FXO 0/0 Pinouts Pin

FXO 0/0 Pinouts Pin   1,2 3 4 5,6   Name — Ring Tip — Description Unused Ring lead of  the  2-wire interface Tip lead  of the 2-wire  interface Unused Table A-3.   NET 1 through NET 4  (T1 0/1 through T1 0/4)  Pinouts Pin   1 2   Name R1 T1 Description Receive  data  from  the  network (Ring  1) Receive  data  from the network (Tip  1) 3 4 5 6-8 — R T — Unused Transmit data  toward  the  network (Ring) Transmit data  toward  the  network (Tip) Unused Table A-4.   10/100BaseT (ETH 0/1 through ETH 0/2)  Pinouts Pin  1 2 3 4, 5 6 7, 8    Name TX1 TX2 RX1 — RX2 — Description Transmit Positive Transmit Negative Receive Positive Unused Receive Negative Unused

Queue Status

When all agents in an  ACD Group are unavailable, an incoming call will queue and cause the Queue Status Display to occur on the  ACD Group Supervisor and/or agent’s display.  The display helps the supervisor keep track of the traffic load within their group. In addition, any display keyset can have  a Queue Status Display Check programmable function key.  The keyset user can press this key any time while idle, and using the VOL   and VOL  , scroll through the Queue Status Displays of all the  ACD Groups.  The Queue Status Displays shows (see the Queue Status Display illustration below): 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. The number of calls in queue. 2  LINE-001   01:30 Name of trunk that has been queued the longest. How long the longest queued call 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 (see the Queue Status display Timing illustration below). Queue Status Display holding time. Queue Status Alarm enable/disable. Queue Status  Alarm sending time.

U-NT and U-LT loops

U-NT and U-LT loops can be used in combination to provide D-packet service for a point-of-sale terminal adapter (POSTA) or other D-packet device. D-packet service is a 16 kbps data transmission service that uses  the D-channel of an ISDN line. To deliver D-packet service, a network connection (U-NT) is programmed to work with a terminal connection (U-LT). The loops must be on the same physical card. For example, if the network connection is a loop found  on the BRI Card in Slot 1, the terminal connection must be  a loop found on the same card. S reference point The S reference point connection provides either a point-topoint or point-to-multipoint  digital connection between Norstar and ISDN terminal  equipment (TE) that uses an S interface. S loops support up to seven ISDN  DNs, which identify TE to the ICS.

Networking Norstar

Networking with Norstar There are a number of ways you  can network Norstar systems together, or network Norstar  systems with other Nortel systems into private networks. What types of lines you use to perform the networking will determine the type of services that can be  shared between systems. Keep in mind that each node (Norstar system) is considered an external system by every other  node within the network, even though, to the users, it appears to be all one system. This affects how you configure call transfer and call out  features on each  system. On the home node, all  features are configured as local numbers. On all other nodes,  all features directed to the home node are configured with external numbers. As well, each node must have a  unique identifying code. What this code will be, and how it is  configured for the user, depends on what type of trunks  and dialing rules you choose to use. If the network has a Meridian as part of the network, the Meridian administrator will determine identification codes for the systems. This section describes various  configurations of private networks. The general settings  that are required to set up the home node for each system are provided to give you a sense of what is required for each  type of network. The common goal is to provide the  user with the sense that the network is one large system that provides common access to colleagues in other buildings, cities, or countries. In some systems they may need to enter  a destination code before the local number to route the call to  the correct system. In other systems, using a common dialing  plan allows users to dial colleagues at any location simply by entering the same number of digits they would use to dial  a colleague at the next desk.

SIE keys for ACD Groups

Any Multiline Terminal can have SIE keys for ACD Groups. When a call  comes into a covered ACD Group, the SIE key will ring  immediately, ring  after a delay or just flash (depending  on system programming and  user-set options). The  Multiline Terminal user  can answer the call  by  just lifting  the handset and pressing the  SIE key. ACD SIE keys help maximize ACD service during high traffic periods or when agents are unavailable. The  covering  extension  does not  have to  be  a member of  the  ACD Group. In  addition, an extension  can have  SIE keys for  as  many ACD Groups as  it  has  available programmable keys. An ACD Group  SIE  key also allows for one-button Transfer  to an ACD Group. Conditions Ringing  for SIE keys may need  to be  programmed through the telephone

DISA

Place call  to  DISA facility of the system. 2.  At receipt of Intercom dial tone/AA  announcement, dial as  desired.   If  DND Warning tone is received, enter an Authorization Code, to receive Intercom dial tone. Conditions 1.  Each CO/IP path is separately assigned for DISA operation during Day, Night and/or Timed system operation mode.  DISA operation is active  only when the system  is in the  assigned operating mode(s). 2.  DISA callers can be routed to a  VMIM/VSF  Auto Attendant announcement in place of Intercom dial tone.  The  announcement can be associated with a CCR  Table or assigned to disconnect  after playback (‘#’). 3.  A DISA caller can be required to enter an Authorization Code  to access  the system’s  external outgoing resources, facilities or features.  If required, the caller is permitted to retry entry of a valid Authorization Code based on  the DISA  Retry count.  Continued failure results in disconnect. 4.  DISA callers are subject  to COS dialing restrictions.  If Authorization Codes are required and the code entered matches a Station Authorization Code, the station’s COS  will apply.   Otherwise, the assigned DISA COS will apply.  In both cases, the  CO/IP  COS for the outgoing path will be active. 5.  The system will disconnect an outgoing DISA call if the  Unsupervised Conference timer expires or disconnect supervision is received.   A disconnect warning tone is provided 15 seconds prior to disconnect. 6.  If a DISA caller encounters a system All Lines Busy, busy tone is received for 5 seconds before ICM  Dial tone is  again presented and the  DISA caller may try another call. 7.  LEDs associated with the DISA CO  Line appearance will provide normal status indications at all stations  except the Attendants.   The LED for the line at  an Attendant will flutter  at 240 ipm when busy. 8.  An iPECS  Phone user can only receive a DISA call with an available CO/IP appearance button.

DISA

Place call  to  DISA facility of the system. 2.  At receipt of Intercom dial tone/AA  announcement, dial as  desired.   If  DND Warning tone is received, enter an Authorization Code, to receive Intercom dial tone. Conditions 1.  Each CO/IP path is separately assigned for DISA operation during Day, Night and/or Timed system operation mode.  DISA operation is active  only when the system  is in the  assigned operating mode(s). 2.  DISA callers can be routed to a  VMIM/VSF  Auto Attendant announcement in place of Intercom dial tone.  The  announcement can be associated with a CCR  Table or assigned to disconnect  after playback (‘#’). 3.  A DISA caller can be required to enter an Authorization Code  to access  the system’s  external outgoing resources, facilities or features.  If required, the caller is permitted to retry entry of a valid Authorization Code based on  the DISA  Retry count.  Continued failure results in disconnect. 4.  DISA callers are subject  to COS dialing restrictions.  If Authorization Codes are required and the code entered matches a Station Authorization Code, the station’s COS  will apply.   Otherwise, the assigned DISA COS will apply.  In both cases, the  CO/IP  COS for the outgoing path will be active. 5.  The system will disconnect an outgoing DISA call if the  Unsupervised Conference timer expires or disconnect supervision is received.   A disconnect warning tone is provided 15 seconds prior to disconnect. 6.  If a DISA caller encounters a system All Lines Busy, busy tone is received for 5 seconds before ICM  Dial tone is  again presented and the  DISA caller may try another call. 7.  LEDs associated with the DISA CO  Line appearance will provide normal status indications at all stations  except the Attendants.   The LED for the line at  an Attendant will flutter  at 240 ipm when busy. 8.  An iPECS  Phone user can only receive a DISA call with an available CO/IP appearance button.

CIX License Control

CIX License Control The  system  size  and feature capability is controlled using  a  software License Key Code. This key code  is obtained from  the Toshiba Internet FYI site during  the  ordering  process and is installed onto the system processor via CIX  eManager. Processor license codes activate system  hardware capacities. Additional sets  of four CO line/digital station ports beyond the  Basic bundled number of ports requires  one LIC-4 BASIC license. See table below. CIX System Processor Basic Bundled  Port Licenses Maximum Ports CIX670 CIX200 CIX100 CIX100-S CIX40 BCTU2A LCTU1A ACTU3A ACTU3A-S GCTU2A 64 32 32 16 192 or  6721 192 112 1122 LIC-4 BASIC license not  required 1. The  BEXU2A  sub-assembly can be added  to expand capacity from  192 to 672 ports. 2. The  upgrade  from 16  to 24  ports and  from 24  to  32  ports requires the  eight port upgrade LIC100S-8 PORTS license. Each  additional  set of 4  line/station  ports requires the  four port upgrade LIC-4  BASIC license  (maximum of 112  ports). DTMF  tone  receiver circuits are required for standard  telephones, Voice  Mail  DTMF integration, Tie,  DID and  DNIS  line service. 16 DTMF  built-in receiver hardware circuits  and 16  ABR circuits – The first  four DTMF circuits and all ABR circuits do not require a license. Each  additional  set  of four DTMF  receiver circuits require one  LIC-4 DTMF  license  (maximum of  16  DTMF circuits). IP  End Point licenses are  required  for IPT,  SIP and SoftIPT phones. The  optional  RS-232 serial port interface (BSIS) provides two circuits  to interface with  SMDI or Toshiba Proprietary  Voice Mail  integration,  Call  Accounting SMDR, and  two for future applications. The  first circuit  does not require a license, but circuits two through four each require one LIC-SER PORT  license. Refer to  the  “Strata CIX Software License Requirements”  on  page 177  for license part  numbers and hardware configurations.

PSTN switching

To assure that the PSTN switching  equipment has sufficient time to restore to the idle condition, the system will hold analog CO lines in a busy state to  users  after release  of a CO line  by a station.   The time between the station disconnect and  when the system changes the  CO line status  from busy to idle is the CO Line Release Guard time. Operation System Operation of this feature  is automatic. Conditions Programming  SYSTEM Related Features

System Administration

Launching the System  Administration Application 1. 2. 3. 4. To start the System Administration application, perform one of the following steps: Double-click  on the MERLIN  Messaging  Administration  desktop  short cut or the  PARTNER  Messaging Administration desktop short cut. Select the MERLIN Messaging Administration  short cut or the PARTNER Messaging Administration short cut from the Start menu. From the Start menu, select  Programs > MERLIN Messaging  Release 4.0  or  PARTNER Messaging Release 7.0>System Administration. (This is the default location.) The Messaging System Administration  window appears, displaying the Messaging Login dialog box. In the IP address or host name box, enter the  Fixed IP address of the messaging  system module that was loaded into the message system. (See Section 2.) In the Login box, enter  sysadmin. In the Password box, enter  the system  administration password. (If this  is your  first time logging into the system,  click the  OK  button. You are prompted to enter the password.) 5. Click  the  OK  button. Once you log  in successfully, you  can  start administering the messaging  system. The System  Administration application windows display the current settings  for  the messaging  system. For more information regarding installation  and use of the  System Administration  application, see the MERLIN  Messaging System Administration  Getting Started Guide or  the PARTNER  Messaging  System Administration  Getting Started Guide under Documentation, System Administration  contained on the Library CD

SIE Key for ACD Groups

SIE Key for ACD Groups Description Any Multiline Terminal can have SIE keys for ACD Groups. When a call  comes into a covered ACD Group, the SIE key will ring  immediately, ring  after a delay or just flash (depending  on system programming and  user-set options). The  Multiline Terminal user  can answer the call  by  just lifting  the handset and pressing the  SIE key. ACD SIE keys help maximize ACD service during high traffic periods or when agents are unavailable. The  covering  extension  does not  have to  be  a member of  the  ACD Group. In  addition, an extension  can have  SIE keys for  as  many ACD Groups as  it  has  available programmable keys. An ACD Group  SIE  key also allows for one-button Transfer  to an ACD Group.

Linked stations

Linked-Pair Station Description A station can be logically linked to a primary station so that the two stations function  as a single station.  When linked,  the two stations effectively act as  a single  station with  the station attributes of  the primary  station.    The status of one station is reflected in  the status of  the other and features activated at one are active at the other.    All  internal or external calls to a linked pair station will ring both stations. All features available to the primary station are available  and controllable by the  secondary station, one  station  may activate Call Forward  and the other may cancel the  forward.    The displays of  the linked stations will  display the  status of  the linked  pair.    If one  of  the linked stations is  busy, the LCD of the other station  will display “IN USE AT  LINK STA”.    When a linked  station is busy, the other idle  linked  station will not  receive ring for CO lines, transferred ring or intercom calls. Consideration   A station can be linked  with only one station.    Any combination of DKTs and SLTs may be assigned  as Linked pairs.   A DSS Console, Door Phone Box  or Port Blocked Station may not  be assigned  as  a linked pair  station.   Linked pair  stations are  treated as having a single station number for all features including LCD displays, station programming,  ADMIN access, ACD statistics, SMDR, etc.   Intercom calls to the Linked stations always signal in the Tone ring mode and cannot be changed using the Forced Hands-free feature.   The station attributes of  the secondary station  will follow attributes of the primary  station, i.e. Day/Night COS, CO Warning Tone, CO Auto Hold, CO Call Drop, DID Call Waiting, Speed Access, Alarm, VMIB Access, DND, FWD,  Paging, CO Access, CO Ring, etc.   An Attendant station  can  be linked with another  station but, the linked station cannot  use attendant features (refer  to Ref. D).   Calls can be  placed or transferred between the stations of  a  Linked pair  using the primary station number.

ARS time

NetworkOutgoingInter-DigitARSTimeWithNetworking,thistimereplaces20-03-04whendeterminingifallnet-workprotocoldigitshavebeenre-ceived.IfARSisenabledatSiteB,thistimecanbeprogrammedfor5(500ms)atSiteA.IfARSisdisabledandSiteBisusingF-Routeforout-bounddialing,thistimeshouldbeprogrammedfor30(threeseconds)atSiteA.

Transferring Calls

Transferring Calls to an Attendant Normally calls are directed to an auto attendant by an IP Office incoming call route. However it can also be useful to transfer calls received at an internal extension to an auto attendant. You can transfer calls to an Auto Attendant by: · Using Programmed Buttons · Using Phone Manager · Using SoftConsole · Using Short Codes 40. 40 39.. 39. Using Programmed Buttons On Avaya phones with programmable buttons, those buttons can be programmed to access auto attendant services. To create an auto attendant button: 1.From the IP Office system configuration, set the action of one of the users programmable buttons to  Dial. 2.Set the associated telephone number to  AA:Name  where  Name  matches the name of the auto attendant. 3.Save this configuration change back to the IP Office. When the user receives a call they want to transfer to the auto-attendant, they can use a programmed button. To transfer a call using the programmed button: 1.Place the call on hold. 2.Press the button programmed for the auto-attendant. 3.Hang-up the call at their extension. This will cause a blind transfer of the held call to the auto-attendant. Using Phone Manager To create an auto attendant speed dial: 1.Within the user's Phone Manager, click the  Speed Dials  tab. 2.Right-click the  speed dial panel and select  New > Speed Dial Group Member. The Speed Dial window opens. 3.In the  Name  field, enter a name for the Auto Attendant. 4.In the  Number  field, enter  AA:Name  where  Name  matches the name of the auto attendant. 5.Click OK. To transfer a call using the Speed Dial: 1.During a call that you want to transfer to the auto attendant click  Hold to place the call on hold. 2.Click the Speed Dials tab. 3.Click the speed dial created for the auto attendant. 4.Click  Complete Transfer to transfer the held caller.

Programmable Buttons

Line/Programmable  Buttons. Used  for  individual  outside lines  or  (if no  line  is  assigned  on  a button)  for  programming telephone or  extension numbers,  or  system  features  (such as  Last Number Redial).  When a line  is  assigned,  press  the line  button to make  a call  on that  specific line  (lights  show status  of line).  When  a number  feature is  programmed,  press  the  button to dial the number  or  use the feature.  The  PARTNER-34D has 36  programmable buttons  (32 with lights and 4  without lights);  the PARTNER-18D has  20 programmable buttons  (16 with  lights and  4 without lights);  the  PARTNER-18  has  16  programmable  buttons  (all  with  lights); the PARTNER-6 has  4 programmable buttons  (all  with  lights). ■ Fixed Buttons.  In  addition  to  the line  buttons, the  telephones  have  some  or  all  of  the following fixed  buttons, which are  already  imprinted: — Intercom  Buttons.  Press  to make  (or  answer)  a  call  to (or  from)  another  extension in  the system. If you  receive a  call  on  a T1  line  with Direct  Inward  Dialing  (DID),  and you  cannot  access that  line  from the line  or  pool  buttons  on  your  telephone, the call  will  appear  on  your Intercom  button.  Press  the  button to  answer  this  outside call. — Feature.  Press  to  change programmed settings or  use system  features. — Conf (Conference).  Press  to add other  parties to  your  call. — Transfr (Transfer).  Press  to pass  a  call  to  another  extension. — Hold. Press  to  put  a call  on  hold. — Spkr (Speaker).  Press  to  turn  on  and off the speaker  and  microphone (if  available), so you can  dial  and have  a  conversation  without lifting  the  handset. The  light  next to  this button  shows  when  the  speaker  is  turned  on. — Mic/HFAI.  Press  to  turn  the microphone on  and off.  The light next  to this  button  shows when  the  microphone is  turned  on.  Leave  on  to use  Hands-Free Answer  on  Intercom (HFAI)  feature. — Volume Control  Buttons.  Press  – to  decrease  or  +  to increase  the volume  as  follows: ■ To adjust  ringer  volume, press  while  the  telephone  is  idle  and  the  handset is  in the cradle. ■ To adjust  speaker  volume, press  while listening  to a  call  through  the  speaker. ■ To adjust  handset  volume,  press while listening  through  the handset. ■ To adjust  background  music  volume, press  while  listening to  music  through the telephone’s  speaker.

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.

Feature name

SL1100 Feature NameV1.0V1.5V2.0V3.0Loop KeysSSSSMaintenanceSSSEMeet Me ConferenceSSSSMeet Me PagingSSSSMeet Me Paging TransferSSSSMemo DialSSSSMessage WaitingSSSSMicrophone CutoffSSSSMobile ExtensionSSSSMobile Extension - Callback to Mobile PhoneSSSSMultiple Trunk TypesSSSSMusic on HoldSSSSName StoringSSSSNavigation KeySSESNight ServiceSSSSOff-Hook SignalingSSSSOne-Touch CallingSSSSOperatorSSSSPaging, ExternalSSSSPaging, External (VRS)SSSSPaging, InternalSSSSParkSSSSPBX Compatibility/Behind PBXSSSSPC ProgrammingSSSEPC Programming - Intuition SetupSSSEPower Failure TransferSSSSPrime Line SelectionSSSSPrivate LineSSSSProgrammable Function KeysSSSSProgramming from a Multiline TerminalSSSSPulse to Tone ConversionSSSSRedial FunctionSSSSRemote (System) UpgradeSSSSRepeat RedialSSSSResident System ProgramSSSSReverse Voice OverSSSSRing GroupsSSSSRingdown Extension (Hotline), Internal/ExternalSSSSRoom MonitorSSSSSave Number DialedSSSSSecondary Incoming ExtensionSSSSSecretary Call (Buzzer)SSSSSecretary Call Pickup

full power of iPECS

Unleash the full power of iPECS, and experience the next generation in business communications technology.  With a full range of powerful software applications, the true potential of your iPECS voice platforms can be realized. As your business grows and your technology needs become more sophisticated, Ericsson applications leverage your current iPECS investment.  Applications deliver features that empower your employees to be more productive, more mobile and more collaborative. They can also enhance your business’ ability to deliver a more responsive and superior customer service experience beyond traditional voice-only customer contact.  Whether your business is faced with the continued adoption and resulting challenges  of “bring-your-own-device” (BYOD), increased need for Unified Communications (UC) or the complexity of managing a geographically dispersed and disparate voice and data network, iPECS applications