside or inside line:  Dial 911
You will be connected to your local Public Safety Answering Point (PSAP).
RECEIVING AN INCOMING CALL
There are two ways you may answer incoming calls; either with the receiver in the cradle or with the receiver not in the cradle.
Operation
To answer calls with the receiver in the cradle:
When your phone rings and the outside line button(s) flash red:
 Lift the receiver and you will be connected to the outside caller
To answer calls with the receiver not in the cradle:
When your phone rings and the outside line button(s) flash:  Lift the receiver
 Press the Answer button ANS and you will be connected to the outside caller
The outside line button will flash green.

DAY/NIGHT/TIMED

DAY/NIGHT/TIMED/SCENARIO RING MODE
Description
The system clock automatically controls the Ring Mode. Ring assignments are applied based on the time of day and day of week. Four modes of ring (Ring Assignments) are provided, Day, Night ,Timed and Scenario.
The Attendant controls the system Ring Service mode changing from Auto Service Mode to Day, Night ,Timed or Scenario service mode. Based on the service mode selected, different ring assignments, COS and answering privileges are invoked for the system users.
Operation
System Attendant
To change Day/Timed/Night Ring Mode manually;
1. Press the [DND] button.
2. Dial 1~5. (1: Day mode, 2: Night mode, 3: Timed mode, 4: Auto Service mode, 5: Scenario mode)
3. If you set Scenario mode(5), you should set scenario group(MFIM100/300/600 : 01 ~16, MFIM1200: ~32).
4. Press the [HOLD/SAVE] button.
To set Day/Timed/Night Ring/Scenario Mode automatically (Auto Service Mode Control)
1. Press [PGM] button, [PGM] button LED flashes 60 ipm, [SPEAKER] LED lights steady.
2. Dial the Attendant Station Program code “#8” to toggle between manual and Auto Mode control.

IX-CPUP/HW Controller Card

The IX-CPUP/HW Controller Card contains a 32-bit processor that controls all ADIX APS system functions. When utilized, either an IX-CPU20/ MEM-M or IX-CPU20/ MEM-L CPU/Memory Module is also required. The IX-CPUP/HW Controller Card has connectors for the CPU/Memory Module and supplies 4 MB of RAM for back-up memory. In addition, the IX-CPUP/HW Controller Card performs highway controller functions, synchronizes digital trunk clock timing and system highway timing (IX-PLLU). The CPU/Memory Module contains the 68EC020/25 MHz Central Processing Unit (CPU), as well as 4 MB of Intel® flash memory for the ADIX operating system. These components replace the IX-CPU32-1, IX-CPU32M, and IX-CPU16M.

Total Access 900e Series

Total Access 900e Series BBU (P/N 1175044L1/L2)
The BBU connects to the Total Access 900e Series through a 6-foot charge/discharge, 2-conductor wire with a keyed modular plug (included with the BBU). The 1175044L1 BBU is a low profile wallmount configuration. It can be rack mounted with the appropriate 19-inch (P/N 117547L1) or 23-inch
(P/N 1175048L1) rackmount adapter brackets. The 1175044L2 is an equivalent BBU with a hinged front access door.
Features of the BBU, P/N 1175044L1/L2, include the following:
• No-spill battery design
• Compact wallmount or rackmount box
• Double BBU rack mounting available
• 7 Ahr battery (up to 8 hours of backup, depending on load)
• Modular plug (provides quick and easy installation)
• All mounting hardware included

Outgoing Messages (OGM)

Outgoing Messages (OGM)
Enter 1 to start
or 0 to stop.
The manager extension can record three kinds of greeting messages (OGM) as follows:
1. DISA message: Used to greet and guide callers so that they access
extension group or outside party without operator assistance.
2. Incoming Call Distribution Group message: Used to greet and guide
callers to an incoming call distribution group.
3. Timed Reminder message: Used for a wake-up call message when the
extension answers the Timed Reminder.

AUTO CALLED NUMBER REDIAL (ACNR)

AUTO CALLED NUMBER REDIAL (ACNR)
Description
This feature allows a station user to request and have the system retry a busy or no answer external call until the call is connected or the feature is cancelled.
Operation iPECS Phone To activate ACNR while receiving busy, no answer
1. Press the [REDIAL] button.
2. Hang-up handset, or press [SPEAKER].
To cancel ACNR while idle
1. Press flashing [REDIAL] button.
To cancel ACNR during an ACNR attempt
1. Lift the handset or press the [MUTE] or flashing [REDIAL] button.
System
1. The system initiates the ACNR process, starting the ACNR Pause Timer.
2. At expiration of the timer, the system activates the station’s speakerphone with the microphone in the mute mode.
3. The system attempts the previous call.
4. When the called party answers, the user may answer by lifting the handset or pressing the [MUTE] button to communicate with called party.

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.

Network eManager

Network eManager Data Import/Export Features ............................................................................2-49 Network eManager DID/DNIS Assignments - Import/Export .....................................................2-49 Network eManager Station Directory - Import/Export ................................................................2-53 System Speed Dial Export/Import .............................................................................................2-55 Station Speed Dial Export/Import ..............................................................................................2-56 Network eManager Network Calling Number Upload ................................................................2-60 Connection Entry for Dial-up ............................................................................................................2-62 Help Topics/Context Help ................................................................................................................2-68 About Network eManager ..........................................................................................................2-68 No Response from Device .........................................................................................................2-68 Network eManager DESI Integration / Telephone Keystrip Printing ................................................2-69 Manage Stand-Alone SES Systems .........

Hot Desk

Hot Desk
Description
A DKT may be assigned as Hot Desk phone allowing users (Agents) to login with the ipLDK-60. When an Agent logs in, the Hot Desk will become active and will take on the attributes defined for the Agent’s station number. When the Agent logs off, the Hot Desk station becomes inactive and the Agent’s calls are forward to the user-entered destination. A different Agent may then login through the inactive Hot Desk station.
Operation
To login at a Hot Desk,
1. Lift the handset or press the On/Off button.
2. Dial the Agent’s authorization code followed by ‘#’.
3. The Hot Desk becomes active with the Agent’s station configuration (station number, COS, ring assign, etc.) and receives incoming calls.
To logout a User,
1. Press the [TRANS/PGM]
2. Dial ‘*’ ‘*’, or press the programmed Agent Logout button.
3. Use the Up/Down buttons to select a Call Forward type.  Station
 Off-net speed
 Mobile extension  VMIB Voice Mail  VM group
4. Press the [HOLD/SAVE] button, the system saves the Agent’s station configuration and the Hot Desk returns to inactive.

Caller ID - Flexible Ringing

The Caller ID - Flexible Ringing feature provides several different options for rerouting calls based on the Caller ID received.
Reject/Reroute “Private” Caller ID Calls
When an analog trunk call is received with “Private” Caller ID information, the SL1100 can reject the call by playing a VRS message or it can route the call to an alternative extension or incoming ring group programmed in PRG 22-18-01.
Reject/Reroute Based on Entry in SPD Table
When an analog, ISDN or IP trunk call is received with regular Caller ID information, the SL1100 can reject the call by playing a VRS message if the Caller ID number matches the Speed Dial group number programmed in PRG 22-16-01 and Speed Dial entry in PRG 13-02-01 and PRG 13-04-01. The analog, ISDN or IP trunk call can also be routed to an alternative extension or incoming ring group if the Caller ID number matches the common or group Speed Dial table (PRG 13-04).
This option can block calls on all trunks or it can be set on a per-trunk basis. Programming Examples for Flexible Ringing by Caller ID:
Various ring tone patterns and melodies for incoming calls are available when using PRG13-04-05; Flexible Ringing by Caller

IP (SIP) trunks are required

A minimum of four IP (SIP) trunks are required due to the NEC KTS infrastructure setup.
NEC recommends that the requirements and programming are completed with as much information as possible before scheduling an activation appointment with 123.Net.
1.5 Conditions
The following limitations apply:
 Some private IP network ranges conflict with SIP Trunking Service providers. This can cause issues when connecting to the SIP Trunking service. Private ranges reserved for the customer’s LAN are:
10.x.x.x
192.168.0.x through 192.168.10.x
 G.711 pass-through is supported for faxing. T.38 is not supported.
 All SIP packets sent to the 123.net SIP server must originate from the public IP address 123.net assigned for your SIP trunks. NAT is not supported.
 Specific traffic from 123.net that will need to be allowed to reach the SL1100:


Features

Account Code – Forced/Verified/Unverified Account Code – Forced/Verified/Unverified Account Code Entry Account Code Entry Add-On Conference Conference All Call Page Paging, Internal Alphanumeric Display Alphanumeric Display Analog Line Extender (Dterm Analog EXT) Not Supported Ancillary Device Connection Ancillary Device Connection Answer Hold Answer Hold Answer Key Answer Key Assigned Night Answer (ANA) Direct Inward Line (DIL) Attendant Add-On Console Direct Station Selection (DSS) Console Attendant Camp-On Call Waiting / Camp On Attendant Positions Attendant Call Queuing Attendant Station Outgoing Lockout Code Restriction, Dial Block Attendant Transfer Transfer Authorization Code Code Restriction Override Automated Attendant Voice Response System (VRS) Automatic Answer with Delay Message Voice Response System (VRS) Automatic Call Distribution Automatic Call Distribution (ACD) Automatic Callback Callback Automatic Day/Night Mode Switching Night Service Automatic Hold Hold Automatic Number Indication (ANI) on T1 T1 Trunking (with ANI/DNIS Compatibility) Automatic Redial Repeat Redial Automatic Release Automatic Release Automatic Route Selection (ARS) Automatic Route Selection
Automatic Trunk-to-Trunk Transfer Call Forwarding, Off-Premise Background Music – Multiline Speaker Background Music Background Music Over External Speakers Background Music Barge-In Barge In Battery Backup – System Memory Battery Backup – System Memory Battery Backup – System Power Battery Backup – System Power Busy Lamp Field on Multiline Terminals One-Touch Calling Call Alert Notification Off Hook Signaling Call Appearance (CAP) Keys Call Appearance (CAP) Keys Call Arrival (CAR) Keys Call Arrival (CAR) Keys Call Forward – All Call Call Forwarding Call Forward – Busy/No Answer Call Forwarding Call Forward – Centrex Not Supported Call Forward – Display Call Forwarding Call Forward – Off-Premise Call Forwarding, Off-Premise Call Forward – Split Call Forwarding Call Monitoring Call Monitoring Call Park – System Park Call Pickup – Group Group Call Pickup Call Pickup Direct Directed Call Pickup Callback Request Message Waiting Caller ID Call Return Caller ID Call Return Caller ID Indication (Analog Trunks) Caller ID Cascade CPU Not Supported Centralized Voice Mail (with E&M Tie Lines) Not Supported Class of Service Class of Service Clock/Calendar Display Clock/Calendar Display CO Message Waiting Indication CO Message Waiting Indication CO/PBX, Tie Line Digit Restriction Code Restriction Code Restriction Code Restriction

Consecutive Speed Dial Speed Dial – System/Group/Station Cordless Telephone Connection Cordless Telephone Connection Customized Message Selectable Display Messaging Data Line Security Data Line Security Delay Announcement Voice Response System (VRS) Delayed Ringing Delayed Ringing Dial 0 For Attendant Operator Dialed Number Indication Service (DNIS) T1 Trunking (with ANI/DNIS Compatibility) Digit Insertion Automatic Route Selection Digital Line Extender (Dterm ISDN EXTender Plus) Not Supported Digital Voice Mail Digital Voice Mail Direct Inward Dialing (DID) Direct Inward Dialing (DID) Direct Inward System Access (DISA) Direct Inward System Access (DISA) Direct Inward Termination (DIT) Direct Inward Line (DIL) Direct Paging Access Paging, Internal Direct Station Selection (DSS) One-Touch Calling Distinctive Ringing Distinctive Ringing, Tones and Flash Patterns Do Not Disturb (DND) Do Not Disturb Door Lock Release Relays Door Box Door/Monitor Telephone Door Box DP to DTMF Switching Pulse to Tone Conversion Drop Key Drop Key Dterm Analog Cordless Terminal Not Supported Dterm Cordless II Terminal Dterm Cordless II Terminal Dterm Cordless Lite II Terminal Dterm Cordless Lite II Terminal Dterm Handset Cordless Dterm Handset Cordless Dterm IP Gateway System Dterm IP Gateway System

advance to the option line

advance to the option line (for editing data):
1. While the option is flashing (e.g., 01 in the above example), press HOLD twice.
• Pressing Volume up and Volume down will scroll through the options for the selected port.
To select another extension and advance to the option line (for entering data):
1. While the option is flashing (e.g., 01 in the above example), press HOLD + Enter extension number + HOLD.
To navigate by extension or station instead of option:
1. While the option is flashing (e.g., 01 in the above example), press [>>] or TRANSFER.
• Select either the extension number or station port.
2. Press Volume Up and Volume Down to scroll through the same option for stations or extensions, depending on which you selected.
3. Press [Select] or HOLD to advance (drill down to) the option.

AUTO FAULT DETECTION AND RECOVERY

AUTO FAULT DETECTION AND RECOVERY
Description
4. CO/IP
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 number

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

Telephone Switching Systems

Telephone Switching Systems
PBX Number AT&T Partner Plus 110 AT&T Partner 2 111 AT&T Partner ACS 112 AT&T Merlin Legend/Magix 113 AT&T System 25 114 Comdial DSU/Unisyn 2-digit ext. 121 Comdial DSU/Unisyn 3-digit ext. 122 Inter-Tel GMX 12/24 140 Iwatsu Adix 8/24 (S/M Series) 150 Iwatsu Adix 16/48 151 NEC Electra, Level I (release 2.00+) 170 NEC Electra II (release 3.00+) 171 NEC/Nitsuko 124i/128i/384i/DS1000/2000 180 Panasonic KXT 1232 11D 190 Panasonic KXTD 1232 191 Panasonic DBS 824 192 Panasonic DBS Alternate 1 193 Panasonic 576 194 Panasonic KX-TDA 195 Samsung DCS 2.2/Compact 220 Sprint CTX/MTX 230 Siemens HiCom 150e Office Pro/Office Com 240 Siemens HiCom 150e Office Point 241 Toshiba DK-8 250 Toshiba DK-16 251 Toshiba DK-16e, DK-40 252 Vodavi Starplus DHS 260 WIN 36D 270 WIN 100D 271 WIN 48/440CT

AUTOMATIC PAUSE INSERTION

AUTOMATIC PAUSE INSERTION
Description
In addition to a manually entered Pause, the system will automatically pause dialing to allow for potential connection delays. The pause will be inserted when any of the following occur:  After a Flash is encountered in a Speed Dial number,
 After a PABX access code is encountered in a Speed Dial or redial number,
 After a Pulse to Tone Switchover is encountered in a Speed Dial or Redial number.  When a connect message is received on an ISDN Line.

ALARM SIGNAL/DOOR BEL

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.

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.
5-

Message Wait/Callback

Message Wait/Callback Enable 556 *556 556 *66 SLT Message Wait/Callback Return 557 *557 557 *67 SLT Speed Dial Access 558 *558 558 #8 SLT Cancel DND/FWD/Pre MSG 559 *559 559 559 SLT CO System Hold 560 *560 560 560 SLT Program Mode Access 561 *561 561 561 SLT Attendant Unavailable 562 *562 562 562 Alarm Reset 565 *565 565 *565 Group Call Pickup 566 *566 ** *1 Universal Answer 567 *567 567 2 Account Code with bin 568 *568 568 568 Walking COS Code 569 *569 569 569 ACD Supervisor On/Off Duty 571 *571 571 571 ACD Supervisor Login 572 *572 572 572 ACD Supervisor Logout 573 *573 573 573 ACD Help Code 574 *574 574 574 ACD Calls In Queue Display 575 *575 575 575 ACD Supervisor Status Display 576 *576 576 576 ACD Supervisor Monitor 577 *577 577 577 ACD Reroute Queued Call w/answer 578 *578 578 578 ACD Reroute Queued Call w/o answer 579 *579 579 579 Camp-On Answer 600 *600 600 600 SLT Call Parking Locations

LOOP START TRUNK

LOOP START TRUNK
Ringer Equivalence No 0.5B Service Order Code 9.0F Facility Interface Code 02LS2 Registered Connection RJ21X
GROUND START TRUNK†
Ringer Equivalence No 1.0B Service Order Code 9.0F Facility Interface Code 02GS2 Registered Connection RJ21X
CALLER ID TRUNK
Ringer Equivalence No 0.5B Service Order Code 9.0F Facility Interface Code 02LS2 Registered Connection RJ21X, RJ11*
E&M TIE LINE CARD TYPE 1
Facility Interface Codes TL31M Service Order Code 9.0F Registered Connection RJ2GX TYPE 2
Facility Interface Codes TL32M Service Order Code 9.0F Registered Connection RJ2HX
NOTE:
T-1 SERVICE (SF)
Facility Interface Code 04DU9-BN Service order Code 6.0P Registered Connection RJ48C
T-1 SERVICE (ESF)
Facility Interface Code 04DU9-1KN Service order Code 6.0P Registered Connection RJ48C
DID TRUNK*
Facility Interface Codes 02RV2-T Ringer Equivalence No. 0.0B Service order Code 9.0F Registered Connection RJ21X
OPS LINE CARD
Facility Interface Codes. OL13C Service Order Code 9.0F Registered Connection RJ21X
ISDN BRI CARD
Facility Interface Codes. 02IS5 Registered Connection RJ49C#
ISDN PRI CARD
Facility Interface Codes. 04DU9-1KN Registered Connection RJ49C or

SILENT TEXT MESSAGE

SILENT TEXT MESSAGE
Description
Silent Text Messaging is used to respond to an OHVO call without disconnecting the existing call. Silent Text Messages are sent by pressing a pre-programmed message button or [DND] button.
Operation iPECS Phone To program a {TEXT MESSAGE} button [PGM] + {FLEX} + [PGM] + ‘51’ + Display Message code (‘00’-‘20’) + [SAVE]
To respond to an Off-Hook Voice Over call with a Text message
1. Upon receiving announcement, press the desired {TEXT MESSAGE} or [DND] button, text message is sent to the display of the calling party, OHVO call is terminated.

Always use the Power Switch on the Main/Expansion KSU.

• If Expansion KSU(s) are installed, turn the power on/off in the order of Expan-
sion 2 KSU, Expansion 1 KSU and then Main KSU.
• DO NOT power off by disconnecting the AC (or DC for battery backup) power.
Always use the Power Switch on the Main/Expansion KSU.
• All calls in progress will be cut off when the system is powered off, and if the
Power Failure is set, the applicable extension will become operational.
• All user's setting (such as Call Forward, Camp-On, etc) will be kept.

T1 Trunking (with ANI/DNIS Compatibility)

T1 Trunking (with ANI/DNIS Compatibility)

Description
The T1/PRI Interface gives the system T1 trunking ability. This blade uses a single universal slot and provides up to 24 trunk circuits. In additional to providing digital-quality trunking, the T1/PRI Interface allows you to have maximum trunking ability with fewer blades. This in turn makes more universal slots available for other functions.
You can program each T1/PRI for any combination of the following trunks:
 CO loop start  CO ground start  Direct Inward Dialing  Tie Lines 2
CD-PRTA uses the first block of 24 consecutive trunks. For example, if you have a CD-4COTB with PZ-4COTF installed for trunks 1~8, the T1/PRI Interface automatically uses trunks 9~32. If you have CD-4COTB with PZ-4COTF installed for trunks 1~8 and 17~24, the T1/PRI uses trunks 25~48. The T1/PRI Interface cannot use trunks 9~16 (even if available) since they are not part of a consecutive block of 24 trunks. Each T1/PRI requires that 24 consecutive ports be available in the system even if not all the ports are used otherwise the blade does not function.
The CD-PRTA can be programmed as a 4/8/12/16/20/24 port Fractional T1/PRI.
ANI/DNIS Compatibility
The system is compatible with Telco T1 Automatic Number Identification (ANI) and Dialed Number Information Service (DNIS) services. A compliment to Caller ID service, ANI/DNIS Compatibility provides:
 Receive Format
The Receive Format must be set as  ANI  DNIS  in Program 34-09-01 option 4, which is treated as a Feature Group D format. (Example of ANI Information KP009727517645STKP7100ST.)

CALL TRANSFER

CALL TRANSFER
Description
SIP Phone supports Consultation Transfer and Blind Transfer. Consultation Transfer is also separated to Screened and Unscreened Transfer by System.
Operation Consultation Transfer
1. During talking on a call, press [TRANSFER] button.
2. Dial destination number.
3. Destination is ringing and answers the call – Screened
Or, Destination is ringing - Unscreened
4.Press [TRANSFER COMPLETE] button or OnHook. Blind Transfer
1. During talking on a call, press [BLIND TRANSFER] button.
2. Dial destination number.
3. Press [SEND] or Call Complete button.
4. SIP Phone will goes to idle automatically. Transfer Recall
1. When call transfer failed or no answer, there will be recall to the transferor SIP
Phone.

H.323 v4 Service

TRUNKING
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

TRANSFER CLI TO SLT
Description
A SLT phone can be received CLI from internal caller instead of station number by programming when it’s tranfering.
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 transfering to another SLT
3. The SLT can see CLI of CO instead of station number

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.
Operation iPECS Phone Placing a Voice Over (OHVO) while receiving busy
1. Dial ‘#’ or press a pre-programmed {OHVO} button.
2. After splash tone, begin announcement.

TRANSFER CLI TO SLT

TRANSFER CLI TO SLT
Description
A SLT phone can be received CLI from internal caller instead of station number by programming when it’s tranfering.
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 transfering to another SLT
3. The SLT can see CLI of CO instead of station number

Trunk Card Total

Trunk Card Total 4 *1 Total 8*1 Free Slot LCOT4 48 LCOT8 LCOT16 T1 24 PRI23 IP-GW4 24 IP-GW4E IP-GW16 Extension Card Total 4 Total 8 DHLC8 48 DLC8 DLC16 SLC8 SLC16 MSLC16 IP-EXT16 CSIF4 24 CSIF8 OPB3 2 4 Option Card Slot/Free Slot CTI-LINK 1

Hotel/Motel - Toll Restriction (When Checked In)

Hotel/Motel - Toll Restriction (When
Checked In)
With Toll Restriction (When Checked In), you can control guest's long distance dialing automatically when they check in. This option allows you to set up two completely different Toll Restriction modes. The first mode determines the type of call your staff can place from a room telephone when the room is checked out (Service Code 639). This is the business mode Toll Restriction. The second mode sets the Toll Restriction limits for your guests as soon as you check them in (Service Code 638). This is the hotel mode Toll Restriction.
In the checked out mode, for example, you may want to allow your staff to call locally and within your area code. This would allow them to contact suppliers and other service providers without going to the front desk each time. In the checked in mode, however, you may want to completely restrict outgoing calls and force your guests to use your metered services. (This can also tie into Single Digit Dialing on page 1-407.)
In many cases, such simplified Toll Restriction scenarios may be adequate. However, since each mode uses the full abilities of the system Toll Restriction programming, you can make the calling restriction as elaborate as it needs to be.
Toll Restriction (When Checked In) also allows you to change the Toll Restriction of a room telephone while the room is checked in. This allows you to provide more permissive Toll Restriction to high priority guests. It also allows you to enforce less permissive dialing privileges to guests if you suspect the potential for abuse.
If a room Toll Restriction is changed using access code 637, that room keeps the new setting until it is either changed using access code 637 or in system programming.
The following diagram shows the basic operation of Toll Restriction (When Checked In). When checked in, the room telephone follows the Check In Toll Restriction Level (set in PRG 42-02-02). When checked out, the room telephone follows the normal Toll Restriction Level (set in PRG 21-04-01). Both levels interact with the dialing restrictions set up

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.