PcPro WebPro

In addition to the keyset programming, the Aspire system provides the ability to use a PC to access system programming. The Windows-based PCPro and the HTML-based WebPro allows you to:
● Edit the telephone system programming options from a remote location.
The PCPro application requires changes to be uploaded to the system before they take effect. The WebPro application applies the changes as soon as the APPLY or OK icon is clicked. ● Access system maintenance functions (like reports and tests).
In addition, PCPro allows you to:
● Save your programming to your PC’s hard disk - then upload it via a LAN (Local Area Network), PPP, serial or USB connection.
PPP is a protocol that allows a computer to use a regular telephone line and a serial interface (modem) to make TCP/IP connections.
Local programming is possible using the LAN, USB or serial connection (the USB connection is not avail- able on the Aspire S) (CTA and CTU adapters cannot be used). Remote programming is possible using the dial- up serial port connection or LAN connection provided there is internet/network access to the IP address of the CPU.
● Download the existing programming in the telephone system via a LAN, PPP, serial or USB connection - and save it to your PC’s hard disk.
● Set up a default database with the settings you use most often.
● Create a unique database for each phone system you have installed. Since you save the site-specific data to your PC’s hard disk, you can easily retrieve a customer’s programming if something goes wrong.
Aspire S/Aspire System Requirements
● Aspire S system software 2.50 or higher ● PCPro: NTCPU LAN, Serial or USB Connection or any version of Aspire system software to PC (USB on Aspire M/L/XL Only)
● Aspire S: ENTU for LAN Connection WebPro: TCP/IP via LAN Connection to PC
PCPro PC Requirements
● CPU: Pentium II 500 MHz equivalent or higher ● 128Mb of RAM
● 20Mb of Hard Drive Space ● Monitor Resolution: 800 x 600 pixels or higher ● Mouse ● Microsoft Windows 2000/XP
● Internet Browser: Internet Explorer 6.0 or higher or ● Network Interface Card (NIC) Netscape 6.0 or higher (depending on connection type)
WebPro PC Requirements
● CPU: Pentium III 600 MHz equivalent or higher ● Monitor Resolution: 800 x 600 pixels or higher ● Mouse ● Microsoft Windows 2000/XP
● Internet Browser: Internet Explorer 6.0 or higher or ● Network Interface Card (NIC) Netscape 6.0 or higher (depending on connection type)

Hold

Hold lets an extension user put a call in a temporary waiting state. The caller on Hold hears silence or Music on Hold, not conversation in the extension user’s work area. While the call waits on Hold, the extension user may process calls or use a system feature. Calls left on Hold too long recall the extension that placed them on Hold. There are four types of Hold:
● System Hold
An outside call a user places on Hold flashes the line key (if programmed) at all other keysets. Any keyset user with the flashing line key can pick up the call.
● Exclusive Hold
When a user places a call on Exclusive Hold, only that user can pick up the call from Hold. The trunk appears busy to all other keysets that have a key for the trunk. Exclusive hold is important if a user doesn’t want a co-worker picking up their call on Hold.
● Group Hold
If a user places a call on Group Hold, another user in the Department Group can dial a code to pick up the call. This lets members of a department easily pick up each other’s calls.
● Intercom Hold
A user can place an Intercom call on Hold. The Intercom call on Hold does not indicate at any other extension.
With Automatic Hold enabled (Program 15-02-07), when the user is on an outside call using the handset, the user can press a flashing line/loop key to answer an incoming call without disconnect- ing their first call. The first caller is automatically placed on hold. This feature does not work using handsfree or when the user is on an ICM and presses a flashing line/loop key (the ICM call is disconnected).
Hold Recall to Operator
Hold Recall to Operator enhances how the system handles calls that have been left on hold too long. With Hold Recall to Operator:
● A trunk call recalls the extension that placed it on Hold after the Hold/Exclusive Hold Recall time.
● The recalling trunk will ring the extension that placed it on Hold for the Hold/Exclusive Hold Recall Callback Time.
● After the Hold/Exclusive Hold Recall Callback Time, the trunk call will ring the operator.
Hold Recall to Operator applies to trunk calls placed on System Hold, Exclusive Hold and Group Hold. It does not apply to Intercom calls.
Conditions
The called extension must lift the handset or press the SPK key before the call can be placed on hold.
Default Set

Computer Telephony Integration (CTI)

Computer Telephony Integration (CTI). It uses the Telephony Application Programming Interface (TAPI) 2.1 protocol. To allow CTI an Ethernet inter- face is present on the NTCPU.
TAPI 2.1 based CTI realises third party call control features such as ACD, Predictive Dialing, and Call Routing.

General Description
The following section assumes you have installed the Aspire TSP. This section contains further infor- mation on configuring TAPI2.1 on Windows NT Server.
The following information is provided:
• Aspire TSP Configuration
• Enabling TAPI Server
• TAPI Server User Administration

Aspire WebPro

WebPro
The WebPro software is installed on the Aspire NTCPU PCB - there is no separate software installation. This means that when the Aspire system software is updated, the WebPro software is updated as well.
Make sure the SW3 switch on the Aspire NTCPU (switch just above the serial port connector) is set to OFF (down).
The Aspire system (Program 10-21-02) and the Dial Up setting must be set to use the same baud rate (19200 by default).
Make sure the PC is connected to the serial or USB port on the Aspire NTCPU with a null modem (cross- over) cable. For LAN connections, use a straight-through cable if connected through a hub. If connected directly to the Aspire S/Aspire LAN connector, use a cross-connect cable.
The WebPro software provides a Help system if you experience difficulty in using the program. Simply press the ‘F1’ key.
1.Make sure the required cable (USB, serial, LAN) is connected from the PC to the Aspire S/Aspire system. 2.When using a LAN connection, skip to Step 5.
When using a connection other than a LAN, you must first connect to the Aspire using the dial-up connection. If not already connected, click on START  SETTINGS  NETWORK CONNECTIONS  select the dial up connection to be used to connect to the Aspire system.
3.In the window that appears, if the user name and password are acceptable, click CONNECT or DIAL depending on your connection type.
Only one person is allowed in programming mode at a time. An error message will be received if trying to log in while another user is already in programming mode.
4.If the PRE-DIAL TERMINAL SCREEN option was selected in the dial-up setting, when it appears, left click on the black area of that screen. Type AT and press ENTER. Once OK has been displayed, click on CONTINUE and wait for the computer to connect to the Aspire system.
If an OK does not appear on the screen, continue to type AT and then press enter until you get an OK on the screen.
5.Once the connection has been established, with either Internet Explorer or Netscape Navigator installed, open the internet
browser application.
6.Enter the IP address of the Aspire system (example: http://192.78.0.1). This address is selected based on the type of con- nection to the Aspire.
N 47

WebPro

WebPro
Using WebPro
The WebPro software is installed on the Aspire NTCPU PCB - there is no separate software installation. This means that when the Aspire system software is updated, the WebPro software is updated as well.
Make sure the SW3 switch on the Aspire NTCPU (switch just above the serial port connector) is set to OFF (down).
The Aspire system (Program 10-21-02) and the Dial Up setting must be set to use the same baud rate (19200 by default).
Make sure the PC is connected to the serial or USB port on the Aspire NTCPU with a null modem (cross- over) cable. For LAN connections, use a straight-through cable if connected through a hub. If connected directly to the Aspire S/Aspire LAN connector, use a cross-connect cable.
The WebPro software provides a Help system if you experience difficulty in using the program. Simply press the ‘F1’ key.
1.Make sure the required cable (USB, serial, LAN) is connected from the PC to the Aspire S/Aspire system. 2.When using a LAN connection, skip to Step 5.
When using a connection other than a LAN, you must first connect to the Aspire using the dial-up connection. If not already connected, click on START  SETTINGS  NETWORK CONNECTIONS  select the dial up connection to be used to connect to the Aspire system.
3.In the window that appears, if the user name and password are acceptable, click CONNECT or DIAL depending on your connection type.
Only one person is allowed in programming mode at a time. An error message will be received if trying to log in while another user is already in programming mode.
4.If the PRE-DIAL TERMINAL SCREEN option was selected in the dial-up setting, when it appears, left click on the black area of that screen. Type AT and press ENTER. Once OK has been displayed, click on CONTINUE and wait for the computer to connect to the Aspire system.
If an OK does not appear on the screen, continue to type AT and then press enter until you get an OK on the screen.
5.Once the connection has been established, with either Internet Explorer or Netscape Navigator installed, open the internet
browser application.
6.Enter the IP address of the Aspire system (example: http://192.78.0.1). This address is selected based on the type of con- nection to the Aspire.

Station Message Detail Recording

Station Message Detail Recording
SMDR provides a printed record of your calls.
Description
DSX | Features | 527
Station Message Detail Recording (SMDR) provides a record of the system’s outside calls. Typically, the record outputs to a customer-provided printer, terminal or SMDR data collection device. SMDR allows you to monitor the usage at each extension and line. This makes charge-back and traffc management easier. SMDR includes both incoming and outgoing calls, and can be turned off system-wide or selectively for each line.
The SMDR call record outputs when the call completes. The system assigns the SMDR record to the last extension on the call. For example, if extension 306 answers the call, talks for 20 minutes, and then transfers the call to extension 302, extension 302 “owns” the entire call record as soon as they hang up.
SMDR requires a customer-provided data collection device connected to the system’s RS-232 port. The default baud rate is 38,400. The data format is fxed at 8 data bits, no parity, with one stop bit (8N1). Connection requires:
• Adaptor P/N 1091014 to connect to the 9-pin RS-232 port on the data collection device.
• A standard 6-conductor line cord to connect the adaptor to the system’s RS-232 port.
SMDR does not buffer records when the data collection device is disconnected.
Call Duration Independent of System Clock
The duration of a call on the SMDR report is calculated independently of the system clock. This prevents changes made to the system Time and Date from inaccurately reporting the call duration after the Time and Date change. The automatic Daylight Savings Time adjustment also will not affect the call duration.

Battery Backup

Battery Backup
The system provides permanent backup of system memory.
Description
DSX
In the event of commercial AC power failure, the NAND Flash memory on the CPU PCB permanently maintains the site database. Additionally, an internal battery on the CPU provides short-term backup of the system date and time (Real Time Clock) and certain station parameters (such as the Caller ID log). The battery will hold the Real Time Clock and station parameters for up to 10-14 days. When commercial AC power is restored, the system restarts with all programming and the time and date intact.
Additional Battery Backup capability can be provided by a customer-supplied Uninterruptable Power Supply (UPS). The length of time the UPS will power the system when power fails depends on the capacity of the UPS unit. Consult with the UPS manufacturer for the specifcs. Refer to the Hardware Manual for additional details.

Voice recorder

A voltmeter is helpful in determining if your inputs are “wet loops” or dry lines. Wet loops typically carry -48V DC when measured off-hook. Wet loops are used to connect to your telecom provider, to your PBX, or to analog phone extensions. Dry lines such as headsets, radios, microphones and speakers carry audio, but no DC signaling voltage.
After you’ve determined a line type (either wet-loop or dry line), select a recording trigger: either Loop Start or VOX. Outside wet loops and inside analog extensions normally use Loop Start signaling. All dry lines use VOX signaling to trigger recording.
VOX Recording for Dry Lines
To bridge a dry line using Voice Operated Switch (VOX):
• Note: VOX mode is used for dry lines (lines that do not carry signaling voltage). Examples are handset connections, headsets, radios, microphones and speakers.
• Use a punch-down block, handset Y connector, breakout box, or RJ- 11 octopus cable to connect each line (two twisted wires) to the logger. Each of the logger inputs connects in “bridge” fashion across the existing wiring.
• Verify that the line audio level is within telecom specifications. Peak level should not exceed +3dBm. Gain controls should be set so that recording levels do not exceed 0dBm. Use Level Boost only if the input level is below -20dBm. If the VU meter peaks, the line level is too high. Reduce it using the manual gain controls, turn off Level Boost, and/or switch Auto Level on.
• Use Level Boost for handsets, microphones, and low-level signals only.
• Rename the line label by clicking on the line name. Type the new name and press Enter. For example, a handset tap or analog extension connected to LINE 01 might be renamed “Support Desk.”
• Set the line configuration mode to VOX by clicking on the line mode.

PAGING Internal,External & All Call Page

PAGING Internal,External & All Call Page
Description
A station, which is permitted to access page facilities, can connect and transmit voice announcements to any or all of the systems Internal,External Page zones. Stations are grouped into “zones” to receive pages to the zone. Stations not assigned to any zone will not receive a page including All Call pages.
A page warning tone, if assigned, will be provided to the Page Zone(s) prior to the audio connection. The user is allowed to continue the page for the specified Page Time-out timer after which the user is disconnected and the Page Zone(s) is returned to idle

Do Not Disturb Override

Do Not Disturb Override
Easily override a co-worker’s Do Not Disturb.
Description

Do Not Disturb Override lets an extension user override another extension’s Do Not Disturb. This allows a priority employee (such as a supervisor or executive) to get through to a co-worker right away while the co-worker’s phone is in Do Not Disturb. DND Override is available to all extensions that have DND Override set in their Class of Service. It is also available to any extension that has a Hotline key for a co-worker, even without the Class of Service option enabled.

Automatic System Answer Button

Automatic System Answer Button
Use this feature to program a button to turn Automatic System Answer (ASA) on and off. This feature helps the operator answer calls during busy periods.
Considerations 7
■ This feature is available only on the system telephone at extension 10.
■ Program an Automatic System Answer Button on a button with lights on the system display telephone at extension 10. (This feature is not supported on a button without lights.)
■ The Automatic System Answer Button returns to the status (on/off) it was in before a power failure occurred or System Reset was used

Door phone

Communication - In Noisier Locations

E-30-PT/E-30-PT-EWP Brushed 316 Stainless Steel (shown in optional VE-5x5)
The E-30-PT is designed to provide quick and reliable communication in noisier areas. The mic sensitivity is set to a low level until the TALK button is pressed then it is raised to a normal level. In this way, the E- 30-PT assures that the called party’s voice will be broadcast over the speaker. In applications where the background noise can be loud- er than the person calling, a handset type phone is recommended.
The E-30-PT features non-volatile memory, a built in dialer, and intelli- gent call progress detection for automatic hang-up when the call is completed. The E-30-PT can be programmed to dial up to 5 different numbers on ring no answer or busy and can be configured to dial these numbers until answered.
The E-30-PT-EWP shares all of the features of the E-30-PT in addition to Enhanced Weather Protection (EWP) for outdoor installations where the unit is exposed to precipita- tion or condensation. EWP products feature foam rubber gaskets and boots, silicon sealed connections, gel-filled 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.
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Non-ADA Hot-Line Phones for:
• Terminals• Stadiums
• Parking lots/ramps• Convention centers
• ATM machines
Gate and Door Entry Phones for:
• Business lobbies
• Vehicular and pedestrian gates
• Residences
CAUTION - When installing on an analog extension of a phone system: Some phone systems do not conform to analog telecom standards and might not be compatible with the E-30-PT phones. For a detailed description of the telephone line specifications required for any of the E-30-PT phones, see DOD# 869.
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Power: Telephone line powered. Minimum 24V DC talk battery voltage, with a minimum loop current of 20mA loop (room temp), or 25mA (extended cold temp range). Loop current may be boosted on low current lines with a Viking Model TBB-1B talk battery booster (DOD# 632). Minimum Ring Voltage: 90VAC RMS Dimensions: Overall-127mm x 127mm x 57mm (5” x 5” x 2.25”), Plastic Electrical Box-102mm x 102mm x 54mm (4” x 4” x 2.12”) Shipping Weight: 1 Kg (2.2 lbs)
Operating Temperature: -26°C to 54°C (-15°F to 130°F) Humidity - E-30-PT: 5% to 95% non-condensing Humidity - E-30-PT-EWP: Up to 100% condensing Connections - E-30-PT: RJ11 jack
Connections - E-30-PT-EWP: Gel-filled butt connectors
• Vandal Resistant Features: 14 gauge louvered 316 stainless steel faceplate with permanent laser etched graphics, speaker/ mic screen, heavy duty metal keypad and “CALL” button and hex drive mounting screws
• Weather Resistant Features: Marine grade 316 stainless steel faceplate, screws and and push button switch. Switch internally sealed per IP67. Mylar speaker. Self-draining mic mount. Faceplate, mic and speaker gaskets. Weather resistant powder paint on optional VE-5x5 (DOD# 424).
• E-30-PT-EWP is designed to meet IP66 Ingress Protection Rating (see DOD# 859 for more information)
• Push to talk button
• Telephone line powered
• Non-volatile E2 memory (no batteries required)
• Programmable to dial up to 5 numbers on busy or ring no answer
• Red off-hook LED indicator
• Volume adjustments for microphone and speaker
• Advanced call progress detection: disconnects on busy signal, return to dial tone, CPC, reorder tone, maximum call time out and programmable silence time out
• Selectable auto-answer feature for monitoring
• Selectable push button disconnect
• Extended temperature range (-15°F to 130°F)
• Flush mountable using included plastic rough-in box
• Optional VE-5x5 surface mount back box (DOD# 4

Door entry Telephone

Add a Viking Entry Phone to an
Existing Phone Line
The C-200 allows single line telephones or a telephone system to share a phone line with a single Viking entry phone. Tenants may answer an entry phone call and converse with the visitor.
The C-200 provides a “Call Waiting” tone when the phone line is in use.
Tenants may call out to the entry phone for mon- itoring purposes. Auxiliary contacts are provided to operate a doorbell, or activate a camera, lights, etc.

If additional features (such as doorstrike, multiple entry phones with caller ID, and key- less entry) are needed, use the Viking model C-200

AUTOMATIC PRIVACY

AUTOMATIC PRIVACY
Description
Privacy is insured on all communications in the system. If desired, the customer may elect to disable the Automatic Privacy feature, allowing another station to join in an existing external conversation uninvited. In such a case, a conference is established.
Operation iPECS Phone To intrude into a call when Privacy is disabled
1. Press a busy (lit steady) individual {CO}/{IP} access button, user connected to the call with existing internal station user.
Conditions
1. With Automatic Privacy disabled, privacy is still assured on all intercom and conference calls.
2. To override privacy, Privacy must be disabled and the intruding station must have Override enabled as well as a direct appearance for the desired {CO}/{IP} line.
3. Only one station can intrude on an active call.
4. An intrusion tone can be provided to the call indicating another station has accessed the line.
5. If either internal party presses another {CO}/{IP}, a {DSS}, {PAGE}, [CONF] or other conflicting button, the party is removed from the “Conference” and must press the {CO}/{IP} button again to reenter the conversation.

Barge-in

Barge-In
Description
Barge-In permits an extension user to break into another extension user’s established call, including Conference calls. This sets up a Conference-type conversation between the intruding extension and the parties on the initial call. With Barge-In, an extension user can get a message through to a busy co-worker right away.
There are two Barge-In modes: Monitor Mode (Silent Monitor) and Speech Mode. With Monitor Mode, B the caller barging in can listen to another user’s conversation but cannot participate. With Speech Mode, the caller barging in can listen and join another user’s conversation.
The use of monitoring, recording, or listening devices to eavesdrop, monitor, retrieve, or record telephone conversation or other sound activities, whether or not contemporaneous with transmission, may be illegal in certain circumstances under federal or state laws. Legal advice should be sought prior to implementing any practice that monitors or records any telephone conversation. Some federal and state laws require some form of notification to all parties to a telephone conversation, such as using a beep tone or other notification methods or requiring the consent of all parties to the telephone conversation, prior to monitoring or recording the telephone conversation. Some of these laws incorporate strict penalties.
Conditions
•An extension user can barge-in on a conference.
•An extension user cannot barge-in on an Intercom call if one of the intercom callers is using Handsfree Answerback. Both Intercom parties must lift the handset or press Speaker key.
•barged into call can be placed on hold by the originator of the outside call. Both the outside caller and the extension that barged into the call are placed on hold.
•A call which is barged into can be placed on Park by the originator of the outside call, but only the outside caller is placed in Park. The extension which barged into the call is dropped.
•Privacy blocks Barge-In attempts.
•Function keys simplify the Barge-In operation.
•When Silent Monitor Mode is used, Mute key can be used to activate speech path to the internal and external parties.

T1 Lines In DSX-80/160

T1 Lines
In DSX-80/160, provides for connection to advanced digital lines and simplifes installation.
Description
T1 lines require a unique T1 PCB (P/N 80061) and give the system a maximum of 24 lines in a single PCB slot. The available T1 line types include:
• Loop Start (DTMF and Dial Pulse)
• Ground Start (DTMF and Dial Pulse)
• Direct Inward Dial (DID) Wink Start (DTMF and Dial Pulse)
• Direct Inward Dial (DID) Immediate Start (DTMF and Dial Pulse)
• E&M Tie Line Wink Start (DTMF and Dial Pulse)
• E&M Tie Line Immediate Start (DTMF and Dial Pulse)
T1 gives the system the advantages of advanced digital calling as well as conserving PCB slots. For example, you can set up a system with 12 loop start lines, six tie lines, and six DID lines and use only a single PCB slot. Additionally, the T1 PCB has its own on-board processor and DSP so it minimally impacts other system resources.
Note: Although the T1 PCB can connect directly to the telco’s T1 smart jack, your telco may require that you purchase and install a separate Channel Service Unit (CSU). This unit installs between the smart jack and the T1 PCB.
ANI/DNIS Support
The system is compatible with telco's T1 Automatic Number Identifcation (ANI) and Dialed Number Information Service (DNIS) services. ANI/DNIS services can be provided on T1 loop start, ground start, and DID lines (but not E&M). ANI/DNIS Compatibility provides:
• Selectable Receive Format
• You can set up the system for compatibility with any combination of ANI, DNIS and Dialed Number (Address) data provided by the telco.
• Flexible Routing for DID Lines
• For DID lines, the system can route the incoming call based on the received DNIS data and the entries stored in the DID Translation Table. See Direct Inward Line on page 185 for more.
• Caller ID
• The system can use the received ANI data to display the caller’s number on the called extension’s display. The ANI data can be up to 10 digits long.
FSK Caller ID
The T1 PCB can also receive FSK-based Caller ID (if provided by the telco), the same as the COIU (analog) line cards. To receive this type of Caller ID, you must enable DSP Caller ID for the T1 line circuits in programming.

Create an Auto Attendant

Create an Auto Attendant
The following process shows by example the setup for an auto attendant for Embedded Voicemail. In this example the auto-attendant should give callers the option to press 0 for reception (hunt group 200) or 1 for sales (hunt group 301).
38
· For details of routing calls to the auto attendant, see Routing Incoming Calls to an Auto Attendant .
To create an auto attendant:
1.Start IP Office Manager and load the required configuration.
2.Note that if time profiles are going to be used in an auto attendant, the time profile has to be created before creating the auto attendant. For more information, see the IP Office Manager help.
3.Click Auto Attendant. Any existing Auto Attendants are listed.
4.Click Create a New Record in the Group Pane. Select Auto Attendant .In the Name field enter the name for the auto Attendant.

ACD

Automatic Call Distribution (ACD)

Description
You can put any agent in any group. An agent can be in more than one group. This allows, for example, a Technical Service representative to answer customer service calls at lunch when many of the Customer Service representatives are unavailable.
Automatic Call Distribution (ACD) uniformly distributes calls among agents of a programmed ACD Group. When a call rings into an ACD Group, the system automatically routes the call to the agent that has been idle the longest. Automatic Call Distribution is much more sophisticated and comprehensive than Department Calling and other group services – it can accurately judge the work load at each A agent and distribute calls accordingly. The system allows up to 2 ACD Groups and 16 ACD agents.
The ACD Master Number is the extension number of the whole group. Calls directly ringing or transferred to the ACD Master number enter the group and are routed accordingly. Although the master number can be any valid extension number, you should choose a number that is out of the normal extension range.
Automatic Call Distribution operation is further enhanced by: ACD Call Queuing
When all agents in an ACD Group are unavailable, an incoming call queues and causes the Queue Status Display to occur on the ACD agent's display. The display helps the agent keep track of the traffic load in their group.
The Queue Status Displays shows:
•The number of calls queued for an available agent in the group.
•The trunk that has been waiting the longest, and how long it has been waiting.
For each ACD Group, you can set the following conditions:
•The number of trunks that can wait in queue before the Queue Status Display occurs.
•How often the time in queue portion of the display reoccurs.
ACD Overflow (With Announcements)
ACD offers extensive overflow options for another 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 customers. 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) such as, “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 agents still are unavailable, the call can automatically overflow to another ACD Group or the Voice Mail. If all agents in the overflow ACD Group are busy, Lookback Routing automatically ensures that the waiting call rings into the first agent in either group that becomes free.
You can assign an ACD Group with any combination of 1st Announcement, 2nd Announcement and overflow methods. You can have, for example, a Technical Service group that plays only the 2nd Announcement to callers and then immediately overflows to Voice Mail. At the same time, you can have a Customer Service group that plays both announcements and does not overflow.

Virtual Extensions

Virtual Extensions
Version 2.0 or higher software provides Distinctive ringing (Intercom / Outside) on Virtual Extension.
Version 2.0 or higher software, a special ring tone is provided when a pre-assigned extension places an Intercom call.
Version 3.0 or higher software, number of Ring Tone pattern is increased to 8 from 4.
With version 3.0 or higher if tone pattern 5 ~ 8 is assigned and the system is downgraded to version 2.X or lower incoming ringing will not be provided. To restore ringing, assign the tone pattern to pattern 1 ~
4.
V Version 3.0 or higher software provides the following enhancements to a virtual extension:
•A virtual extension can now display the caller ID of an internal caller (Callers station name is displayed, if station name is not available the extension number is displayed). Also, a virtual extension can now display the caller ID of an internal or external caller when the virtual is not set to ring (Previously the virtual extension must be set to ring or CID is not displayed).
•A virtual extension now has ‘One shot’ ringing which enhances the feature by allowing, either, a single burst of ringing, or normal ringing tone.
Description
Virtual Extensions are available software extensions in the SL1100. A Virtual Extension assigned to a line key, can appear and ring on an individual station or multiple stations and be used for outbound access.
Up to 50 VE keys are provided.
Conditions
•The 84 available ports/Extensions are assigned on a per extension basis for Virtual Extension key mode.
•The 84 available ports/Extensions are assigned per extension for CAR key mode or Virtual Extension key mode.
•More than one extension can share a Virtual Extension key.
•An extension can have more than one Virtual Extension key assigned.
Assigning a Virtual Extension key of the extension the key is assigned on is not supported.
•Up to 32 incoming calls can be queued to busy Virtual Extension key.
•You cannot have a CAR key and Virtual Extension on the same telephone.
•Virtual Extensions do not support the following features:
-Barge-In
-Conference
-Conference, Voice Call/Privacy Release
-Reverse Voice Over
-Tone Override
-Voice Over
•When a valid system station calls a Virtual Extension appearing on another station, Voice and MW softkeys appear in the display of the calling station, but they do not operate.
•When talking on a Virtual Extension you cannot mute the handset.
•Incoming calls to a virtual extension that appear on stations that are used with the CTI applications, PC Assistant, or PC Attendant, do not show up as a second call in the CTI application.

DHCP

DHCP Client
Description
WARNING: When the VoIPDB is installed on the CPU, the built in LAN port on the CPU becomes disabled. Only the LAN Port on the VoIPDB will be operational.
DHCP Client will access an external DHCP server every time the LAN cable is connected to the CPU/ VoIPDB or when the System is powered up. The System can receive the following information from the DHCP server:
IP Address, Subnet Mask, and Default Gateway.
Conditions
•The DHCP Server should be configured to provide the system the same IP address every time. For example in the DHCP server extend the lease time to infinite or setup the server to provide the same IP address based on the systems MAC Address.
•When changing PRG 10-63-01 (DHCP Client Enable/Disable) a system reset is required for this change to become effective.
•DHCP client can set following programs automatically; however other IP related programs (such as PRG 84-26) have to set manually as required.
-IP Addresses: PRG 10-12-01 (CPU), PRG 10-12-09 (VOIPDB)
-Subnet Masks: PRG 10-12-02 (CPU), PRG 10-12-10 (VOIPDB)
-Default Gateway: PRG 10-12-03
•DHCP Client (PRG 10-63) and existing DHCP Server feature (PRG 10-13) can not be used at the same time.
•While the System accesses the DHCP Server, to receive IP Addressing information, the CPU RUN LED flashes as follows. If the System fails to receive an IP Address from the DHCP server the system will use the IP Address assigned in PRG 10-12.

300 ms
On Off
300
ms 300
ms
200
ms 200
ms 700
ms
•Once after IP Address and Subnet Mask are set, if different IP Addresses or Subnet Mask is delivered during normal operation mode, both LED2 (Red lit) and RUN LED (flash as above) indicate system requires reset.

Tone Service

Howler Tone Service
Description
Howler Tone Service provides a Howler Tone when a station remains off-hook after a call is completed or when a station is off-hook and digits are not dialed in a programmed time.
Conditions
Howler tone is generated 30 seconds after a call is disconnected and the telephone is left off-hook or the telephone is left off-hook without dialin