Operation modes

Operation modes ASUS WL-330N provides five operation modes: Wireless Router, Access Point (AP), Hotspot (WiFi Account Sharing), Repeater, and  Wireless Network Adapter. NOTES: •    Use  the  Device  Discovery  utility  in  the  support  CD  to  get  the  WL-330N's WL-330N's's dynamic IP address. •    Before  setting  up  the WL-330N to Hotspot, Repeater or Network Adapter WL-330NtoHotspot,RepeaterorNetworkAdapter to  Hotspot,  Repeater  or  Network  Adapter mode,  ensure  that  you  connect  the  computer  and  the  WL-330N  using  a network cable. •      If  you  cannot  switch  modes  successfully,  reset  the  system  to  its  factory default  settings  through  pressing  the  Restore  button  while  the  ASUS  WL330N is ON. Wireless Router mode In the Router mode, connect the  WL-330N to an ADSL or a cable modem, and the network clients share an IP address for Internet connection. In this mode, the  WL-330N provides wireless signals, NAT, firewall, and IP sharing functions to the wireless clients.

MERLIN Messaging

Apendix 2 – Installing the System Administration Software on a LAN
The PARTNER or MERLIN Messaging System Administration (SA) application software can be used instead of the Touch-tone user interface to configure and program the messaging system. You must use the SA application to configure the Unified Messaging application.
Installing the SA Application
The SA application is installed on a PC that is connected to the LAN.  Only a user on that PC can install the SA application.  The SA application will communicate with the Messaging system through its LAN port.  Installation of the MERLIN Messaging or PARTNER Messaging SA application can be performed from the MERLIN Messaging System Release 3.0 or PARTNER Messaging Release 6.0 Library CD. You select “PARTNER Messaging or MERLIN Messaging System Administration” under “Install Software” from the main Library screen. This will automatically launch a windows software install wizard. Follow the instructions displayed by the install wizard to complete the installation.
Launching the SA Application
To 1. start the SA application, perform one of the following steps:
Dou▪ ble-click on the MERLIN or PARTNER Messaging Administration desktop short cut.
Sel▪ ect the MERLIN or PARTNER Messaging Administration short cut from the Start menu.
Fro▪ m the Start menu, select Programs > MERLIN Messaging Release 3.0 or PARTNER Messaging Release 6.0. (This is the default location.)
The2. Messaging Login window appears, displaying the Messaging Login dialog box.
In 3. the IP address or host name box, enter the Fixed IP address of the Messaging System module that was loaded into the message system (See Section 2)
In 4. the Login box, enter sysadmin.
In 5. the Password box, enter the system administration password. (If this is your first time logging in to the system, click the OK button. You are prompted to enter the password.)
Cli6. ck the OK button.
Once you log in successfully, you can start administering the Messaging system. The SA application windows display the current Messaging settings. For more information regarding installation and use of the SA application, see the MERLIN or PARTNER Messaging System Administration Getting Started Guide under Documentation, System Administration contained on the Library CD

Tech Spec

Technical Specifications
AC / DC Power Adapter
› 100~ 240 V AC @ 50/60Hz › DC48V, 1A
Physical Dimensions & Weight
› Width: 278mm
› Depth: 233mm
› Height: 34mm
› Weight: 0.86kg
Firewall
› General security policy, access control › Web site restriction, port forwarding, port triggering › NAT / NAPT, DMZ host, rule-based packet filtering › Connection information, security log
Security
› Virtual Private Network (IPSec, PPTP, L2TP) › Remote administration access control › Digital certificate management
Quality of Service (QoS)
› General QoS profile, bandwidth restriction › Rule-based traffic priority and traffic shaping › DSCP / 802.1p / priority queue configuration › Connection utilization and statistics
Routing
› Static routing (routing table management) › Dynamic routing (RIP v1/v2) › NAT / NAPT, IGMP / Multi-cast
L2 Switching
› 8-port 10 / 100 BASE-TX with 4 built-in PoE (total     PoE budget: 20 Watts)
› STP / RSTP, VLAN, LAN bridge › Broadcast and multi-cast storm control, loop detect
Wireless LAN
› 802.11 b/g/n (2.4 GHz)
› WEP, WPA, WPA2 or WPA / WPA2 and web    authentication
› 802.1x for enterprise
› Multiple SSIDs (virtual APs), MAC filtering › WPS (WiFi protected set up) support › Wireless multi-media (WMM) › Channel width and frequency selection
IP-PBX / SIP
› Ericsson-LG advanced IP-PBX features (call transfer,     call forward, call park, call pick up, call waiting camp     on, CO queuing, speed dial, station groups, mobile     extension, 3-party voice conference, IP fax relay     [T.38])
› SIP trunk – 4 trunks with DECT, 6 trunks w/o DECT › CO trunk – only one option can be mounted on the     SBG-1000 either in the factory or locally (1CO, 2CO,     4CO, 1 BRI or 2BRI)
› Extension – 23 IP extensions – 6 Ericsson-LG    proprietary DECT terminals › Built-in SIP proxy, registrar, user agent, failover to     PSTN
Administration
› Web-based administration (HTTP / HTTPS) › CLI (telnet and telnet over SSH) › User management (role and permissions) › Date and time (NTP / TOD with daylight savings   option)
› Smart installation wizard
Services
› File server (disk management, back up and restore) › Printer server (LPD, IPP, Microsoft shared printing     support)
› DHCP / DNS server, dynamic DNS, UPnP
Management
› Device information and map view, SNMP, TR-069 › Network connection management, monitoring and     diagnostics
› Email notification and syslog support, system log

Service Code Setup, Administrative

Use Program 11-15 : Service Code Setup, Administrative (for Special Access) to customize the special access Service Codes which are used by the administrator in the Hotel/Motel feature. You can customize additional Service Codes in Programs 11-10 ~ 11-14 and 11-16. The following chart shows: • The number of each code (01 ~ 14). • The function of the Service Code. • What type of telephones can use the Service Code. • The default entry. • Programs that may be affected when changing the code.

Caller ID Call Logging

Caller ID Call Logging and Dialing (F23) 8 Once the system administrator assigns the Caller ID Call Log Line Association, Caller ID Log  Answered Calls, and/or the Caller ID Log All Calls features to log Caller ID calls, you use Caller  ID Logging and Dialing (F23) to view the log. Caller ID Call Logging and Dialing is available on  system display telephones for all lines for which you subscribe to a Caller ID service. Use this  feature to view Caller ID information for central office calls.
You must program Caller ID Logging and Dialing onto a line button with LEDs to use
the feature.
Up to 400 call records can be stored for the system. Each line associated with an extension to log
Caller ID calls is guaranteed a minimum of 20 call records.
You also can automatically dial the number stored in the log.
The call records stored in each user’s call log and available for viewing depend on the following:
■
■
■
■
■
Unanswered transferred calls are logged automatically, whether or not the line and extension  are associated with the Call Logging features.
If Caller ID Log Answered Calls is used alone, all Caller ID calls that are answered by that  extension are logged.
If Caller ID Log Line Association is used alone, all unanswered Caller ID calls that ring on a  line associated with the extension are logged.
If both Caller ID Log Answered Calls and Caller ID Log Line Association are used, all
Caller ID calls that are answered by that extension and all unanswered Caller ID calls that  ring on a line associated with the extension are logged.
If both Caller ID Log Line Association and Caller ID Log All Calls are used, all answered
Caller ID calls and all unanswered Caller ID calls received at any extension on specific lines  are logged. This combination can be assigned to a maximum of one extension per system. The Caller ID information appears on three screens:
■ The first screen shows the caller’s number (or the reason that the number is not available).
■ The second screen shows the caller’s name (or the reason that the name is not available).
■ The third screen shows the date and time of the call.  In addition, the system logs the line the call came in on, whether the log entry was viewed, whether  the call was answered or not answered, and whether an attempt was made to return the call by  using the dialing option.

When battery power is getting low,

Battery Replacement 1 1 The PARTNER ACS processor  module uses  two user-replaceable  AAA alkaline  batteries.  These batteries  provide enough power  to  retain  the system programming  settings  during  a power  failure for  45  days  to  six  months, depending  on  the  freshness  of  the  batteries. When  battery  power  is getting  low,  the system displays  a  ChgBat  W/PowerOn  or  ReplaceSysBat  W/Power  On message  on  the  top line  of  display  telephones  at extensions  10  and  11  in  place of  the  default day/ date/time  message.  Users at  these  extensions  should  be instructed  to notify  you  when  they  see this  message.  You should  replace the batteries  within  45  days  of seeing  the  message.

General Mailbox Owner

Programming the General Mailbox Owners
Touch-Tone Input
777
0#
[nnnnnn] #
9
1
3
[nn] #
PARTNER Messaging provides four General Mailboxes–one for each  Automated Attendant. The General Mailbox extensions are 9991, 9992, 9993,  and 9994. You cannot delete these extensions or change their phone status.  Automated Attendant Service calls are directed to the Automated Attendant’s  General Mailbox when the Automated Attendant’s Dial 0/Timeout Action is set to  record a message in the General Mailbox and: • Caller does not make a selection from an Automated Attendant Service  Menu.
• Caller presses 0 while in Automated Attendant Service.
• Caller presses 0 while using the Directory to transfer.
The General Mailbox Owner is the extension whose Message Waiting Light is  turned on whenever a message is placed in the General Mailbox. The factory  setting for the General Mailbox Owner is extension 10. You can program a  different General Mailbox Owner for each General Mailbox, or you can have the  same owner for all the General Mailboxes.
In Appendix B, refer to PARTNER Messaging Planning Form 1 — System  Parameters for the General Mailbox Owners.

Conference, Voice Call/Privacy Release

• Conference, Remote
• Conference, Voice Call/Privacy
 Release
• Continued Dialing
• Data Line Security
• Delayed Ringing
• Department Calling
• Department Step Calling
• Dial Pad Confirmation Tone
• Dial Tone Detection
• Dialing Number Preview
• Digital Trunk Clocking
• Direct Inward Dialing (DID)
• Direct Inward Line (DIL)
• Direct Inward System Access (DISA)
• Direct Station Selection (DSS)
 Console
• Directed Call Pickup
• Directory Dialing
• Distinctive Ringing, Tones and
 Flash Patterns
• Do Not Disturb (DND)
• Door Box
• Drop Key
• Ecologically Sound Power Saving
 ModeE911 Compatibility
• Flash
• Flexible System Numbering
• Flexible Timeouts
• Forced Trunk Disconnect
• Group Call Pickup
• Group Listen
• Handset Mute/Handset Cutoff
• Hands-free and Monitor
• Hands-free Answerback/Forced
 Intercom Ringing
• Headset Operation
• Hold
• Hotel/Motel
• Hotel/Motel - Do Not Disturb
• Hotel/Motel - DSS Console
 Monitoring
• Hotel/Motel - Message Waiting
• Hotel/Motel - Room Status
• Hotel/Motel - Room Status Printout
• Hotel/Motel - Room-to-Room Call
 Restriction
• Hotel/Motel - Single Digit Dialing
• Hotel/Motel - Toll Restriction (When
 Checked In)
• Hotel/Motel - Wake Up Call
• Hot Key-Pad
• Hotline
• Howler Tone Service
• Illuminated Dial Pad
• InMail
• InMail-Automatic Access to VM
 by Caller ID
• InMail-Cascade Message
 Notification
• InMail-Email Notification
• InMail-Find-Me Follow-Me
• InMail - Language Setting
• InMail Park and Page
• InMail Upload Download Audio
• Intercom
• IP Multiline Station (SIP)
• IP Single Line Telephone (SIP)
• IP Trunk - (SIP) Session Initiation
 Protocol
• ISDN Compatibility
• Last Number Redial
• Licensing
• Line Preference
• Long Conversation Cutoff
• Loop Keys
• Maintenance
• Meet Me Conference
• Meet Me Paging
• Meet Me Paging Transfer
• Memo Dial
• Message Waiting
• Microphone Cutoff
• Mobile Extension
• Mobile Extension - Callback to
 Mobile Phone
• Multiple Trunk Types
• Music on Hold
• Name Storing
• Navigation Key
• Night Service

Multi-line terminal

A Multiline Terminal user can use a customer-provided headset in place of the handset. Like using
Handsfree, using the headset frees up the user’s hands for other work. However, Headset Operation
provides privacy not available from Handsfree.
As the headset plugs into a separate jack on the bottom of the telephone, the handset can still be
connected to the telephone. This gives you the option to use the handset, headset or the
speakerphone for calls.

Forwarding with Follow Me

Call Forwarding with Follow Me
Description
While at a co-worker’s desk, a user can have Call Forwarding with Follow Me redirect their calls to the co-worker’s extension. This helps an employee who gets detained at a co-worker’s desk longer than expected. To prevent losing important calls, the employee can activate Call Forwarding with Follow Me from the co-worker’s telephone.
Call Forwarding with Follow Me reroutes calls from the destination extension. To reroute calls from the initiating (forwarding) extension, use Call Forwarding.
Conditions
•Call Forwarding an extension in a Department Group prevents that extension from receiving Department Pilot Calls.
•Multiple Stations can set Call Forward Follow Me to one station.
•Calls to extensions with DND active do not follow Call Forwarding programming. DIL calls ring an idle Department Group member, and then follow PRG 22-08 programming then PRG 22-05

Mute the Microphone;

An iPECS Phone can turn off audio transmission from the handset, speakerphone or headset microphone, “Mic Mute”.
Operation iPECS Phone To Mute the Microphone;
1. Press the [MUTE] button, the [MUTE] button LED is on and the microphone (Handset, Speakerphone, Headset) is muted; the connected party receives silence.
To activate the microphone;
1. Press the illuminated [MUTE] button, the [MUTE] button LED is off, and the microphone is activated, transmitting audio to the connected party.
Conditions
1. Changing from speakerphone to handset or vice versa during a mute condition will eliminate the mute status.
2. Returning to idle or placing another CO/IP or intercom call will change the mute status to its normal (active microphone) condition.

Restore the system voice prompts

Determine the version of system voice prompts currently installed on the InMail CompactFlash card. Refer to Selecting the CompactFlash Card on page
• Back up the system voice prompts and recorded names, messages and greetings stored on the In- Mail CompactFlash card to your PC hard disk. Refer to Backing Up the InMail CompactFlash Card on page
• Restore the system voice prompts and recorded names, messages, and greetings stored on the your PC hard disk to the InMail CompactFlash card. Refer to Restoring a Backed-up Database to the In- Mail CompactFlash Card on page
• Update the system voice prompts stored on the InMail CompactFlash Card (required for feature up- grades). Refer to Initializing the InMail CompactFlash Card on page
• Optionally, initialize (completely erase) the InMail CompactFlash card and load the latest system voice prompts. Refer to Initializing the InMail CompactFlash Card on page
• Optionally add or remove any of the supported language prompt sets.

Door open

DOOR OPEN
Description
The iPECS hardware is equipped with relays that activate External Control Contacts. The contacts can be assigned to one of several functions including a Door Open Contact. When used as a Door Open Contact, the contact is connected to a door-lock release mechanism. When assigned stations receive the Door Bell signal, the user may dial the Door Open code to activate the contact.

IX-EXPML

The IX-EXPML2 Expansion Module is used exclusively when adding a fourth shelf to the system. The IX-EXPML1 Expansion module must be used when adding a second, third, or fifth shelf to the system.
The IX-EXPML2 expansion module adds six universal card slots and 96 universal ports to the ADIX APS system. In IX-EXPML2, the first card slot supports a maximum of 24 ports. The second card slot supports a maximum of 8 ports. The third through sixth card slots each support a maximum of 16 ports. This unit has dedicated space for the IX-PWSL main power supply as well as the IX-DCDCM, IX-RNGUM power supplies. An internally-mounted IX-PWSL power supply is required in the IX-EXPML2 expansion module.

Dial mode

Dial Mode (#201) 4
Use this feature to identify individual lines as touch-tone or rotary. Check with your local telephone company if you are not sure which type of line is being provided to you.
Considerations 4
■ If you are having difficulty using touch-tone telephones on rotary lines, you may need to adjust the Rotary Dialing Timeout (#108).
■ If the system has rotary lines, you can use Touch-Tone Enable (F08) to send touch-tone signals over a rotary line (for example, to access bank-by-telephone services).
■ If Dial Mode is set to Rotary, star codes are entered by dialing 11 instead of *. If you have users at extensions with Outgoing Call Restriction (#401) set to Local Only who are calling out on lines with the Dial Mode set to Rotary, you should add “11” to an Allowed Phone Numbers List (#407) and assign the list to these extensions. Otherwise, when the users at the restricted extensions dial 1 to begin a star code, the system interprets this as an attempt to place a long-distance call, the call is blocked, and the user hears the reorder tone.
■ The 1600 DSL module supports only touch-tone signaling.
Programming

Department Calling group

If you and your co-workers handle
each other's calls, you might want to be in a Department Calling group . Someone calling your group's number goes through to any- one who's available. You can even have Department Step Calling () send your personal calls to your group when you're not avail- able. To answer a call already ring- ing a co-worker's phone, use Group Call Pickup ().
When you're on a call and you want the others in your area to listen in on the conversation, activate Group Listen (). Your co-work- ers hear the call through your telephone's speaker.
If you frequently call the same co-worker, you can have Ringdown  automatically call them for you. All you have to do is lift your handset.

Door Phones

Door Phone(s)
Door phones can be used to call phones selected in system programming. When a door phone calls, you hear a distinctive ringing tone, one or five times (set in system programming). You can also call a door phone and monitor the surrounding area.
LCD telephones display the door phone name ID when calls are made to or from door phones.
The number of possible door phones varies by Strata CIX system, 01~24 maximum for larger systems. Check with your System Administrator to find out the names and locations of your system’s door phones and record them below.

Optional Account Codes

Optional (Unforced) Account Codes
Optional Account Codes allow a keyset extension user to enter an Account Code while placing a trunk call or any time while on a call. This type of Account Code is optional: the system does not require the user to enter it. If the keyset user is already talking on a trunk call, their conversation continues uninterrupted while they enter an Account Code.
Single line telephone users can only enter an Account Code while placing their trunk call. Forced Account Codes
Forced Account Codes require an extension user to enter an Account Code every time they place a trunk call. If the user doesn’t enter the code, the system prevents the call. The system can require Forced Account Codes for all trunk calls, or just for toll calls (as determined by Toll Restriction programming). Note that Forced Account Codes do not pertain to incoming calls.
Verified Account Codes
With Verified Account Codes, the system compares the Account Code the user dials with a list of codes programmed into the Verified Account Code Table. If the Account Code is in the table, the call goes through (provided it is not prevented by an extension’s Toll Restriction programming). If the code is not in the table, the system prevents the call. Verified Account Codes, if enabled, apply only to Forced Account Codes.
Using Account Codes and Speed Dial
To simplify Account Code operation, Personal and System Speed Dial bins can contain Account Codes. Keep the following in mind when using Speed Dial and Account Codes:
● The Account Code can be either the first or last entry in the bin, and must be preceded and fol- lowed by the # character. For example, the Account Code 1234 must be entered as #1234#.
● The Program 0201 - # Key to Enter Account Codes (page 629) option must be enabled in sys- tem programming. In addition, the Program 0201 - Enable Account Codes in Speed Dial (page 629) option must also be enabled.
● The Speed Dial bin can contain an Account Code followed by an outside number, or just the Account Code. The Account Code must be preceded and followed by a # entry. If the bin contains just the Account Code, the user must be sure to press the bin key before dialing the outside number.
● If the system has Verified Account Codes enabled, the Account Code entered in the Speed Dial bin must match an entry in the Verified Account Code Table.

Converged communication

iPECS SBG-1000 is a truly converged business communications services platform in a single appliance

IP-telephony platform, SIP and DECT, a broad range of terminals, and advanced telecom applications
Dynamic embedded routing protocols, integrated 802.11b/g/n WiFi, QoS and built-in NAT
Built-in VPN, enhanced security protocols, firewall, DMZ and rules-based packet filtering
Embedded high-speed USB 2.0 port for IT services, connection to local servers for file, fax and printer sharing
Soft client makes smartphones an extension of the communications platform, 3G / 4G mobile broadband plug-in option
Advanced Qsig and IP networking, local and remote management and interoperability with TR-069 and SNMP

Basic voice calling

iPECS UCS Client: Basic voice calling plus advanced functionality such as video, IM, file sharing, audio conferencing, visual voice mail
• Multi-Tier Mobility: DECT and Wi-Fi phones for in-office cordless, mobile extensions to extend voice features to any phone and iPECS Communicator for full smartphone integration
• Integrated Auto Attendant / Voice Mail (AA/VM): Multi-level, multi-language AA, built- in Voice Store and Forward (VSF) Gateway
• Voice Mail Notifications: Email with or without .wav file attachment or as an alert to mobile phone
• IP Attendant: Windows-based application with powerful monitoring, Busy Lamp Field (BLF) status and attendant console features

• Automatic Call Distribution (ACD): Flexible incoming call routing, real-time agent monitoring and supervision and call record statistics
• Centrally-Controlled T-Net: Greatly enhances survivability, server and geographic redundancy and management of local and remote sites
• System Geographic Redundancy: Hot standby call server for seamless hand-over in case of main server failure
• Multi-Language: Simultaneous support for up to 16 languages in all features
• Power Backup: Power redundancy features wit

Premium Features

iPECS UCP – Premium UC Features

› Conference call and Collaboration ̶ Max six members in video conference ̶ File transfer among colleagues
̶ Application sharing for simultaneous access and editing of any type document in real time
̶ Share desktop with other UCS users
̶ Share web page address with other UCS
users
̶ Whiteboard to share drawings and free-form
text in real-time
› Microsoft Exchange Server Integration
– More precise schedule synchronization with Exchange Server

PBX or Centrex Services

Changing Settings to Support
PBX or Centrex Services 3
Your system may work behind a PBX or Centrex system:
■ PBX services are provided by a private telephone switch.
■ Centrex services are provided by your local telephone company from a Central Office (CO) outside your premises. These services include the Centrex lines connected to your control unit modules and some set of features—such as hold, conference, or transfer—available on those lines. Centrex services may be offered in your area under a different name. For specific Centrex features to be available to you, your company must subscribe to those features. For specific information about using Centrex features, see the Centrex documentation provided by your local telephone company.

Pool Access

Pool Access Restriction (#315) 4
Use this feature to restrict a pooled extension from receiving and/or making outside calls on all lines belonging to a specific pool. For example, you may want customer service representatives to make calls using the WATS lines that belong to auxiliary pool 881, but not receive incoming calls on the lines in that pool; in this case, you assign pool 881 to the customer service representative’s telephones and restrict the pool to Out Only.
This procedure is the most extreme way to restrict dialing. For example, an extension set to In Only or No Access for a particular pool cannot select that pool to dial out—even for numbers on the Emergency Phone Number

Data Call Button

Data Call Button
A flexible button on a digital telephone can be assigned as a 'DWD&DOO button, which can be used to dial internal data calls. The telephone must be equipped with an RPCI-DI for 'DWD&DOO button applications.
Direct Station Selection (DSS) Buttons
Digital telephone users can ring selected stations by pressing a flexible feature button assigned for a DSS function. The LED associated with the button provides the busy status of the station and the station’s [PDN]. Each flexible button can be assigned as a DSS button to a different station [PDN]. DSS buttons can also be assigned on DADMs.
Direct Station Selection (DSS) Console Features
On DK40i and DK424, digital and electronic telephones can operate with DSS consoles, which offer the following features:
o Automatic CO line Hold

Change language

change the language in which display messages appear if the extension has a system display telephone. The language is set for each extension, so telephones in the same system can display different languages.
Considerations 4
If SMDR is used, the call report header is printed in the language specified for extension

Direct Inward Dialing (DID)

T1 module is required for Direct Inward Dialing (DID).
■ You must use the PARTNER ACS R7.0 PC Administration software to program Direct Inward Dialing (DID) on T1 lines.
■ You can have a call on a T1 line with Direct Inward Dialing (DID) ring at an extension that cannot access a T1 line from a line or pool button. If the line or pool button is not programmed at an extension, or the line or pool button is busy, the call rings at one of the extension’s Intercom buttons.
■ Outgoing calls can be made on T1 lines administered for Direct Inward Dialing (DID).
■ Automatic System Answer features are disabled on T1 lines with Direct Inward Dialing (DID).
■ Direct Extension Dialing features are disabled on T1 lines with Direct Inward Dialing (DID).
■ Caller ID information is unavailable on T1 lines. The message “Direct In Dial” or “DID” is displayed for incoming calls on T1 lines with Direct Inward Dialing (DID).
■ You should not assign T1 lines with Direct Inward Dialing (DID) to Hunt Groups. If a Hunt Group contains T1 lines with Direct Inward Dialing (DID), callers will receive busy signal when all extensions in the Hunt Group are busy (for example, off-hook), have Do Not Disturb activated, or are in programming mode.

Incoming call routing

Stations can be grouped for incoming call routing and Call Pick-up purposes. Ten types of groups can be defined:
 Circular  Terminal  ACD  Ring  Pick-Up  External Voice Mail  VMIM/VSF-Voice Mail  Feature Server UMS Group  Net VM (Centralized External VM)  Unified Communication Solution Server Circular Station Group
In Circular Hunt, calls to a station in the group will go to the station, if unavailable or unanswered in the hunt no answer time; the call will be directed to the next station defined in the group. The call will continue to hunt until each station in the group has been tried. The call remains at the last station or passes to a designated overflow station or group.
A Circular Station Group can be assigned with a pilot number (the Station Group Number) so that calls to the pilot number will hunt. In this case, the call will be directed to the first station in the group and, if needed, hunt through each station in the group until reaching the last station. The call may remain at the last station, passed to an overflow destination or sent to a voice mailbox.
Terminal Station Group
Calls to a station in a Terminal Station Group that encounter an unavailable or unanswered status will be routed through the hunt process. The call will proceed to the next listed station in the group until reaching the last listed station in the group. The call may remain at the last station or be routed to an Overflow destination.
A Terminal Hunt Group can be assigned with a pilot number (the Station Group number) so that calls to the pilot number will hunt. In this case, the call will route as described for Circular Pilot Number hunting.
ACD Station Group
Calls can be sent to an ACD group by dialing the Station Group Number or assigning CO/IP lines to ring directly to the Station Group. Calls are directed to the station in the group that has been idle for the longest continuous time, Uniform Call Distribution. If all stations are busy or unavailable when the call is received, the call may be routed to an alternate location or may continue to wait (queue) for an available station in the group. After queuing to the group, the caller may be routed to an overflow destination, which can be a Station, Station Group or Voice Mailbox.
An ACD supervisor can be assigned to monitor the group and act to oversee operations of the group. The ACD Supervisor can print group statistics and activate alternate routing as well as assist agents.

External Hotline

External Hotline (#311) 4
Use this feature to identify an external hotline extension. When a user lifts the handset of an external hotline, a predetermined outside number is dialed automatically. The external number might be, for example, a frequently called service bureau. The external hotline must be a single- line telephone, not a system telephone, and should not have a dialpad.
After you identify an external hotline extension, you must store the external telephone number for the hotline extension as Personal Speed Dial code 80.
Considerations 4
■ Under certain conditions of heavy telephone usage, the external hotline may be unable to dial the programmed number immediately (for example, if many of the tip/ring devices connected to your system dial out at the same time).
■ You can identify several extensions as external hotlines.
■ Do not assign an external hotline to extension 10, 11, or to the first two extensions of any 206 or 308EC module, which are reserved as power-failure extensions.
■ Use Hotline (#603) to identify an internal hotline extension.
■ Use Line Assignment (#301) to assign outside lines to the external hotline extension, Pool Extension Assignment (#314) to assign pools to the external hotline extension, and Automatic Line Selection to set the extension to select outside lines or pools first.
If your use of the external hotline requires immediate dialing of the programmed number, assign a line for use only by this extension.
■ Set Line Ringing to No Ring for all lines or pools assigned to the external hotline to prevent incoming calls from ringing at the extension.
■ Make sure there are no call, line, or pool restrictions assigned to an external hotline.
■ Do not use Station Lock at an external hotline because it will prevent the outside number from being dialed.
■ Remove external hotlines from Night Service Group Extensions (#504), Calling Group Extensions (#502), and Hunt Group Extensions (#505).
■ Do not assign Forced Account Code Entry (#307) to the external hotline.

Licenses

CIX40 System Licenses
The analog CO lines, digital telephones, standard telephones, four voice mail ports, DTMF circuits, attendant consoles, ACD VA ports and BSIS serial ports do not require a license for operation on the CIX40. IP Ports on the optional GIPH and MIPU cards, and more than four (up to eight) voice mail require licensing on the GCTU2A processor as follows:
• Each Toshiba IPT2000 anf IP5000 series telephone and SIP Telephone requires one LIC-CIX- IP_Port license.
• Each Strata Net IP channel requires a LIC-CIX-STRN-CH license.
• Toshiba SoftIPTs require a LIC-SOFTIPT license and one LIC-CIX-IP_Port license for each SoftIPT.
• Two additional voice mail ports are provided by each LIC-2 GVPH license, up to the system maximum of eight.
• Licenses are required on the Media Application Server (MAS), Stratagy® SES, and customer provided PCs for applications such as ACD, Net Phone, TASKE®, Oaisys®, Feature Flex®,

Key mode

Hybrid Mode 1
Hybrid mode offers you flexibility in accessing outside lines from your telephone. As in Key mode, individual lines can be assigned to system extensions. Additionally, multiple outside lines can be grouped together in pools. The system can have up to four pools, including a main pool and three auxiliary pools. Each pool is identified by a pool access code—880, 881, 882, and 883 respectively.
Pools are represented on system telephones by pool buttons. Unlike line buttons, pool buttons give you access to multiple lines from a single button. Each auxiliary pool is associated with only one pool button. Since the main pool typically contains most of your company’s outside lines, it is associated with two pool buttons. This setup allows you to place a call by using one of the main pool buttons, place that call on hold, and make another call by using the second main pool button. Or you can establish a conference call by using lines in the main pool. The main pool and each auxiliary pool can be assigned to an extension, for a maximum of five pool buttons per extension.
System telephone users can press any of the available pool buttons on their telephones, or they can enter the pool access code at intercom dial tone to make an outside call. (Single-line telephone users must dial 9 or enter the pool access code at intercom dial tone to access a pool since their telephones do not have pool buttons.) After you press a pool button or enter a pool access code, the system selects a free line from the pool for you to make the call. You can access a pool as long as there is at least one available line in the pool.
Hybrid mode allows users who have system telephones with fewer buttons to have access to multiple outside lines and various types of pools. You can make efficient use of outside lines by grouping those of a similar type or function together. For example, you can create an auxiliary pool of WATS or international lines and assign the pools to different groups of users. Additionally, individual lines can be assigned to a manager’s extension so that he or she always has access to an outside line.

Assigning lines

Assigning Lines 3
Use the feature Number of Lines (#104) to determine in Key mode the number of outside lines that are automatically assigned to all system extensions, or to determine in Hybrid mode the number of lines assigned to the main pool. After you use the Number of Lines feature, you can use other features to refine the assignment of lines.
Line assignment is also based upon whether an extension is a Key or Pooled extension. Refine the line assignment accordingly after you use the Number of Lines feature.
Number of Lines (#104) 3
Use this feature to determine in Key mode the number of outside lines that are automatically assigned to all system extensions. In Hybrid mode, use this feature to determine the number of lines assigned to the main pool. (This number does not have to match the actual number of lines connected at the control unit.) This feature should be used only for initial system setup.

Main KSU

To avoid shock or equipment damage, do not plug in or turn on the system power before completing the hardware installation. The Main KSU has the built in CPU, eight digital station ports, four analog extension ports, and four analog trunk ports. Each KSU can have up to two expansion cards installed and the expansion cards are:
• IP4WW-080E-B1 — 8 port digital station card • IP4WW-PRIU-C1-PRI/T1 Trunk Card
• IP4WW-008E-B1 — 8 port analog station card Note: The PRIU card can be installed in the 3rd expansion slot of KSU 1 and KSU 2. No other cards can be installed in these slots.
• IP4WW-4COIDB-B1 — 4 port analog trunk card Note: The 4 port analog trunk card is a daughter board that can be mounted on the IPWW-080E-B1, IPWW-008E-B1 and the KSU motherboard.
• IP4WW-VOIPDB-C1 — Required for Voice Over
• IP4WW-CFVMS-C1 — Compact Flash used for • IP4WW-CFVML-C1 — Compact flash Voice Mail and VRS functions. Provides used for Voice Mail and VRS functions. approximately 15 hours of recordings.

Fractional PRI

Working with fractional PRI Installer password required
When you purchase PRI from your service provider, you can request the number of B-channels that are allocated for you to use. For example, you may want to use only 12 B-channels instead of 23 B-channels. If this is your situation, you should disable all the B-channels that you do not need.
It is recommended that the number of lines that are deprovisioned on a DTI card (configured as PRI) be the same as the number of B-channels that are disabled. For example when B-channels 13-23 are disabled, you should deprovision

minimum of four IP (SIP) trunks

1.4 Requirements
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 American Telephone Company.
1.5 Limitations
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
 T.38 Faxing is not supported by American Telephone Company.
 G.711 is the only CODEC supported with American Telephone Company.
 The default CODEC Frame Size of 20ms must remain in command

Console

Attendant Console
Attendant Interposition Calling/Transfer
Attendant Interposition Calling/Transfer
General Description
This feature allows any Attendant to directly converse with another Attendant and also allows Attendants to transfer calls from their console to another Attendant's console in systems where Multiple Console Operation has been provided.
Operating Procedure
To call from Console A to Console B
1. Attendant A presses an idle loop key.
2. Attendant A dials the Interposition Calling/Transfer access code and Attendant B's identification number.
3. The call is indicated at console B (on the ANS key or TF key).
4. Attendant B presses the ANS key or TF key.
5. Attendant A converses with Attendant B.
6. Attendant A and B press the RLS key.
To transfer from Console A to Console B with a call in progress
1. Attendant A dials the Interposition Calling/Transfer access code and Attendant B's identification number.
2. The call is indicated at Console B (on the ANS key or TF key).
3. Attendant B presses the ANS key.
4. Attendant A presses the RLS key to transfer, or may consult before release.
Service Conditions
1. Each console is assigned an identification number to allow Interposition Calling or Transfers.
2. An Attendant can receive one Interposition Call or Transfer at a time.
3. After receiving an Interposition Transfer, the Attendant has full capabilities for redirecting the call.
4. When Night Service is in effect at the called console, Interposition Calling and Transfers result in reorder tone.

Two-way Recording

Two-way Recording
Description
This feature allows a user to record a CO line call to the station’s mailbox. Two-way Recording can be enabled only for DKT stations and the station must have a Flexible button assigned as a {Record} button or, for the 6200 series, the Record softkey may be used. When a call is recorded, the system activates a Message Wait to the station.
Operation
To assign a {Record} Flexible button at the DKT,
1. Press the [TRANS/PGM] button.
2. Press the desired Flexible button.
3. Press the [TRANS/PGM] button.
4. Dial ‘5 4’.
5. Press the [HOLD/SAVE] button.
To record a conversation while on a CO line call,
1. Press the {Record} button or softkey.
2. The conversation is record
To stop recording,
1. Press the {Record} or Flash button or hang-up.
2. Conversation recording stops and the message is saved.
Consideration
 Users should announce the recording, no recording tone is provided.
 During a recording, the {RECORD} button flashes to indicate a recording in progress.
 If the external VM requires DTMF for control signaling or there is no external Voice mail, the conversation is recorded to the user’s VMIB Mailbox.
Reference
A. Station User Programming
B. External Voice Mail
C. Station Hunt Groups
D. VMIB Voice Mail
Admin Programming
Two Way Recording (PGM 112, Flexible button 10) Enables a station to use Two-way Recording

Toll restrictions

3501-01:Toll Restriction Country Type See the Tool Bar
Use this option to set the Toll Restriction country type. You should select US Domestic (1) for systems installed in North America.
Features Related to this Program Telephone Features
• Toll Restriction IntraMail Features
• None
Program Options
Description Option US Domestic (North American) Toll Restriction [Default] 1 Latin American Toll Restriction 2
3501-02: Emergency Number See the Tool Bar
This option is xed at 911 for systems using US Domestic (North American) Toll Restriction. You cannot change this setting.
3511: Settings
3511-01: Active Key Pad [Lines: Toll Restriction: Options: Settings (3511): Active Dial Pad]
Use this option to enable or disable Active Key Pad (continuous dialing) for outside calls.
Features Related to this Program Telephone Features
• Toll Restriction IntraMail Features
• None

DN button to a line button

enabled, line calls move from a telephone DN button to a line button after they are answered. After the call is answered, the DN button is cleared to receive another call. With this operation the DN acts as an answer button for the phone. This only applies if the line answered has a CO, GCO, or Pooled line button appearance on the phone. If disabled, line calls remain on the DN after they are answered. 106 Day/Night Mode Day of Week Mapping 106, Hold The Day of the Week schedule defines each day as the type of day the schedule shall follow. These types of days are called Work Day, Non-work Day, and Holiday. Each day of the week can be classified. 106-00 Tenant Number n = Enter 1~8 Select the Tenant number for which the daily schedules will be configured. 106-01 Monday n = 1. Work Day (default) 2. Non-Work 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-02 Tuesday n = 1. Work Day (default) 2. Non-Work 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-03 Wednesday n = 1. Work Day (default) 2. Non-Work 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-04 Thursday n = 1. Work Day (default) 2. Non-Work 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-05 Friday n = 1. Work Day (default) 2. Non-Work 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-06 Saturday n = 1. Work Day 2. Non-Work (default) 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday. 106-07 Sunday n = 1. Work Day 2. Non-Work (default) 3. Holiday Enter the type of day to follow for daily schedule. 1- Work Day; 2- Non-Work Day; or 3- Holiday.

Hold Reminder

Hold Reminder Tone (#127) 4
Use this feature to set the time interval at which the PARNTER system will provide a reminder tone for a call on hold at a system telephone. When you place a call on hold, the PARTNER system will provide the Hold Reminder Tone at your extension if the call remains on hold for the time interval specified. You can disable the Hold Reminder Tone. (If you disable the Hold Reminder Tone, the PARTNER system will not provide a reminder tone when a call is on hold.)
Considerations 4
■ When you program this feature via Set administration, the setting you specify will apply to all analog lines. It will not apply to T1 lines. You must use the PARTNER ACS R7.0 PC Administration software to program the Hold Reminder Tone for T1 lines

Modifying Mailbox Extensions

Modifying Mailbox Extensions
This section provides procedures to:
• Change a Directory Listing or Recorded Name.
• Change the Maximum Message Length for messages left in a mailbox or for messages created.
• Change the Mailbox Size.
• Reinitialize the Mailbox Password.
• Reinitialize a Mailbox.
• Change Outcalling Permission.
• Change the Mailbox Language.
• Change the Record-a-Call Message Length.
• Change the Phone Status.
• Delete a Mailbox Extension.

Problems and Corrective Actions

PROBLEM: System terminates recording of a name, greeting, message, or an Automated Attendant Main Menu, Submenu or Announcement before the user finishes.
Possible Cause Corrective Action Recording may have reached
the time limit set for it. Be sure the recording does not
Speaker’s voice sounded like
exceed the time limit, then re-record it.
a Touch-Tone digit. Re-record the message.
PROBLEM: Caller hears a busy signal when calling into Voice Mail Service. Possible Cause Corrective Action
All ports are in use or maintenance is being performed on the PARTNER Messaging module.
There are no ports assigned
Try again later. If the condition persists, call the helpline at 1 800 628-2888 or your authorized dealer.
to Hunt Group 7. Assign the ports to Hunt Group 7
(programming procedure #505).
PROBLEM: The first part of a Menu, Submenu, Announcement, or Personal Greeting is cut off.
Possible Cause Corrective Action You may have begun
recording too early. Re-record, speaking after the

VMEX

SIP VIRTUAL MOBILE EXTENSION (VMEX)
Description
SIP gives us the opportunity to provide a truly integrated and cost effective mobile extension solution. The closer integration lets the mobile users to use the mobile exactly as they are used to.
The goal have been to integrate the mobile phone even further with the PBX (out-band signaling) and add a more cost effective solution for bigger systems (recursing diversion). The out-band signaling is mandatory. Recursing diversion is not required in smaller solutions, but still more cost effective. In a bigger solution, more then 20-30 MEX users, it is needed to compete with Aastra, Cisco and various hosted solutions, including the hosted solution we sell.
Out-band signaling makes it a truly transparent solution. The user places an outgoing call exactly as he is used to and it works on any cell phone. This means that the user can use speed dials and do not need to have any special phone or application installed in the phone. Also the use of DISA, which is considered too complicated by many users, will be obsolete.

Data Module

Data Module down
1. Run a Maintenance session to ensure that the module is not disabled. See Module status on page 537.
2. Disable the Data Module using Module status.
3. Enable the Data Module using Module status.
4. If the Data Module is still down, power down, then power up the ICS and the Data Module.
If the problem persists
1. If AC power is present and the LED indicator on the Data Module is off, replace the Data Module.
2. Replace the Fiber cable.

richly-featured hybrid IP/TDM communications platform

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• Full Featured: Full PBX feature set including auto attendant, voicemail, call transfer, call forward, call park, call pickup, speed dial, station group, 3-party voice conference and more
• Flexible: Ability to use Digital (DKT) and Analog (SLT) phones along with the full selection of iPECS IP phones and DECT phones. A wide range of mobile applications and soft clients are also supported.
• Secure: Supports IPSec and sRTP security protocols
• Affordable: Low TCO with simple and straightforward licensing structure
• Leading Technology: Advanced IP networking with local and remote management through an intuitive HTML5-based GUI

Gain control

Automatic Gain Control (AGC)
• Adjustable VOX sensitivity
• Compatible with Valcom Multi-Zone Control Units and Station Adaptors
• Two input channels: Switch selectable primary and priority, or two channels with equal priority; both inputs are switch selectable for Loop Start or VOX operation
• Two audio outputs:
600 Ohms with loop supervision to a page control unit or 8 Ohms line level output to amplified speakers
• Background music input
• Total of three normally open relay contacts for PLAY, RECORD and BUSY mode (one contact each)
• LED status indicators for POWER, RECORD, PLAY and BUSY
• Dip switch option programming
• Message abort feature
Access
• Vacant Loop Start Trunk Port or
C. O. Line Port
• Page Port
• Line Level Audio Feed
• An adaptor is required for C. O. Line or station access (see V-9970 or V-9940)

Exclusive Hold

Hold
1.To place a call on hold, press Hold. Your LCD shows the line on hold. The held Line’s LED flashes green while appearances of the line at other stations flash red.
• To return to the held call, press the held Line button.
• If you do not return to the held call within a specified time, it rings back to your telephone. The call remains camped-on to your station indefinitely.
• If the held party hangs up, the call is released.
2. Repeat Step 1 to place additional lines on hold.
Note See the Call Pickup section to pick up a call on hold from another extension.
Exclusive Hold
This feature enables you to place a call on hold so that only you can retrieve it.
³ While on a call, press Hold twice. That line’s LED flashes green while appearances of the line at other stations are steady red (in use).
Automatic Hold
This feature enables you to place a call on hold by pressing another Line button – no need to press Hold. Make sure this feature is programmed on your system; if not, existing calls will drop.
³ While on a call, press another extension button to receive/originate a new call. The accessed line’s LED flashes (in-use). The first call is put on hold and your extension’s LED flashes. You can also switch between calls without having to press Hold.
Consultation Hold
1. While on a call, press Cnf/Trn. The call goes on hold.
2. Dial another line.
3. Transfer the call or return to the held call by pressing its Line button.

System shutdown

System Shutdown
This command can be used to shutdown systems with IP Office Release 6 or higher software. The shut down can be either indefinite or for a set period of time after which the IP Office will reboot.
! WARNINGS
· A shutdown must always be used to switch off the system. Simply removing the power cord or switching off the
power input may cause errors.
· This is not a polite shutdown, any calls and services in operation will be stopped. Once shutdown, the system
cannot be used to make or receive calls until restarted.
· The shutdown process takes up to a minute to complete. When shutdown, the CPU LED and the IP500 base card LEDs 1 and 9 (if trunk daughter card fitted) will flash red rapidly. The memory card LEDs are extinguished. Do not remove power from the system or remove any of the memory cards until the system is in the this state.
· To restart a system when shutdown indefinitely, or to restart a system before the timed restart, switch power to the system off and on again.

Event messages

Event messages
Event messages appear as items in the System administration log or the System test log of the Maintenance session. Most of these event messages can only be caused by an unusual combination of events, and should rarely occur.
Each event is assigned a severity number. An S preceding this number, S4 for example, may appear in the event message. S9 is the most severe. If the log is full, new event messages with a higher severity number replace existing event messages of a lower severity. For this reason, you should check event messages at regular intervals. You can then deal with all messages before they are replaced.

Lock-key operation

Lock-key operation
Description
ipLDK-60 supports a number of software applications including Private Networking, ez- Attendant, ez-Phone, Phontage and TAPI. The system software includes 60-day trial license of the applications. In addition, the software includes the ability to support two Phontage IP softphones. To permanently add a demonstration application or expand the supported Phontage softphones, additional software must be added to the system based on the system serial number. Using the system serial number a key code is generated and, when entered over a telnet or PC Admin session, will add the application to the system software.

Lock-key operation

Lock-key operation
Description
ipLDK-60 supports a number of software applications including Private Networking, ez- Attendant, ez-Phone, Phontage and TAPI. The system software includes 60-day trial license of the applications. In addition, the software includes the ability to support two Phontage IP softphones. To permanently add a demonstration application or expand the supported Phontage softphones, additional software must be added to the system based on the system serial number. Using the system serial number a key code is generated and, when entered over a telnet or PC Admin session, will add the application to the system software.

IP Office U-Law

· IP Office U-Law
A system fitted with this type of card will default to U-Law telephony. For pre-IP Office Release 7.0 software, the system will default to IP Office standard mode. For IP Office Release 7.0+, the system will default to IP Office Essential Edition - Quick Version mode Key System operation. Intended for North American locales.
· IP Office A-Law
A system fitted with this type of card will default to A-Law telephony. For pre-IP Office Release 7.0 software, the system will default to IP Office standard mode. For IP Office Release 7.0+, the system will default to IP Office Essential Edition - Quick Version mode PBX System operation. Intended for locales outside North America.
· IP Office Partner Version
A system fitted with this type of card will default to U-Law telephony and IP Office Essential Edition - PARTNER® Version mode Key System operation. Supported only in North American locales.
· IP Office Norstar Version
A system fitted with this type of card will default to A-Law telephony and IP Office Essential Edition - Norstar Version mode Key System operation. Supported only in Middle East and North African locales.

DID Mapping table

DID Mapping Table
This table is used to set the destination for incoming calls that include DID digits. These are routed by matching the DID and ICLID information received with the call to an entry in the table. This overrides the Coverage Destination settings of the channel on which the call was received. Calls routing by DID mapping are not affected by the phone system being put into night service.
If the system is in Key system mode and no match is found, the call is routed to the first extension in the system. If the system is in PBX mode and no match is found, the call is routed to the Operator group.
The table is applied to all channels.
· DID Number
If the incoming DID of a call on the trunk matches the DID set here, it will be routed to this destination. The system supports up to 4 digits DID (additional digits after the first 4 are ignored). Leave blank if only CLI matching is required.
· Incoming CLI
If the incoming caller number on the trunk matches the Incoming CLI set here, it will be routed to this destination. Leave blank if only DID matching is required.
· Destination
When this field is selected, the drop down list allows selection of the destination for matching calls. The options
differ depending on whether the system's System Mode is set to Key System or PBX System.
· Extension
49
Route incoming calls to a particular extension.
· Phantom Extension
IP Office Release 6.1+ supports phantom extensions . One of these can be selected as the destination for
calls.
· Hunt Group
23
Incoming calls can be routed to one of the 6 sequential hunt groups .
· Calling Group
78
For systems with their System Mode set to PBX System, incoming calls can be routed to one of the 4
49 78
collective calling groups .
· Operator Group
For systems with their System Mode set to PBX System, incoming calls are routed to the Operator
78
Group .
· Voicemail
49
Route incoming calls to the systems voicemail to collect messages. This requires the caller to know the mailbox number and passcode.
· 76: Modem
For Release 6.1+, the option 76: Modem can be selected to route the call to the systems built in V32 modem function. This is intended for basic configuration access by system maintainers.
24
· Auto Attendant
For Release 6.1+, any of the configured voicemail auto attendants can be selected as the call destination.

OFF-HOOK SIGNAL

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.

LCR

The Least Cost Routing (LCR) feature allows the system to automatically select the least costly route available according to the number dialed, the time of day/day of week, the class of service (COS) assigned to the station/trunk group priority level assigned. When a station user dials an outside number, the LCR feature analyzes the number, then automatically chooses an outside line from the group that is programmed as most economical.
The LCR feature puts the responsibility of choosing the least expensive route for each area code and exchange code on the system administrator, not on the station user. In order to make a routing decision, the LCR feature is programmed in the system database. The successful operation of this feature is completely dependent on the accuracy of the programming. The STSe may be programmed on a per station basis to force the use of LCR for outgoing access. This allows the system administrator to maintain greater control over dialing patterns and the lines used for placing outgoing CO calls.
LCR Tables
There are eight (8) different tables which are set up to monitor the dialing of digits of a station and to select the best route programmed for the call.
3-Digit Routing Table -- allows the system to handle call routing in areas that require a 1 before a long distance number and in areas that do not require the 1 for 7-, 8-, 10-, & 11-digit dialing.
6-Digit Routing Table -- is used to determine the route for one or more office codes programmed within the same area code.
Exception Code Table -- is used to route operator assisted calls and any other calls which would use a one or two-digit number rather than a three-digit area code.
Route List Table -- contains four Time Period Routing Lists that are used in conjunction with other LCR table settings.
Insert/Delete Table -- is used to add/remove digits from a number before or after it is dialed.
Daily Start Time Table -- allows the user to match the Time Periods discount structure to the carrier’s rate schedule.
Weekly Schedule Table -- allows the user to assign the Time-of-Day and Day-of-Week to use the least costly route for a specific dialed number.
Toll Information Table -- allows common call routing for all toll information calls: 1-(xxx)555-1212, (xxx)555-1212, 1-555-1212, and 555-1212.
DEFAULT LCR Database -- provides basic routing for all local and long distance dialing.

iPECS-UCP

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– Provides capacity for up to 2,200 devices, allowing it to handle most any need

iPECS-UCP
Anywhere, Anytime Connectivity
› Access the power of your iPECS call servers your way regardless of your device or location using smartphone, tablet or PC applications
Integrated Applications Tailored to your Needs › iPECS offers a range of enhanced applications from Ericsson-LG and other specialist application providers, including Microsoft Outlook or Lync as well as others
Reliable and Resilient
› Total reliability is the only option for your communications. With inherent modular architecture, iPECS UCP provides geographic redundancy, hot standby power redundancy and Central Control T-Net

eMG80

iPECS eMG80
The iPECS eMG80 is a richly-featured hybrid IP/TDM communications platform for voice and mobility services, optimized for small and growing businesses. With its modular and flexible design, businesses can easily and affordably expand into premium UC and more sophisticated enterprise applications.
• Full Featured: Full PBX feature set including auto attendant, voicemail, call transfer, call forward, call park, call pickup, speed dial, station group, 3-party voice conference and more
• Flexible: Ability to use Digital (DKT) and Analog (SLT) phones along with the full selection of iPECS IP phones and DECT phones. A wide range of mobile applications and soft clients are also supported.
• Secure: Supports IPSec and sRTP security protocols
• Affordable: Low TCO with simple and straightforward licensing structure
• Leading Technology: Advanced IP networking with local and remote management through an intuitive HTML5-based GUI

Diagnostic Interface Module

The SV8100 has a Diagnostic Interface Module (DIM) built into the CD-CP00-US (CPU) blade. The DIM can monitor the activity of the system under the control of commands entered by the engineer. The DIM is accessed via ia the Ethernet interface of the CD-CP00-US blade.
SECTION 2 BEFORE YOU START
As well as monitoring the system, the DIM can also be used to change the operation of the system.
For this reason DO NOT enter the following commands, as they will cause a system restart:
R RESET R RESTART R SHUTDOWN R or any other command that looks like a reset request.
Some DIM commands give a real time output when the command is entered, others will give an output until you enter the command that turns it off. You can turn on multiple DIM outputs by entering relevant commands one after the other.
The SV8100 will continue to operate normally with the DIM is running.

Installation, Cabling

Installation/Connection Cabling
Category 3 or 5 twisted pair cable is recommended for all Valcom distributed amplified paging installations. Screw terminals are provided for the basic connections. RJ45 jacks are provided for chaining multiple V-9964 units together. Removing the narrow right side panel of the V-9964 provides access to controls, connections and option switches. To remove the panel, loosen the two screws holding the panel in place and lift the panel.

VCM modules

For Small Community Networks VCM modules are required in all systems being connected. The IP lines should be configured to connect the IP Offices in a star configuration, however the data network itself can be meshed. Also the names and numbers (groups, line, services, etc) on the separate IP Office systems should be unique to reduce potential maintenance confusion.
Each IP Office system broadcasts UDP messages on Port 50795. These broadcasts typically recur every 30 seconds but BLF updates are potentially more frequent. There are no updates if there is no activity and the overall level of traffic is very low – typically less than 1 kbps per system.

Base station

Scanning the base station radio frequency
You must register the handset with a base station before you can perform this procedure. 1 Press the Func key to enter system programming mode.
The main menu appears.
2 Use the Up, Select, and Down softkeys to navigate to the Maintenance menu. 3 Press Select.
4 Log on to the Maintenance menu.
5 Use the up and down softkeys to navigate to the Scan Base RF option. 6 Press Select.
7 Use the up and down softkeys to navigate to the Yes option. The base station radio frequency status appears on the LCD.

Hold Reminder Tone

Hold Reminder Tone (#127) 4
Use this feature to set the time interval at which the PARNTER system will provide a reminder tone for a call on hold at a system telephone. When you place a call on hold, the PARTNER system will provide the Hold Reminder Tone at your extension if the call remains on hold for the time interval specified. You can disable the Hold Reminder Tone. (If you disable the Hold Reminder Tone, the PARTNER system will not provide a reminder tone when a call is on hold.)
Considerations 4
■ When you program this feature via Set administration, the setting you specify will apply to all analog lines. It will not apply to T1 lines. You must use the PARTNER ACS R7.0 PC Administration software to program the Hold Reminder Tone for T1 lines.

CIX40

CIX40 System Cabinet Versions
The CIX40 system is available on two hardware versions. The Release 1 cabinet is labeled CHSU40A. The Release 2 is labeled CHSU40A2. The system label is on the outside of the CIX40 cabinet on the right side. Throughout this manual R1 HW refers to the earlier CHSU40A systems and R2 HW refers to the later CHSU40A2 systems.
CIX40 Release 2 Cabinet System Capacities
The GCTU2 processor (also used in the CIX40 R1 cabinets) comes with a built-in maintenance modem (AMDS), one Music on Hold (MOH) interface, one IP interface (NIC), one External Paging interface, one relay contact, and one Secure Digital card slot.
The basic CIX40 R2 Hardware system is equipped with GCTU2 and GMAU3 supporting:
• Eight digital telephone ports
• Four CO line ports with Caller ID interface
• One standard single line (2500-set) telephone port.
The system can be expanded, using optional interface cards, to a total of up to 64 ports supporting the items shown i

RCC

Remote Call Control (RCC) for MS Lync iPECS RCC integrates call control in the MS Lync client for seamless voice calling, Lync presence and linkage to user’s desk phone.
iPECS ClickCall
By using iPECS ClickCall, simply drag and drop phone numbers from any Windows application or web page onto your PC or laptop to make outgoing phone calls from your desk phone.

UCS

iPECS UCS Client: Basic voice calling plus advanced functionality such as video, IM, file sharing, audio conferencing, visual voice mail
• Multi-Tier Mobility: DECT and Wi-Fi phones for in-office cordless, mobile extensions to extend voice features to any phone and iPECS Communicator for full smartphone integration
• Integrated Auto Attendant / Voice Mail (AA/VM): Multi-level, multi-language AA, built- in Voice Store and Forward (VSF) Gateway
• Voice Mail Notifications: Email with or without .wav file attachment or as an alert to mobile phone
• IP Attendant: Windows-based application with powerful monitoring, Busy Lamp Field (BLF) status and attendant console features

• Automatic Call Distribution (ACD): Flexible incoming call routing, real-time agent monitoring and supervision and call record statistics
• Centrally-Controlled T-Net: Greatly enhances survivability, server and geographic redundancy and management of local and remote sites
• System Geographic Redundancy: Hot standby call server for seamless hand-over in case of main server failure
• Multi-Language: Simultaneous support for up to 16 languages in all features
• Power Backup: Power redundancy features with main cabinet installation

Key System Mode

Key System Mode
Each phone is configured with 2 Intercom buttons which cannot be changed. It is also configured with line appearance buttons for specific lines using the Number of Lines settings and individual button programming.
Internal calls are made by selecting one of the two Intercom buttons provided on each phone and then dialing the number of another extension or of the system feature required.
External calls are made by selecting one of the line appearance buttons programmed on the phone and then dialing the external number required.
If the user dials without selecting an Intercom or Line button first, the user's automatic line selection setting is used to determine which button, if available, gets used.
PBX System Mode
Each phone is configured with 3 call appearance buttons (2 only on ETR phones). These can be used to make both internal and external calls. The dialing of an external call can be indicated by the dialing starting with a specific prefix (9 or 0) if required, otherwise any number not matching an internal extension or function is assumed to be external.
The line used for an outgoing external call is determined by configuration settings. ARS Selectors are created which can be groups of lines or specific functions using any available ISDN lines. Different external number prefixes are then mapped to those ARS Selectors. When a user dials an external number, it is matched to a selector and uses the function and one of the lines specified by that selector.
Line appearances can still be used to make and answer calls on a particular line. They can also be used to make a calls.

eMG Platform

eMG Platform eMG80-KSUADPKG
eMG Platform eMG80-MODU
eMG Platform eMG80 Hybrid Starter Bundle
eMG Platform eMG80 Digital Starter Bundle
SBG Platform SBG 1000 Starter Bundle
SBG Platform SBG 1000 Premium Bundle
SBG Platform SBG-1000
SBG Platform SBG-1K-CIU1
SBG Platform SBG-1K-CIU2
SBG Platform SBG-1K-CIU4
SBG Platform SBG-1K-CSIU
SBG Platform SBG-1K-SIPE
SBG Platform SBG-1K-TAPI
SBG Platform SBG-1K-IPN
SBG Platform SBG-1K-TNLS
SBG Platform SBG-1K-IPCRS
SBG Platform SBG-1K-IPCRC
SBG Platform SBG-1K-COMA
SBG Platform SBG-1K-COMI
SBG Platform SBG-1K-3SIPS
SBG Platform SBG-1K-3SIPC
eMG Platform eMG80-KSUAD
eMG Platform eMG80-KSUID
eMG Platform eMG80-EKSU
eMG Platform eMG80-CH204
eMG Platform eMG80-CH408
eMG Platform eMG80-CD408
eMG Platform eMG80-PRIU
eMG Platform eMG80-HYB8
eMG Platform eMG80-DTIB8
eMG Platform eMG80-CS416
eMG Platform eMG80-SLB16
eMG Platform eMG80-WTIB4
eMG Platform eMG80-VVMU
eMG Platform eMG80-MEMU
eMG Platform eMG80-RMB
eMG Platform eMG80-IPCL
eMG Platform eMG80-IPCLV
eMG Platform eMG80-VMCL

Changing System Administration Password

Changing System Administration Password
It is recommended that you change the System Administration Password after the initial administration of PARTNER Messaging has been completed, or after a service technician has accessed your system and you have provided the technician with the password. The System Administration Password can also be changed through the terminal interface.

IP office basic

Trunks
The IP Office Basic Edition - Quick Mode can support SIP trunks through its LAN connection. These are configured using IP Office Manager, they cannot be managed through phone based administration.
In order to support SIP trunks, the system must include the following resources:
· SIP Trunk Licenses
These licenses are used to configure the number of simultaneous SIP trunk calls supported, up to a maximum of
20. A IP Office Basic Edition - Quick Mode system supports 3 channels without licenses.
· Voice Compression Channels
These are required to convert between the audio compression methods used for IP telephony and those used for analog and digital trunks. Each IP500 Combination card (up to 2) installed in the system provides 10 voice compression channels for the system. One voice compression channel is used for each SIP cal

Show group

Hunt groups
This feature gives you statistical information about hunt group calls.
From Usage Metrics, start with the display showing Hunt groups.
1. Press ≠. Show group: displays.
2. Enter the group number. Hg1 metrics displays.
3. Press ≠. The display shows the date the information was last cleared.
4. Press ‘. TotalCalls: displays.
The displayed value is the total number of calls received by the hunt group, up to a maximum value of 99999.
5. Press ‘. Ans: displays.
The displayed values are the number of calls that were answered up to a maximum value of 99999 and the percentage of answered calls versus total calls up to a maximum value of 999.
6. Press ‘. Avg ans: _ sec displays.
The displayed value is the average time in seconds that it took to answer a call, up to a maximum value of 999.
7. Press ‘. Abndn: displays.
The displayed values are the number of calls that were abandoned up to a maximum value of 99999 and the percentage of abandoned calls versus total calls up to a maximum value of 999.

IP office features

Various IP Office features and applications require entry of licenses into the system's configuration. The licenses are
242
unique 32-character codes based on the feature being activated and the serial number of the feature key dongle 190 installed with the IP Office system.

The serial number is printed on the feature key dongle and prefixed with FK. It can also be viewed in the system configuration using IP Office Manager.
· For IP500 V2 systems, the feature key dongle takes the form of an SD card inserted into the control unit. The card is a mandatory item for these systems even if they use no licensed features.

Setup Auto Attendant Actions

Setup Auto Attendant Actions
This table allows you to assign which key presses have associated actions for the auto attendant.
· Type: Software level = 6.1
The auto attendant menu options can be varied according to the time of day. These radio buttons are used to select which set of actions are current displayed and editable within IP Office Manager.
· The button below the key actions table can be used to copy the current set of actions for the selected Type to all the other types in the current auto attendant. This overrides all the existing settings for actions and destinations.
· Key
The standard telephone dial pad keys, 0 to 9 plus *, # and Fax.
· Action
The following actions can be assigned to each key.
· No Action
The corresponding key takes no action.
· Dial by Name
Callers are asked to dial the name of the user they require and then press #. The recorded mailbox name prompts of matching users are then played back for the caller to make a selection. The name order used is set by the Dial by Name Match Order setting. Users without a recorded name prompt or set to Ex Directory are not included. Users can record their name by accessing their mailbox and dialing *05.
· Dial By Number
This option allows callers with DTMF phones to dial the extension number of the user they require. No destination is set for this option. The Direct Dial-By-Number setting above determines how the digits dialed with this action are used.
· Transfer to Auto Attendant: Software level = 6.1+
This option transfers the caller to another indicated auto attendant. This will skip the greeting menu of that auto attendant, playing just the current menu options greeting instead.
· Transfer to Greeting: Software level = 6.1+
This option transfers the caller to a set of prompts for recording the emergency greeting and for selecting whether the emergency greeting is active or not.
· If a system password has been set, the caller is asked to enter that password before they can continue.
· When the emergency greeting is active, it is played to other auto attendant callers before any other auto attendant greeting.
· When the emergency greeting is active, a warning is displayed on the auto attendant's Alarm Extension.

Battery charge controller

Battery charge controller

The T7406E incorporates a special handset battery charge controller that uses a four-stage charging program. The four-stage charging program sequences charging from a soft start to a fast charge, and then to a topping charge, followed by a maintenance charge.
This four-stage charge sequence presents an optimum-charging plan that helps maximize the overall life of the battery by managing two critical elements in the battery charging process: charge rate and overcharging protection. NiMh batteries do not exhibit memory problems. You do not need to drain the batteries before you recharge them.
Low battery status is indicated on the handset by the visual charge status on the display and the audible warning tone. The battery icon appears in the upper right-hand corner of the handset display. When the battery is low, the battery icon flashes. If the handset is in use and the battery is low, the battery icon flashes and a warning tone beeps every 15 seconds. If the phone is in standby mode, low battery status is not indicated.

base station DIP switches

Setting the base station DIP switches for a two-base station application
If you install two base stations, and there is interference between the two when in normal operating most, you can set one base station to operate in the band A (upper band) and the other base station to operate in band B (lower band). For the DIP switch settings for each operating mode, refer to Appendix A, “DIP switch settings,” on page 51.
1 Locate the base station DIP switches at the back of the first base station.
2 Set the DIP switches to normal operating mode for a two-base station application—base station A.
3 Locate the base station DIP switches at the back of the second base station.
4 Set the DIP switches to normal operating mode for a two-base station application—base station B.
The base stations each now operate in one half of the ISM band.

System Numbering

System Numbering
IP Office Basic Edition - PARTNER Mode systems can be configured to use either a 2-digit or 3-digit numbering for user extensions.
· If 2-digit numbering is selected, the extensions are numbered 10 to 57. This numbering cannot be changed. In 2- digit mode only 48 extensions are supported.
· If 3-digit numbering is selected, the extensions are numbered from 100 upwards. This numbering can be changed in the range 100 to 579 using system web management. In 3-digit mode a maximum of 100 extensions are supported.
· In both cases, those extensions not matched by physical ports are automatically assigned as phantom extensions

Dial Plan

Dial Plan
Extension Numbering
IP Office Basic Edition - PARTNER Mode systems can be configured to use either a 2-digit or 3-digit numbering for user extensions.
· If 2-digit numbering is selected, the extensions are numbered 10 to 57. This numbering cannot be changed. In 2- digit mode only 48 extensions are supported.
· If 3-digit numbering is selected, the extensions are numbered from 100 upwards. This numbering can be changed in the range 100 to 579 using system web management. In 3-digit mode a maximum of 100 extensions are supported.
· In both cases, those extensions not matched by physical ports are automatically assigned as phantom extensions 61 .
· The system numbering mode can be changed through the phone based administration menus,
66
Special Dialed Numbers
The following can be dialed after selecting an Intercom or call appearance button

Phantom extensions

Phantom Extensions
For IP Office Basic Edition - PARTNER Mode 6.1 and higher, extension users are created in the system configuration for all possible users regardless of whether they are matched by installed extension ports. Those user extensions without a physical port are referred to as 'phantom' extensions.
The main purpose of phantom extensions is to provide voicemail mailboxes that are not associated with an existing physical extension. These mailboxes can be accessed and used by the auto attendant menus and other functions.
· The system assumes that the base control unit is always fully populated with up to 24 extensions, either real or phantom or a mix, to which it assigns extension numbers in sequence. It does this before assigning extension numbers to any real extensions on attached external expansion modules up to the system extension limit. If the system extension limit has not been exceeded, any remaining extension numbers are assigned to additional phantom extensions.
The system web management menus allow selection of a phantom user extension number in the same was as for normal physical extension numbers. That includes using a phantom extension as the destination in an auto attendant, trunk DID call map, SIP call by call mapping, etc.
· Calls to a phantom extensions are treated as follows:
· Calls go immediately to the phantom user's voicemail mailbox. Forwarded or transferred calls go to the mailbox of the user doing the transfer or forward.
· If the phantom extension is included in a hunt group, they are ignored.
· Callers can use the phantom user's mailbox DTMF breakout settings, if configured, to be transferred to another destination.
· Calls can be transferred to a phantom extension. Since the calls go immediately to voicemail, no transfer return is supported.
· Joining or bridging to a call that has been sent to a phantom extension’s mailbox will drop the phantom extension from the call in the same way it does for a physical extension.
· Calls to a phantom extension cannot be picked up.
· The phantom extensions are supported within the auto attendant actions Dial by Name, Dial by Number and Transfer to Number.
· Mailbox access for message collection and mailbox configuration is achieved by dialing 778 from any telephone, then entering the phantom extension number and the mailbox access code if it has already been configured. Mailboxes with a configured access code can also be accessed by external calls.
· Phantom extensions can be used as the line coverage extension for a line. In this case, the phantom extension's VMS Coverage Rings setting is used before the call goes to the phantom user's mailbox.
· Auto Dial Intercom buttons can be set to route calls to a phantom extension.
· When using system web management, when selecting extensions in the various menus, a phantom extension is indicated by a # character. The extensions Equipment Type is fixed as Phantom.
· The phantom extension's Automatic VMS Coverage setting can be used to disable mailbox operation. If this is done, calls to the phantom extension will hear busy tone.

iPECS Network Management Solution (NMS) iPECS NMS is a powerful, web-based multi-site monitoring and management tool that greatly improves IT staff efficiency. NMS maintains and monitors a list of all registered system components including call servers, gateways, modules, network switches, IP phones and softphones and notifies IP staff in real-time and through automatic email alerts when critical alarms are encountered. Remote Call Control (RCC) for MS Lync iPECS RCC integrates call control in the MS Lync client for seamless voice calling, Lync presence and linkage to user’s desk phone. iPECS ClickCall By using iPECS ClickCall, simply drag and drop phone numbers from any Windows application or web page onto your PC or laptop to make outgoing phone calls from your desk phone.

iPECS Network Management Solution (NMS)
iPECS NMS is a powerful, web-based multi-site monitoring and management tool that greatly improves IT staff efficiency. NMS maintains and monitors a list of all registered system components including call servers, gateways, modules, network switches, IP phones and softphones and notifies IP staff in real-time and through automatic email alerts when critical alarms are encountered.
Remote Call Control (RCC) for MS Lync iPECS RCC integrates call control in the MS Lync client for seamless voice calling, Lync presence and linkage to user’s desk phone.
iPECS ClickCall
By using iPECS ClickCall, simply drag and drop phone numbers from any Windows application or web page onto your PC or laptop to make outgoing phone calls from your desk phone.

Default programming

The default programming data shown in Table applies even if the option units (GCDU2/ GSTU1/GVPH1) are not mounted. This enables voice mail to be used as is if the GVPH1 is inserted.
• When GCDU2 is unmounted, CO4-6 or CO4 to CO7 and DKTs 9-16 are “make-busy” and cannot be used.
• When the GCOCIH is not installed CO4 to CO7 or CO4 to CO8 are “make busy.”
• When GSTU1 is not installed, SLT2 is not “make busy.”
• When GVPH1 is not installed, VM ports 1-4

iPECS UCS

iPECS UCS Client: Basic voice calling plus advanced functionality such as video, IM, file sharing, audio conferencing, visual voice mail
• Multi-Tier Mobility: DECT and Wi-Fi phones for in-office cordless, mobile extensions to extend voice features to any phone and iPECS Communicator for full smartphone integration
• Integrated Auto Attendant / Voice Mail (AA/VM): Multi-level, multi-language AA, built- in Voice Store and Forward (VSF) Gateway
• Voice Mail Notifications: Email with or without .wav file attachment or as an alert to mobile phone
• IP Attendant: Windows-based application with powerful monitoring, Busy Lamp Field (BLF) status and attendant console features

• Automatic Call Distribution (ACD): Flexible incoming call routing, real-time agent monitoring and supervision and call record statistics
• Centrally-Controlled T-Net: Greatly enhances survivability, server and geographic redundancy and management of local and remote sites
• System Geographic Redundancy: Hot standby call server for seamless hand-over in case of main server failure
• Multi-Language: Simultaneous support for up to 16 languages in all features
• Power Backup: Power redundancy features with main cabinet installation

CA48

PARTNER-CA48 Intercom Autodialer 2
You can connect a PARTNER-CA48 Intercom Autodialer to a PARTNER telephone at an operator’s extension. The Intercom Autodialer is shipped with an adapter, a D8W line cord, a power cord, a power unit, and a button-labeling sheet.
You can wall-mount the Intercom Autodialer to work next to a wall-mounted system telephone. See the instructions provided with the Autodialer

B channel (Bearer channel)

B channel (Bearer channel). An ISDN standard transmission channel used for voice or data transmission.
Base station: A hardware component that is mounted on walls and ceilings to provide a radio link to an office or other area where portable telephones are used. Each base station houses two radios that allow portables to send and receive calls through the ICS.
Background Music: A feature that allows you to hear music from the speaker of your Norstar telephone. It is available only if a music source has been attached to the ICS and the feature has been enabled under Feature settings in Sys prgrmmng.
Basic password: A one- to six-digit password that prevents unauthorized access to the most commonly-used telephone settings. The Basic password can be assigned and changed in Passwords programming.
Basic Rate Interface (BRI): An ISDN interface which uses two
B channels and a D channel (2B+D).
Bearer channel: See B channel.
Business series terminals These terminals were developed to replace the Norstar M-series terminals. Therefore, their functionality closely parallels the M-series terminals.
button caps: Interchangeable plastic caps that fit over the buttons of legacy Norstar M-series telephones. They are used to indicate the features programmed onto each programmable memory button. Button caps are either pre-printed or have clear windows that allow you to insert labels.
Button Inquiry: ≤•‚ With this feature you can check the function of each programmable button on your system telephone.
C
Call Duration timer: ≤‡‡ A feature that allows you to see how long you spent on your last call, or how long you have been on your present call.
Call Forward: ≤›
A feature that forwards all the calls arriving at your telephone to another telephone in your Norstar system. To have calls forwarded outside the system, use Line Redirection.
Call Forward No Answer: A feature that forwards all calls arriving at your telephone to another designated telephone in your Norstar system after a specific number of rings. Call Forward No Answer is assigned under Capabilities in Terminals&Sets programming.
Call Forward On Busy: A feature that forwards all calls at your telephone to another designated telephone if your telephone is busy. This feature is assigned under Capabilities in Terminals&Sets programming.
Call Forward Override: An automatic system feature that allows you to call someone and ask them to stop forwarding their calls to you.
Call Information: ≤°⁄⁄ Call Information allows you to display information about incoming calls. For external calls, you can display the caller name, telephone number and the line name. For an internal call, you can display the name of the caller and the internal number. You can obtain information about ringing, answered, or

PC Card

PC Card Slots 1
The processor module has two PCMCIA (Personal Computer Memory Card International Association) interface slots (hereafter referred to as PC Card slots). You can buy PC Cards to use in these slots for the following purposes:
■ Use a Backup and Restore PC Card to backup or restore telephone and system programming.
■ Use a PC Upgrade card. After powering down the system, you insert the PC Upgrade Card and turn the power back on. While the system upgrades, the bicolor (red/green) power LED on the processor flashes for about 20 seconds (40 seconds when a 1600 DSL module has been installed; one minute when a T1 module has been installed); then the power LED becomes steady green. If there is firmware on the PC Card that is downloaded to the module, then it will take longer for the power LED to become steady green. All of your system and extension programming is saved and ready to work with the new release.
■ Use a PC Card to store Automatic System Answer and Direct Extension Dial (ASA/DXD) messages.
■ Use a PARTNER Voice Messaging PC Card to provide messaging features (store personal greeting and store and retrieve callers’ messages) for an increased number of mailboxes.
■ Use PARTNER Remote Access PC Card, which allows you to program the system remotely or locally from a PC and perform backup and restore functions. In order to program the system remotely, you must use the PARTNER ACS R7.0 PC Administration software, which is available from your Avaya representative or authorized dealer.
If you use the PARTNER ACS R7.0 PC Administration software, the content of the PC Cards that are installed in the PARTNER ACS processor module as well as all modules installed in the carrier will be displayed in an inventory screen.

Processor Module

Processor Module 1
The PARTNER ACS processor module provides the software intelligence that controls the system’s features. It has jacks for five outside lines, nine enhanced tip/ring extensions, a Music- On-Hold audio source, a grounding screw, a jack that supports an adjunct for two Contact Closures, and a jack for a call reporting (SMDR) device, such as a printer. You can connect a loudspeaker paging system to any line jack on the PARTNER system. (If your loudspeaker paging system requires a “dry contact relay,” you must connect the loudspeaker paging system to line jack
5.) The processor module also has two PC Card slots, a two-color red and green light-emitting diode (LED), and two AAA user-replaceable batteries. The module provides support for integrated Caller ID information on system display telephones. The system requires one processor module

Auto Speaker

iPECS Phones programmed for Auto Speaker Select can access a CO/IP line or an internal call by pressing the appropriate button without the need to lift the handset or press the [SPEAKER] button. Audio from the CO/IP line or called station is sent to the speaker as if the user pressed the [SPEAKER] button and the speakerphone's MIC is activate.

Optional (Unforced) Account

Optional (Unforced) Account Codes
Optional Account Codes allow a keyset extension user to enter an Account Code while placing a trunk call or any time while on a call. This type of Account Code is optional: the system does not require the user to enter it. If the keyset user is already talking on a trunk call, their conversation continues uninterrupted while they enter an Account Code.