Telephone Systems Installers New York NY NYC Phone systems installations Brooklyn,Queens,Bronx,VoIP

Unforced 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.

IP-telephony platform, SIP and DECT

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

UCS Client

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

Conference call and Collaboration

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

Changing Settings

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 Restriction

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

display different languages

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

T1 module

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.

Call Pick-up

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.

System 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®,

Hybrid 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.

Equipment damage

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.

PRI Installer

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

Four SIP trunks

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 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

Attendant Interposition Calling

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.

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 Restriction IntraMail

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

Line calls

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 expansion

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.

PROBLEM: System terminates recording

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

SIP Virtual mobile extension

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 down

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.

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

Automatic 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)

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 message

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.

Call Waiting

Call Waiting (#316) 4
Use this feature to activate Call Waiting. When you are on a call and a second intercom, transferred, or outside call comes in, you hear two beeps to indicate you have a second call waiting to be answered. Press the switchhook (or Recall or Flash button, if available) to put the current call on hold and retrieve the second call. To return to the first call, press the switchhook again.

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.

Operator Group

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 signaling

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

Least Cost Routing

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.

UCP

Scales with your Growth › iPECS UCP is designed to deliver the flexibility you need as your organization grows
– 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

iPECS 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

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/Connection 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.

Small Community Networks

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.

Scanning the base station radio frequency

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

CIX40 system is available on two hardware versions

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

Remote call control

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.

Multi-Tier Mobility

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 eMG80-KSUADPKG

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

Admin 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.

The IP Office Basic Edition - Quick Mode

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

Hunt groups

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 and applications

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

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

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.

Setting the base station DIP

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.

IP Office Basic Edition

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.

Network Management Solution

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 2-3 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

PARTNER-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

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 Slots

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 Select

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.

Unforced Account Codes

Call log feature

enhanced Call log feature allows you to log all calls to a telephone, or to gather logs for specific lines assigned to a telephone. The Call log set feature allows you to determine which assigned lines will collect logs. Refer to Call log set on page 358 and Call log on page 436. (all profiles)
• The second dial tone table allows the user to enter up to 10 one to four-digit numbers that, when dialed, will cause the system to produce a second dial tone, at which time the user can enter the remaining call digits. Refer to Configuring the second dial tone table on page 403. (all profiles)
• Profiles 1 and 4, PRI: The Send Name Display feature allows you to specify if you want the business name and OLI to be transmitted over specific PRI lines.

TAPI

Enabling TAPI Server and User Administration
To allow other Domain users to use the TAPI Services the TAPI server must be enabled. This is per- formed by using the TCMSETUP command.
7) Click Start/Run.
8) Enter the following command in the Run command line. TCMSETUP /S [/n] domain\username [password]
The following example installs the Server using the Administrator account within the CTI domain using the Administrator’s password Passw0rd.

LAN Administrator

Your LAN Administrator will provide these parameters. These parameters can only be installed into the PARTNER or MERLIN Messaging system using the module’s RS-232 serial interface and a PC-based communication application, such as HyperTerminal. The HyperTerminal application is available on any PC running a Windows (95, 98, NT, 2000, or XP) operating system. The PC must have its serial port connected to the RS-232 labeled port on the front panel of the messaging system module. This will require a D8W cable (up to 25-feet), a 355AF adapter, and a male DB-25-to-female-DB-9 connector. For more information regarding this serial connection, see the section “Establishing a session” in the “Terminal Based Reporting, Diagnostics and Maintenance” section of the MERLIN Messaging Release 4.0 or PARTNER Messaging Release 7.0 Installation, Programming, and Trouble Shooting Guide Online contained on the Library CD.
The following steps describe how to install the assigned Static IP, Subnet Mask and Default Gateway addresses on the messaging system:
1. Connect the RS-232 cable from a PC to the RS-232 serial port on the messaging module.

Modem

MODEM BOARD (see Figure 3-14)
This board has no selectable options and installs on the IOM board (see Figure 3–15). The MODEM board provides a 56Kbps/V90 modem for communicating with the system remotely. Place the modem board face down over the two connectors, taking care to properly line them up, and lining up the holes in the MODEM board and IOM board, and press gently on the back of the board to seat it.
3.10 IOM BOARD (see Figure 3-15)
This board has no selectable options and installs on the system main cabinet (see Figure 3–16). Line the IOM board up with the card slot and slide it in. Press gently on the center of the IOM board front panel to seat it and secure the board with the two screws.
3.11 TRUNK B CARD (see Figure 3–17)
This card has no selectable options. Insert as many Trunk B cards as are needed into any universal slots (see Figure 3–1). Push firmly in the middle of both card ejectors on each card to ensure that it is fully inserted into the back plane connector.

Network Time Protocol (NTP)

When enabled, the system automatically determines and sets the time of day employing Network Time Protocol (NTP) or ISDN time messages. When using NTP, the system, at ten
(10) minute intervals, requests the time from the specified NTP time server and receives GMT time. This feature allows the System Time to synchronize with the NTP time server automatically. If the time deviates more than two (2) seconds, the system clock is adjusted to match the NTP server.
When using ISDN, the system receives the time of day in ISDN messages and automatically adjusts the time if the time in the system deviates from the ISDN time.

ACS 509

Features Available with Release 7.0 1
■ The PARTNER ACS 509 processor module, which contains up to five lines and nine extensions.
■ Loudspeaker Paging Line to allow you to connect a loudspeaker paging system to any line jack on any PARTNER module.
■ Enhanced backup/restore capabilities to allow you to back up the PARTNER system programming to either the internal memory of the PARTNER ACS R7.0 processor module or a PC Card.
■ An RJ-45 serial port (Admin port) on the PARTNER ACS R7.0 processor module to allow you to connect to the PARTNER system either directly from the COM port of a PC or via an external modem that is connected to the Admin port and administer the system via PARTNER ACS R7.0 PC Administration software.

SLT phone

A SLT phone can be received CLI from internal caller instead of station number by programming when it’s tranfering.
Operation
SLT
To transfer to SLT when it’s CO call
1. An User answers a call from CO with CLI
2. The User is transfering to another SLT
3. The SLT can see CLI of CO instead of station number
Conditions
1. If this function works well, the system should be set ORI in ADM 114-B18.

Connectors

Connectors
Before programming System Data, the battery must be connected to CN4 to allow memory retention when a power failure or brownout occurs. When a brownout or power failure does occur, and the battery backup circuit is not activated, System Data resets to the default values, all stations in the system reset to the default values, and any data programmed on individual stations is cleared

ANSWERING CALLS

ANSWERING CALLS
Ringing Calls
1. Lift the handset.
2. Converse.
Note: When assigned the Delayed Ringing feature, incoming calls will first ring at a primary station. If the call is unanswered after a preprogrammed time interval, calls will also ring at the assigned secondary station.
Ringing Calls To A Second Incoming Extension/CAR/VE Key
1. Lift the handset.
2. Converse.
Note 1: A Second Incoming Extension key must be assigned to appear and ring.
Note 2: A Line key must be available to answer an incoming outside call.
Voice Announce Calls
1.Ensure the MIC LED is lit.
2. Adjust the Speaker volume (W or V) as needed.
3. Respond handsfree.
Note: The handset may be used at any time during the conversation.

Ringing Calls

SV8300 Maximum System Capacity

SV8300 is a full-featured IP based communications system providing a rich set of existing system features, with pure Voice over IP (VoIP) communications, across corporate Local and Wide Area Networks (LAN and WAN).
The DT700 series telephones provide a converged infrastructure at the desktop, with a 10Base-T/100Base-TX connection to the LAN and built-in hub for a PC connection to the telephone itself. The system can provide peer-to-peer connections between the DT700 series telephones with voice compression, offering existing IP telephone features with an enhanced user interface. On the WAN side, the system can provide peer-to-peer connections over IP networks with the voice compression, on CCIS over IP or Remote Unit over IP.
The SV8300 can provide legacy line/trunk interfaces to support the existing Time Division Multiplexing (TDM) based infrastructure, such as analog telephones, digital telephones (DT300 series), analog networks and digital networks (T1/E1, ISDN, etc.).
The 19” chassis provides 104 total ports (80 digital terminals) and can be expanded, using three additional 19” chassis, for a maximum of 416 ports (368 digital terminals). Through IP connection and four additional 19” chassis, the system can be expanded to a maximum of 1024 ports on the SV8300 (refer to Table 2-2 SV8300 Maximum System Capacity – Slot/ Port/Channel on page 2-5).
Communications between legacy stations/trunks and IP stations/IP networks are made using a VoIP daughter board, which converts packet- based voice data to TDM-based voice data, and vice versa. Both peer-to- peer connections and TDM-based connections are controlled by the CPU blade. The CPU incorporates a built-in Device Registration Server (DRS) and a single interface point of IP connection to IP terminal, PCPro and OAI / ACD servers. Figure 1-1 Simplified SV8300 System Connectivity on page 1-2 illustrates a simplified view of the SV8300 system connectivity.

Hibrid mode

Hybrid Mode
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.

Controlling system access

Controlling system access
Modular ICS programming provides several ways of protecting your Norstar system from unauthorized access or use. These features are used in combination to provide control both to users and to lines.
Class of Service
Class of Service (COS) refers to the capabilities that Norstar provides to users who access the system from the public or private network. A password defines which set of capabilities the user has access to.
The COS includes:
• filters that restrict dialing on the line
• an access package defines the set of line pools that may be accessed and whether or not the user has access to the paging feature
The COS that is applied to an incoming remote access call is determined by:
• the filters that you apply to the incoming trunk
• the Class of Service password that the caller used to gain access to the Norstar system
In cases where DISA is not automatically applied to incoming calls, the remote caller can change the class of service by dialing the DISA DN and entering a COS password.

Boot ROM

Boot ROM (Flash ROM)
For Boot ROM, the flash ROM of 1 Mbytes(512 Kbytes × 2 EA) is used. Boot ROM has the program for the initial operation of system(system operation program), and performs the downloading the main program from the flash ROM, which is the Smart Media Card into the SDRAM.
SDRAM
128 Mbytes(Extension into 256 Mbytes can be made) are basically to the 256 SDRAM, and is composed of four 32Mbytes elements. SDRAM has the main program resident to operate, and saves the data necessary for processing the system. In addition, SDRAM is designed to operate in No Wait for the maximum performance of the system.
SRAM
2 Mbytes are provided to SRAM, and is composed of two 1Mbytes element. SRAM can perform the memory backup, and saves the various DB information. The various DB information necessary for the system operation are to be saved at SRAM, but they should be always saved also at the smart media. By doing so, the DB information can be safely saved even when the system is critically damaged from the outside.
NAND_Flash ROM
NAND_FROM is composed of 512 Kbytes, and is used as the subsidiary memory device of OS-7400 system. It saves various DB information.
Smart Media
Smart media is the NAND flash memory card that has saved the main program(OS) for operating MP40 at itself. In addition, is is used as the auxiliary memory of OS-7400 System. The currently used smart media is 32 Mbytes, and 8-bit parallel bus connection mode is used for the connection to the system, which can be used without the change of hardware even though the capacity increases(But the change of the software is needed.) Smart media should operate always with itself mounted on the system, and should be saved in the vinyl packed when the card is delivered.
Engine (SAMSUNG ASIC: STL7065C)
The engine of MP40 uses one of Samsung ASIC. If the extension rack connected to MP40 is LCP, the engine provides the function by which the individual compensation for the voice signal connected by the subscriber card is made.

Digital Call Screening and

Digital Call Screening and Messaging System with 12 Minutes of Flash Memory
The AR-1 Automated Receptionist is a digital call screen- ing and messaging system designed to answer calls with an informative greeting then, let callers ring through to talk to someone live, select from other prerecorded messages or hang up.
Programming functions and message recording are ac- cessed locally or remotely using a standard Touch Tone phone. Recordings from tape, CD, PC, etc. can also be locally loaded via the AR-1’s 3.5mm analog audio input. Twelve (12) minutes of total record time and up to 8 dif- ferent messages can be recorded into the unit. The mes- sages consist of one Day greeting, one Night greeting, one Transfer message, four Menu messages and one Goodbye message.
The AR-1 has a call screening option which allows callers with the correct code to bypass the greeting message and ring through. The unit has a programmable Touch Tone Distinctive ring through option allowing callers to enter 0,7,8, or 9 to ring the phone port in 4 different Distinctive/Custom ring patterns. The unit has an optional automatic ring through mode allowing calls

T1

Initial Configuration of the T1 Module 2
To configure and maintain a T1 module, you must use the PARTNER ACS R7.0 PC Administration software. You cannot program a T1 module via System Programming from extensions 10 or 11. For instructions on how to install and use PARTNER ACS R7.0 PC Administration, see the PARTNER® ACS R7.0 PC Administration Getting Started.
To set up a T1 module in a PARTNER system:
1. Start PARTNER ACS R7.0 PC Administration.
a. Perform one of the following steps:
— Click on the Start menu, and select Programs>PARTNER ACS R7.0 PC Administration>PC Administration. (This is the default location. If you chose to save your files to an alternate location during installation, your path will be different.)
— If you created a Start menu short-cut, click on the Start menu, and select PC
Administration.
— If you created a desktop short-cut, double-click on the PC Administration icon on the desktop.
The New Translation Creation dialog box appears.

DAY/NIGHT ANSWERING

DAY/NIGHT ANSWERING
After you set up DILs to InMail, Call Routing Mailbox 1 answers outside calls. This is because:
• All trunks by default are assigned to Answer Table 1.
• Answer Table 1, which runs Monday-Friday from 8:30 AM to 5:00 PM, uses Call Routing Mailbox 1.
• The Default Mailbox, which runs at all other times (evenings and weekends), also uses Call Routing Mailbox 1.
To Set up simple Day/Night Answering:
• In 47-11-03 Default Mailbox Category and in 47-11-03 Default Mailbox Number change the Default Mailbox for Answer Table 1 to 2.
• From the System Administrator Mailbox:
1. Dial SA + I and record a new Instruction Menu message for Call Routing Mailbox 1.

Emergency phone number list

Emergency Phone Number List (#406) 3
Use this feature to create a list of up to 10 telephone numbers that all users can dial regardless of dialing restrictions, provided they have access to an outside line. Typical list entries include fire, police, and emergency services numbers.
You can use Emergency List entries to override numbers that would otherwise be disallowed. For example, if you created Disallowed List entries to prevent calls to area code 201, but you want to allow calls to a specific number in that area code (such as the number of a manager who is on call after hours), put the number in the Emergency List.

W-1000/W-1000-EWP W-2000A/W-2000A-EWP

Answer Your Door or Gate From Any Phone in Your Home or Business W-1000/W-1000-EWP W-2000A/W-2000A-EWP Now you can answer your door or gate from the safety of your telephone! The W-1000 (flush mount) and W-2000A (surface mount) doorboxes are designed to be installed on the unused telephone line input of nearly any phone system. One or two doorboxes can also share an existing residential phone line and provide door strike and CCTV camera control when used in combination with a C-1000B doorbox controller. When the “Call” button is pressed, the doorbox generates a standard ring cadence for an adjustable number of rings. To converse with the visitor, the inside party simply answers the call from the phone of their choice. The W-1000-EWP and W-2000A-EWP share all of the features of the W-1000 and W-2000A in addition to Enhanced Weather Protection (EWP) for outdoor installations where the unit is exposed to precipitation or condensation. EWP products feature foam rubber gaskets and boots, sealed connections, gelfilled butt connectors, as well as urethane or thermal plastic potted circuit boards with internally sealed, field-adjustable trim pots and DIP switches for easy on-site programming. For more information, see DOD# 859.

Door entry Points

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

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

Caller ID Call Return

Call Waiting/Camp-On S S S S Callback S S S S Caller ID Call Return S S S S Caller ID S S S S Caller ID - Flexible Ringing S S S E Caller ID - Shared Logging S S S S Central Office Calls, Answering S S S E Central Office Calls, Placing S S S S Class of Service S S S S Clock/Calendar Display/Time and Date S S S S Code Restriction/Toll Restriction S S S S Code Restriction Override/Toll Restriction Override S S S S Code Restriction, Dial Block/Toll Restriction, Dial Block S S S S Conference S S S S Conference, Remote S S S S Conference, Voice Call/Privacy Release S S S S Continued Dialing S S S S Cordless DECT Telephone S S S S Data Line Security S S S S Delayed Ringing S S S S Department Calling S S S S Department Step Calling S S S S DHCP Client S S S S Dial Pad Confirmation Tone S S S S Dial Tone Detection S S S S Dialing Number Preview S S S S Digital Call Logging N/A N/A N/A S Digital Trunk Clocking S S S S Direct Inward Dialing (DID) S S E S Direct Inward Line (DIL) S S S S Direct Inward System Access (DISA) S S S S Direct Station Selection (DSS) Console S S S S Directed Call Pickup S S S S Directory Dialing S S S S Distinctive Ringing, Tones and Flash Patterns S S S E Do Not Disturb (DND) S S S E Door Box S S S S Drop Key S S S S Ecology S S S S E911 Compatibility S S S S Flash S S S S Flexible Ring N/A S S S Flexible System Numbering

Programing tools

Programming tools
This section describes how to set up a telephone for system programming.
You will note that two different sheets of programming overlays came with your Programming Record. Use the one that was created for the telephone you plan to use to program your system.
You can use a two-line display telephone, such as a 7316E digital phone for programming. Use the appropriate overlay for the model of telephone you use, to help you understand where the programming navigation buttons are. The table below shows the Digital phone button icons. If you are programming with a Norstar legacy telephone, the icons may differ slightly:

ACD Announcement

ACD offers extensive overﬂow options for each ACD Group. For example, a caller ringing in when all agents are unavailable can hear an initial announcement (called the 1st Announcement). This announcement can be a general greeting like, "Thank you for calling. All of our agents are currently busy helping other cus- tomers. Please stay on the line and we will help you shortly." If the caller continues to wait, you can have them hear another announcement (called the 2nd Announcement) like, "Your business is important to us. Your call will be automatically answered by the ﬁrst available agent. Please stay on the line." If all the ACD Group's agents still are unavailable, the call can automatically overﬂow to another ACD Group or the Voice Mail Automated Attendant. If all agents in the overﬂow ACD Group are busy, Lookback Routing automati- cally ensures that the waiting call will ring into the ﬁrst agent in either group that becomes free.

Using Remote Call Forwarding

Using Remote Call Forwarding on a System Telephone 8
To forward calls manually:
1. Press f11.
2. Dial your extension number.
3. Dial the two-digit Personal Speed Dial code (80-99) that contains the destination outside telephone number at which you want the calls to ring.
To remove Remote Call Forwarding manually:
1. Press f11.
2. Dial your extension number twice.
To forward calls using the programmed button:
1. Press the programmed button.
If an originating extension and a Personal Speed Dial code (80-99) were programmed on the button, Remote Call Forwarding is activated (on a button with lights, the green light is on). Skip Steps 2 and 3.
2. If no originating extension was programmed, dial your extension number.
3. If no destination telephone number was programmed, dial the two-digit Personal Speed Dial code (80-99) that contains the destination outside number at which you want the calls to ring.
Remote Call Forwarding is activated (on a button with lights, the green light is on).
To remove Remote Call Forwarding by using the programmed button:
1. Press the programmed button.
If an originating extension and a destination extension were programmed on the button, Remote Call Forwarding is deactivated (on a button with lights, the green light is off). Skip Steps 2 and 3.
2. If no originating extension was programmed, dial your extension number.
3. If no destination extension was programmed, dial your extension number again. Remote Call Forwarding is deactivated (on a button with lights, the green light is off).

Master Communications, Inc

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