Administrator password

11   Initialize 12   Installer  password 13   Administrator  password 14   System clock 141   Set  time/date 142   Automatic  time  setting 143   Clock  adjustment 15   System  timing  parameters 151   Flash duration 152  Transfer  forward  timer 153   Recall  timers 1531   Exclusive hold 1532   System  hold 1533   Hold  recall  rings 154   ACD timers 1541   ACD exit  timer 1542   ACD wrap  timer 1543   ACD  hold recall  timer 155   ACD wrap  timer 156   Cell  phone delay 157   Device  timers 158   VIP Attendant  exit  timer 16   System  feature  parameters 161   Recording  alert  tone 162   Connect  tone 163   Station feature  set activation 164   Esi-Link  location  no./line  group  access  selection 165   Auto  attendant parameters 166   CO  line  parameters 167   Voice  mail parameters 169   Feature  set activation 17   System speed-dial 18   Maintenance/SMDR  serial  port 2  CO  line programming 21   Line  programming 211   Analog  CO  line  programming 212   T1  programming 2121   CO line  programming 2122   T1  frame format  and  line  coding 2123   Line  build-out 2124   CSU  emulation   213   PRI  programming 2131   CO line  programming 2132   Line  build-out 2133   CSU  emulation 2134   Switch  protocol 2135   DID 214   SIP  trunk  programming 2141   SIP  trunk  programming  day/night  mode 2142   SIP  account programming 2145   SIP  pilot table programming 22   Translation  table  programming 221   Centrex/PBX  access  code   222  Toll  restriction  exception  tables   223   ARS  (Automatic  Route  Selection) 224   DID and DNIS/ANI  translation table 225   PRI  pilot number  translation table 226   Local  allow  table 23  Line  parameters 231   Line  receive  volume 232   Analog  line disconnect 233   T1  line  receive  volume 234   PRI  line  receive  volume 24   Caller  ID  programming 3   Extension programming 31   Extension definition  and  routing 32   Extension feature authorization 321   Standard  feature  authorization 322   Advanced  feature  authorization 33   Department  programming 331   Department  definition  and routing 332   VIP ACD  parameters 34   Dial  plan  assignment 341   Flexible  number  assignment 342   Network  numbering 35   Extension button  mapping 37   ESI device  programming 371   Access  schedules 372   RFID  tag programming 373   View RFID  tag numbers 374   ESI  Presence  Management  parameters 375   ESI  Presence  Management  Reader  parameters 30   Station  move1 4   Auto  attendant  programming 41   Auto attendant  branch  programming 42   Announce  extension number 43   Automatic  day/night  mode  table 5  Voice mail programming 51   Maximum message/recording length 52   Message purge control 53   Guest/info mailboxes 54   Group  mailboxes 55   Message notification  options 551   Station delivery  options 552   Delivery/paging  parameters 56   Cascade notification  mailboxes 57   Q  &  A  mailboxes 58   Move and  delete  messages 6  Recording 61   Record  system  prompts 62   Record  directory  names 63   MOH programming 631   MOH source 632   Record  MOH   633   MOH volume 7   Reports 71   System  reports 711  Programming  report 712   Diagnostic  reports 72   ESI  Presence Management  access  door  report 73   ACD department  detail  report 74   Voice  mail  statistics  report 75   System speed-dial  list 76   NDDS  report 8   IP  programming 81   Display  licenses 82   Local  programming 821   IP  programming 822   Local  phone starting  address 824   Network  Services  Processor 83   Esi-Link  programming 831   Local  location  number 832   Esi-Link  location  programming 833   Delete  Esi-Link  location 834   Esi-Link  publish  list programming 835   Compression  algorithm 84   ESI  SIP Card  programming 85   ESI ASC  programming 86  ESI Mobile Messaging selection

Cordless handset

Installing ESI’s Cordless Handsets Note:  Except  where noted,  this  section applies  to  both the  ESI  Cordless  Handset  II  and original ESI  Cordless Handset  models. Each ESI  Cordless Handset  comes with: • A charger/cradle  to  charge the  Handset. • An  AC  adapter  for  use  with  only  the  charger. • A base station  to  provide  a  digital  interface between  the  ESI  phone system  and  ESI  Cordless Handset.  This  base  station  needs  no  AC  power: if the  Cordless  Handset  is  digital,  the  base  station is  line-powered;  if the  Cordless  Handset is  IP,  the  base  station  uses  Power  over  Ethernet  (PoE). •  Wall-mount(s),  a  belt clip,  and a  Quick Reference  Guide. Each  ESI  Cordless  Handset  is  keyed to  only  one  base  station  and takes  up  one  port  (digital  or  IP)  on  a  port  card. Base station  installation Due  to  each  site’s  unique  characteristics,  the  range  and  distance  information  we’ll provide  herein  is   only approximate. Characteristics that  positively  affect  performance: •  The base  station  should  be  installed so  it  has a  clear  line-of-sight  with  the Cordless  Handset. •  The  base  station antenna should  always  be  pointed  in  its uppermost  vertical  position. Characteristics that  negatively  affect  performance: •  Large amounts  of  metal  shelving  (such as  in  manufacturing or  warehouse areas). •  Close  proximity to  (within  one mile  of)  a radio tower. •  Concrete  walls that  divide  spaces  where  Cordless  Handsets  are used  (assuming the base  stations  are in one location). Don’t  install  the  base  station: •  Close  to  a wall  with  metal  studs. •  On  a  metal wall. •  Next  to  a  device  that  emits  RFI  or  EMI1  —  e.g.,  a television,  radio,  computer,  computer  printer,   fluorescent  light  fixture,  or  fax machine. •  Next  to  any other  900 MHz  device  —  e.g.,  a  hand-held inventory control  device. •  In  a  ceiling  that  has  foil-backed  insulation. •  Behind  doors that  typically are closed,  tinted  windows,  one-way glass,  or  other  areas  that  limit  or   cut  off transmission  to  the  Cordless  Handset. Base  stations must  be  installed  at  least  10  feet  apart,  regardless  of  whether  the base  station is  for  the  small-model  or  large-model  Cordless  Handset.  Don’t  install  more  than  six base  stations  in  one  area   (such  as  a  network  room).  Choose  a location  at  least  30  feet  away  if  more  than  six  base  stations  are needed in a building.

External paging device

External paging device connection ESI-1000 and  ESI-600, and  ESI-50 On  either  of  these systems,  a dry-contact  overhead-paging device can be connected through the RJ-11  OH Paging  connector,  which is  located  on the front  of  the main  board faceplate just  below the  NSP’s Ethernet  connector.  Although this  is  a  six-pin connector,  only two  pairs  are needed  between  the paging  device  and the connector: •  To  pin-out  the  connector  for  normally  open  operation,  connect  the  audio  wires  to  pins  3  and 4  and the control  pair  to  pins 1  and 2. •  To  pin-out  the  connector  for  normally  closed  operation,  connect  the  audio  wires  to  pins  3  and 4  and  the control  pair  to  pins 5  and 6. ESI-200 and  ESI-100 A dry contact  overhead-paging  device  can  be  connected to  the  system  through the first  port  card's  66  block.1 The overhead  paging  port  is  fixed (located  on the  main  board)  as  code  599 for  programming  purposes  and user  access.

System Programming

System  programming overview 1  System  parameters 11   Initialize 12   Installer  password 13   Administrator  password 14   System clock 141   Set  time/date 142   Automatic  time  setting 143   Clock  adjustment 15   System  timing  parameters 151   Flash duration 152  Transfer  forward  timer 153   Recall  timers 1531   Exclusive hold 1532   System  hold 1533   Hold  recall  rings 154   ACD timers 1541   ACD exit  timer 1542   ACD wrap  timer 1543   ACD  hold recall  timer 155   ACD wrap  timer 156   Cell  phone delay 157   Device  timers 158   VIP Attendant  exit  timer 16   System  feature  parameters 161   Recording  alert  tone 162   Connect  tone 163   Station feature  set activation 164   Esi-Link  location  no./line  group  access  selection 165   Auto  attendant parameters 166   CO  line  parameters 167   Voice  mail parameters 169   Feature  set activation 17   System speed-dial 18   Maintenance/SMDR  serial  port 2  CO  line programming 21   Line  programming 211   Analog  CO  line  programming 212   T1  programming 2121   CO line  programming 2122   T1  frame format  and  line  coding 2123   Line  build-out 2124   CSU  emulation   213   PRI  programming 2131   CO line  programming 2132   Line  build-out 2133   CSU  emulation 2134   Switch  protocol 2135   DID 214   SIP  trunk  programming 2141   SIP  trunk  programming  day/night  mode 2142   SIP  account programming 2145   SIP  pilot table programming 22   Translation  table  programming 221   Centrex/PBX  access  code   222  Toll  restriction  exception  tables   223   ARS  (Automatic  Route  Selection) 224   DID and DNIS/ANI  translation table 225   PRI  pilot number  translation table 226   Local  allow  table 23  Line  parameters 231   Line  receive  volume 232   Analog  line disconnect 233   T1  line  receive  volume 234   PRI  line  receive  volume 24   Caller  ID  programming 3   Extension programming 31   Extension definition  and  routing 32   Extension feature authorization 321   Standard  feature  authorization 322   Advanced  feature  authorization 33   Department  programming 331   Department  definition  and routing 332   VIP ACD  parameters 34   Dial  plan  assignment 341   Flexible  number  assignment 342   Network  numbering 35   Extension button  mapping 37   ESI device  programming 371   Access  schedules 372   RFID  tag programming 373   View RFID  tag numbers 374   ESI  Presence  Management  parameters 375   ESI  Presence  Management  Reader  parameters 30   Station  move1 4   Auto  attendant  programming 41   Auto attendant  branch  programming 42   Announce  extension number 43   Automatic  day/night  mode  table 5  Voice mail programming 51   Maximum message/recording length 52   Message purge control 53   Guest/info mailboxes 54   Group  mailboxes 55   Message notification  options 551   Station delivery  options 552   Delivery/paging  parameters 56   Cascade notification  mailboxes 57   Q  &  A  mailboxes 58   Move and  delete  messages 6  Recording 61   Record  system  prompts 62   Record  directory  names 63   MOH programming 631   MOH source 632   Record  MOH   633   MOH volume 7   Reports 71   System  reports 711  Programming  report 712   Diagnostic  reports 72   ESI  Presence Management  access  door  report 73   ACD department  detail  report 74   Voice  mail  statistics  report 75   System speed-dial  list 76   NDDS  report 8   IP  programming 81   Display  licenses 82   Local  programming 821   IP  programming 822   Local  phone starting  address 824   Network  Services  Processor 83   Esi-Link  programming 831   Local  location  number 832   Esi-Link  location  programming 833   Delete  Esi-Link  location 834   Esi-Link  publish  list programming 835   Compression  algorithm 84   ESI  SIP Card  programming 85   ESI ASC  programming 86  ESI Mobile Messaging selection 1Shown in  the same  order  as  it  appears  in  the programming  menu on  an ESI desktop Feature Phone. D.5

D channel (Data channel)

D channel (Data channel):  An ISDN  standard transmission channel which  is packet-switched, and is  used for call setup, signalling  and data transmission. Data channel:  See  D  channel. Data Communications  Interface (DCI):  A Norstar device  that  allows  you to attach an RS-232  data  device to the Norstar system. data terminal:  A device, such as a modem, that can be used to  transfer data instead of sound  over a telephone network. You cannot use Norstar programming to set up such devices. See the  documentation that accompanies  the  device. date:  See Show  Time or  Time and Date. defaults:  The  settings for  all  Norstar features when the  system  is  first installed.  Settings  are changed from their defaults in programming. In  this manual,  default settings are  shown in bold  text. Delayed  Ring Transfer (DRT) to prime:  After a specified number of rings,  this  feature  transfers  an unanswered call  on an external line, to the  prime telephone associated with that line.  This feature  is  activated  under Feature settings  in  Sys prgrmmng. destination code:  A two- to  12-digit number that  the system interprets and then  translates  into the digits that  you want dialed  out.  Both  the code  and its associated  dialed digits  are  assigned under  Routing  service in Services programming. DID trunk:  See Direct  Inward Dial trunks. DID Trunk  Cartridge:  The Trunk Cartridge that allows you to connect DID trunks to the  Norstar  system. dialing restriction:  See Restriction filter. dialing modes:  ≤•°¤ This feature  allows you  to set the  dialing mode  of your telephone.  Norstar supports three dialing  modes: Automatic Dial,  Pre-Dial, and Standard Dial. All  three  modes support on-hook dialing, meaning you  can dial a call without picking up the  receiver. The special features  of  the  Automatic  and Pre-Dial  modes are available  only when you dial on-hook. Digital Mobility phones  74XX: These telephones connect  to the system through station modules  connected to a Nortel Networks  Digital  Mobility controller. Digital  Trunk  Interface:  The Trunk Cartridge connects  digital T1 AND ISDN trunks to the  Norstar system. Direct-dial:  A  feature that  allows  you to dial a designated telephone in your Norstar system  with  a  single  digit,  such as  the main  receptionist. As many  as five  direct dial sets  can be established. Each  telephone in  the  system is assigned to one  direct-dial  telephone. There is a single,  system  wide  digit for calling  the assigned direct-dial  telephone of  any telephone.  Direct-dial telephones are established in System programming. Telephones are  assigned  to a direct-dial telephone under  Capabilities  in Terminals&Sets  programming. Direct-dial #:  A digit used  system- wide to call  the  Direct-dial  telephone. The digit is  assigned  under  Access codes  in  Sys  prgrmmng. Direct-dial number:  The digit  used to call  the direct-dial telephone.

Mirroring operation

Mirroring operation On system  power-up  —  e.g.,  at  initial  installation or  whenever  a  drive  is  replaced  —  the  M3  will  first  verify  that each  drive is  an  ESI-formatted  drive.  If  so,  it  then  will  transfer  all  data  from  the primary  drive to  the mirroring drive.  This  process  can take  anywhere from  a few minutes to  one  hour,  depending  on  system  activity,  amount of  voice message  storage,  and  configuration.  System  operation  won’t  be affected during  the  data  transfer, because  this  transfer  will  occur only  when  call-processing  is  making  no  disk  drive  access  requests. If a  primary  drive  is  replaced, data  will be  copied  in  the  same  fashion  from  the  mirror  drive  to  the  new  primary drive.  Again,  the replacement  drive must  be  a new,  unprogrammed ESI  drive.  If it’s  not, the  system  may copy  all  data in  the wrong  direction  —  i.e.,  from  the  new (mostly empty)  primary  drive to  the mirror  drive! Therefore,  ESI  recommends that  the  mirror  drive be  moved  to  the  primary  drive  mounting position and  the new drive be  mounted on  the  mirror  drive mounting position.

Doorphone

Basic features Doorphone  —  A call  can be  initiated  to  a  pre-assigned destination (an  extension,  department,  mailbox,  auto-attendant branch  ID,  or  Esi-Link  location) by  pressing  the  CALL  button on  the ESI  Presence  Management  RFID  Reader. Manual door  lock control  (remote door  unlock)  —  Where  an  ESI  Presence  Management  RFID  Reader  is  installed, a call  can be  made  to  a  station,  department,  mailbox,  auto-attendant  branch ID,  or  Esi-Link  location  by  pressing  the CALL  button.  If  a user  answers that  call,  the  user  can then  use  the remote  door  unlock  feature  key to  unlock the associated  door.  The user  remains  connected to  the doorphone  when  using the remote  door  lock  control  feature.  The CALL button destination can be different for day and night modes. Important:  Only  ESI  desktop phones can be used to  remotely unlock  the door. The code for  the remote  door  unlock  feature  key  is  5 8 1. RFID features Automatic door  control  —  Electronic  keys  are allowed,  or  restricted  from,  activation  of  electric door  strike  locks via  programming.  Each  electronic  key  that’s  allowed  door  control  is  assigned  the extension  number(s)  (or  “ALL” for  all Readers)  of  the RFID  Readers  for  which it’s  allowed to  control  door  locks.   Authentication confirmation  —  When  an RFID  Reader  scans  an  electronic  key,  and the  system  validates or  rejects the  electronic  key for  door  control,  the  RFID  Reader  will  display  a message  and  play  a tone  or  optional  greeting.  Valid electronic  keys receive the  message  WELCOME  or  GOODBYE.  Rejected  electronic  keys  receive the message ACCESS DENIED. Access schedules  —  Each  electronic  key may be  allowed to  open  doors at  any time,  or  allowed  and restricted  at certain times of  day  and day  of  week,  by  assigning the tag  number  to  any  one  of  seven programmable  schedules. DSS  and DND key  presence indication  —  DSS  keys  assigned  with  an  electronic  key  user’s  extension  or  mailbox number can  provide  a unique  indication  when that  person is  out  of  the office. Personal  Call  Routing  —  Electronic  key users that  have  phone extensions  can  have  the option  of  pre-assigning features  that  are  applied  automatically  when  tagging  in  or  out.  One  or  more  of  the following features  can  be  set: •  Personal  greeting selection  (greeting  1,  2,  or  3). •  Call  forward  all  calls (to  a  voice  mailbox,  extension,  department,  Esi-Link  location,  or  external  number)  or  send  all calls directly  to  voice  mail. •  Voice mail  message  delivery/notification on/off. Electronic  key  users associated  with  guest  mailboxes  can only  select  a  personal  greeting. Quiet time  —  Each  electronic  key user  can  schedule  a “quiet  time”  for  each  day  of  the week,  during  which the system  automatically  turns off  message notification and/or  call  forwarding  (if  assigned).  For  example,  calls  that  would have  forwarded are,  instead,  directed  to  voice mail. Time  and  attendance  management  —  Provided  by the  optional  software  package,  WaspTime1, for time  and attendance  tracking,  reporting,  and integration with  popular  payroll  and financial  packages. Door-ajar  alarm2  —  Some  doors will  generate  an  alarm  if  the door  has been left  open  for  too long.  If  this  alarm  is generated by the door, a message will appear in the display of the phones that ring when the CALL button is pressed.

Sensitive Handling Precautions

Contents Static  Sensitive Handling Precautions  Required • There are  two methods for  System  Initialization.  The  first method  is  to Clear  All  Data,  except LEN0000  as a  CAT terminal,  and then program  the  System  Data.  The  second method  is  to use  the  Resident System  Program,  which  causes  the  system  to  configure  itself  automatically  to  the  default  settings,  wherever  the  line/ trunk  cards  are  installed.  Refer  to the  System  Data  Sheet,  for  the  default  settings. • Turn on the  “SW1”  switch  on all  the  PZ-PW86  units. • The “ON” lamp  must  be  lit  on all  the  PZ-PW86  units. 1.1. All  Clear,  Except  LEN0000 CAT STEP  1: On the  MP  Card,  set SW3 to “B”  and  press SW1. STEP  2: When  the  “MN”  lamp  on the system  is lit,  set SW3 to the  “0”  position  and p

Program button

Scroll buttons (Up, Down) — Allow you to scroll through the items in

the telephone display panel. See “Using the NBX Telephone

Display Panel” in Chapter 7.

4 Program button — Reserved for future use.

5 Programmable Access buttons and label area — Allow you and your

administrator to assign features to specific buttons. See “Programmable

Access Buttons” and “Status Lights for System Appearance Buttons”

later in this chapter.

6 Programmable Access and label area — Allow you and your

administrator to assign features to specific buttons. See “Programmable

Access Buttons” later in this chapter.

7 Hold button — Places a caller on hold. See “Putting a Call on Hold”

in

Chapter 7

.

8 Transfer button — Sends the currently active call to another telephone.

See “Transferring a Call”

in Chapter 7.

9 Conference button — Establishes a single call with up to three

additional internal parties, external parties, or both. See “Establishing a

Conference Call” in Chapter 7.

10 Redial button — Redials the last telephone number or extension that

you called. See “Redialing a Call”

in Chapter 7.

11 Speaker button — Enables you to use the speaker phone feature. Press

the Speaker button before you dial the call, when your telephone is

ringing, or while a call is in progress. To turn the speaker phone off and

resume the conversation, pick up the ha

Abbreviated Dialing (AD

The Abbreviated Dialing (AD) feature allows you to store selected telephone
numbers for quick and easy dialing. Each number can be a complete or
partial telephone number, an extension number, or a trunk or feature access
code and can contain up to 24 digits. Abbreviated Dialing offers four possible
types of lists—Personal, Group, System, and Enhanced, and you can have a
total of three lists. (Of these three lists, you can have only one System List
and one Enhanced List.) You program numbers on a Personal List; numbers
on Group Lists are programmable by the controller of the list; System Lists
and Enhanced Lists are programmable only by the system manager.
To place a call using an AD list button or feature access code
1. Press and then the appropriate dial pad key on which a list
access code is administered,
or Dial the appropriate AD List code for List 1, or List 2, or List 3.
2. When you hear dial tone, dial the desired list item (1, 2, 3...).
Note: An Enhanced List may also be available; see your system
manager.
You can also program a number on an AD button that is not stored on an AD
list. This Automatic Dialing number can be up to 16 digits.
To program a number on an AD button
Note: The system manager must program a feature button as an AD
button before you can program a number onto it.
1. While off-hook, press and then the dial pad key assigned to the
AD Program feature,
or While off-hook, dial the Program access code.
You will hear dial tone.
2. Press and then the dial pad key you wish to program.
For example, to store a number in AD entry F3, press and then
3.
3. Dial the outside number, extension, or feature access code you want to
store (up to 24 digits).
4. Press # to save the entry.
You will hear confirmation tone and then dial tone.
5. Repeat Steps 2 through 4 to program additional buttons.
6. Hang up to end programming.
Abbreviated Dialing/Automatic Dialing (AD)

VLAN

VLANs are logical segments within a corporate LAN. By assigning VLAN settings to IP-PTs, it is possible to
separate the packets transmitted by an IP-PT according to the type of data, and specify which VLAN each data
type will be sent over. This allows you to avoid generating unnecessary network traffic on each segment, and
to reduce the load on the network. As a consequence, speech quality can be assured.
Therefore, we recommend using the VLAN feature to perform VoIP communication effectively.
Some IP-PTs (e.g., KX-NT136) are equipped with 2 ports, primary and secondary, for packet communication.
Allocating these ports to different VLANs enables you to split the paths for packets depending on whether the
packet contains voice signals or data.
VLAN settings (VLAN ID and VLAN priority) for the primary port affect voice data transmitted by the IP-PT,
whereas VLAN settings for the secondary port apply to data transmitted by a PC connected to the IP-PT. When
sending packets, the IP-PT can attach information on which VLAN the packets are to be transmitted over
(VLAN Tagging). The switching hub that receives these packets reads the VLAN information and sends the
packets over the appropriate VLAN. This helps to ensure bandwidth for IP-PT voice transmissions.
In this way, an IP-PT with 2 ports can transmit voice packets from the primary port with higher priority than
other packets from the secondary port.

MPPI

Major New Features in Version 1.2 of HiPath 3000
Important Notices
G281-0788-00, April 2002
1-14 HiPath 3000 V1.2-3.0, System Description
3000sb1.fm
New Options
In connection with the control boards, the following new optional boards are available.
CMS (Clock Module Standard)
The CMS module replaces the old CGMC clock generator. The
CMS must be used in the following situations:
● Clock generator for the master system in the network
● Clock generator for HiPath cordless during operation with the
SLC16 board
● Clock generator for a single-cell cordless solution (2-channel
mode) with BS3/S base station
CMA (Clock Module ADPCM)
The CMA module is implemented exclusively in the HiPath 3550,
HiPath 3350, HiPath 3500 and HiPath 3300 systems. The module
is required as a clock generator for HiPath cordless (4-channel
mode) with the BS3/1 base station when the new control boards
are connected directly to the UP0/E ports.
LIM (LAN Interface Module)
The LIM module enables you to integrate all HiPath 3000 models
into TCP/IP-based LANs economically for
● central system administration (TCP)
● call detail recording (CDRC)
● APS transfer or
● fault management.
A 10BaseT interface (10 Mbit) with an RJ45 jack is available for
connecting to the LAN.
IMODC (Integrated analog modem)
The format of the printed-circuit board on the integrated analog
modem for remote maintenance (remote access via analog
trunks) is new. The module is plugged into the appropriate slots
on the control boards (this is not possible in HiPath 3250 or Hi-
Path 3150).
MPPI (optional music on hold from Beyertone)
The plug-in MPPI module can be used universally in all HiPath
3000 models making an economical solution available for an al-
ternative music on hold.

Service code

For certain functions, you can append data to the key basic function. For example, the function 26 appended by data 1 makes a Group Call Pickup key for Pickup Group 1. You can also program Function Keys using Service Codes. To clear any previously programmed key, press  000  to erase any displayed code.

DISA number

Dialed DISA number gets ringback instead of stuttered dial tone The remote caller has dialed a  DISA number, but has instead reached a target line,  or has been routed  to the prime telephone for the auto-answer trunk. The Norstar system has mapped the incoming digits from a switch onto a target line, or has been unable to map the digits anywhere. Solution 1. Verify  that  the switch is  sending the correct number of digits for the Received number length defined in your system. 2. Verify all the digit strings that the switch should be sending. 3. Check that you have defined the correct DISA DN for your system. 4. Make sure that the published telephone numbers for your network are correct. Dialed DISA number gets dial tone instead of stuttered dial tone Possible problem The remote caller has  dialed a DISA number, but has reached the Norstar system instead. The Norstar system has mapped the incoming digits from a switch onto the Auto DN. Solution 1. Verify  that  the switch is  sending the correct number of digits for the Received number length defined in your system. 2. Verify all the digit strings that the switch should be sending.

bin is accessed

Per speed dial bin (000 ~ 999) assign the telephone number that is to be called when the bin is accessed. If 13-01-01 is set to (0) do not include the trunk access code. If 13-01-01 is set to (1) include the trunk access code.

Terminals connect

Confirm the terminal connected with same ESI. If they work normally, confirm the breakdown or the wiring for the terminal. Exchange the terminal that doesn't work and the working terminal, and confirm it’s working. An external factor of the noise etc. is thought. Please reconfirm wiring and the installation, etc. Please inquire of the manufacturer when the problem occurs after it confirms it.

set the system passwords

Password Setup  to set the system passwords. For password entry, the system allows eight users to be defined. Each user can have a: • Unique alphanumeric name (up to 10 alphanumeric characters) • Password entry of up to eight digits (using 0 ~ 9, # and • Password level *) The IN level password is used by the System Installer for system programming. The SA or SB level password cannot access the IN level programs. The reverse type (white on black) just beneath the Description heading is the program access level. You can only use the program if your access level meets or exceeds the level the program requires. (SA level password can access to SA or SB programs, and SB level password can access to SB programs only.)

Code restrictions

Code Restriction/Toll restriction limits the telephone numbers a user may dial.  By allowing extensions to place only certain types of calls (e.g. local and toll free), you can better control long distance costs.  The system applies code restriction/toll restriction based on the users assigned code restriction class.

 

The system allows for up to 15 different code restriction classes.

 

incoming digits

The Norstar system did not receive some or all of the incoming digits. Solution 1. Check that the system hardware is receiving signals properly. 2. Verify  that  the switch is  sending the correct number of digits for the Received number length defined in your system. 3. If the switch at the far end is sending pulse signals, make sure they are being sent at the proper rate. Pulse digits must be 300 ms or more apart for Norstar to receive them. Possible problem If remote callers are having difficulty getting through from the public network, there may be  a problem with your DID Trunk Cartridge. Solution 1. Connect a single-line DTMF telephone to a DID trunk input on the DID Trunk Cartridge. 2. Use the single-line telephone  to enter a Received number that has been programmed for a  target line in your system. Listen for ringback.

Stuttered number

number gets stuttered dial tone instead of ringback Possible problem The remote caller has tried to reach a Norstar target line, but has reached a trunk with DISA instead. Alternatively, the  Norstar system has mapped incoming digits onto the DISA DN. Solution 1. Verify all the digit strings that the switch should be sending. 2. Check that you have defined a corresponding Received number for every target  line in your system. 3. Check that you have defined the correct DISA DN for your system. 4. Make sure that the published telephone numbers for your network are correct. Dialed number  gets  dial  tone instead of  ringback Possible problem The remote caller has tried to reach a Norstar target line, but has reached the Norstar system instead.  Norstar has  mapped the incoming digits onto the Auto DN. Solution 1. Verify all the digit strings that the switch should be sending. 2. Check that you have defined the corresponding

Remote feature code

Remote feature code gets no response Possible problem A Norstar user has called into  another Norstar system and is trying to activate a remote feature but gets no response after dialing the feature code. Solution 1. Make sure that  the  remote caller is dialing the feature code correctly. Use the asterisk  (*) character, followed  by the feature code, to activate a remote feature. Do not use  ≤ for accessing features  on a remote Norstar system. 2. Make sure that the remote user  is dialing tones, not pulses, once the call is answered.

Module down

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

Receive Alarm

Receive Alarm  — yellow LED on indicates a problem with the digital transmission  being received. This halfduplex link is unusable. • Receive Error — yellow LED indicates a minor  error as  a result of degraded digital transmission. Possible causes are an ohmic connection, water ingress, or too long a loop. • Transmit Alarm  — red LED on indicates an inability to transmit. Alarm indication signal (AIS) is being transmitted to the terminating switch. This half-duplex link in unusable. • Transmit Error  — yellow LED on indicates a remote alarm indication (RAI) carrier  failure alarm (CFA) is being sent to the  terminating switch. If the Transmit Alarm is not on, this indicates a far-end or cable problem. • In service — a flashing green LED indicates that the T1 or PRI trunks are out of service because of a running loopback test, or because the DTI is being initialized. • Loopback test — red LED on while a continuity loopback test is running. • All LEDs flashing continuously  — the DTI is being initialized. 3. Run a Maintenance session and any loopback tests as appropriate. 4. Check the pinout of the cable that connects the DTI to the termination point from the T1  or PRI service provider or the external channel service unit, and check that the cable is properly connected. 5. Check with your T1 or PRI  service provider to see if through-fed repeaters are used on the T1 or PRI span. The DTI does not provide the DC conne

External paging

External paging 1. Use the Button Inquiry feature (≤•‚) to verify  the feature of a programmable memory button. 2. Check the wiring between the 50-pin connector and the paging amplifier or between the connections shown in the external paging wiring chart. External paging  wiring Feature Page out (Tip) Page out (Ring) Pin 40 (Black-Slate) 15  (Slate-Black) Page (Make) Page (Common) 41  (Yellow-Blue) 16  (Blue-Yellow) 3. Test external paging  ≤fl¤  to ensure that it is working. The nominal output signal from the Norstar ICS is 100 mV across 600  Ω. Music on Hold/Background Music trouble Although Music on Hold and Background Music are separate features, they share the same  wiring and customer-supplied music source. 1. Ensure that the proper feature access code (≤°fl) is turned on. Adjust the volume  using the volume control bar. 2. Use the Button Inquiry feature (≤•‚) to verify  the feature on a programmable memory button. 3. If there is trouble with Music on Hold or Background Music, check  Featr settings  in  System prgming.

Selected line pool displays: No free lines

Selected line pool  displays: No free lines If the user selects a line  pool and the telephone displays No free lines, refer to this section for solutions. Possible problem If this happens often, there are not enough lines in the line pool to serve the number of line pool users. Solution If the line pool contains loop  start trunks, enter programming and move under-used loop start  trunks from other line pools into the deficient line pool. OR If the line pool contains E&M  trunks, order more trunks from the telephone company or private network vendor. Install

Dial tone absent

Dial tone absent on external lines 1. Use Button Inquiry (≤•‚) to display the number of the external line you are testing. 2. Check for dial tone using a test telephone at the connections for the external  line on the distribution block. 3. Make sure that a Trunk Cartridge for the line is properly installed in the ICS or Trunk Module. 4. Make sure that the Trunk Module fiber cable is properly connected to the Expansion Cartridge on the ICS. Refer  to  Problems with trunk cartridges service  on page 611 and  Trunk or Station Module down  on page 619. 5. Run a Maintenance session to ensure that the line is not disabled. See  Port/DN status

password entry

Programming Password Setup to set the system passwords. For password entry, the system allows eight users to be defined. Each user can have a: • Unique alphanumeric name (up to 10 alphanumeric characters) • Password entry of up to eight digits (using0~9,#and*) • Password level The IN level password is used by the System Installer for system programming. The SA or SB level password cannot access the IN level programs. The reverse type (white on black) just beneath the Description heading is the program access level. You can only use the program if your access level meets or exceeds the level the program requires. (SA level password can access to SA or SB programs, and SB level password can access to SB programs only.)

Outgoing trunk preference

OutgoingTrunkLinePref-erenceUsethisoptiontosettheextensionoutgoingTrunkLinePreference.Ifenabled,theextensionuserreceivestrunkdialtonewhentheyliftthehandset.TheuserhearstrunkdialtoneonlyifallowedbyTrunkAccessMappro-gramming(Programs14-07and15-06).RefertotheLinePreferencefeatureformoredetails.

Multi-Party Voice Conference

Multi-Party  Voice  Conference Description The system  allows  multiple internal  and external  parties  to be connected on a call,  conference. An unlimited number  of  3-party  conferences  may  be established using  iPECS  Phones.  In addition,  each  MCIM  (Multi-party  Conference Interface Module)  up to 32 parties  with the g.711 or  g.729 codec  and 24 with the  g.723 codec  may  be connected in a single voice conference. The MCIM  will  support  any  combination of  parties  and conferences  to the  maximum  total number  of  parties  in conference.  Multiple  MCIMs,  see chart  below,  are installed to support multiple multi-party conferences with a maximum of 32 parties in any single conference. iPECS iPECS iPECS iPECS iPECS Operation ----Micro 50 300 600 1200 iPECS Phone 1 MCIM  unit  maximum 2 MCIM  units  maximum 4 MCIM  units  maximum 8 MCIM  units  maximum 30 MCIM  units  maximum To establish  an  ad-hoc  conference 1.  Establish  first  call. 2.  Press  the  [CONF]  button.  The LED  will  light,  the connected party  is  placed on exclusive hold and the user  receives  dial  tone. 3.  Place second call. 4.  When connected,  press  [CONF],  new  call  is  placed on exclusive hold. 5.  Repeat  steps  3 and 4 above to add additional  conference parties. 6.  Press  [CONF]  button  to establish conference. To place  a  conference  on hold 1.  Press  the  [HOLD]  button,  the  [CONF]  button LED  will  flash. To retrieve  held  conference 1.  Press  [CONF]  button,  all  parties  reconnected.

Conference Room

Conference  Room Description In addition to ad-hoc  conferencing,  users  may  establish a Conference Room.  Other  internal and external  parties  are invited to the conference  and can join the conference without  further action by  the user  who established the Conference Room.  A  user  can  transfer  an active call  to a Conference Room.  A  Conference Room  can be  password protected so that  only  parties  that enter  the password are allowed to join the Room. Up to 9 Conference Rooms  can be set-up  and each can support  a maximum  of  32 parties  with the g.711 or  g.729 codec  or  24 parties  with the g.723 codec.  Conference Rooms  employ channels  from  an MCIM  (Multi-party  Conference Interface Module).  Each MCIM  supports  up to 32 parties  and multiple  MCIM  units  may  be installed as  shown  in  the chart  below. iPECS-Micro iPECS-50 &  100 iPECS-300 iPECS-600 iPECS-1200 Operation Attendant  Phone To view  Room  participant  list 1 MCIM  unit  maximum 2 MCIM  units  maximum 4 MCIM  units  maximum 8 MCIM  units  maximum 30 MCIM  units  maximum 1.  Press  the  [PGM]  button. 2.  Dial “054”. 3.  Dial  Room  number  (1~9). To delete a  Conference  Room 1.  Press  the  [PGM]  button. 2.  Dial “055”. 3.  Dial  Room  number  (1~9). iPECS Phone To set-up a  Conference  Room 1.  Press  the  [PGM]  button. 2.  Dial  53 to create a Conference Room. 3.  Dial  the desired Conference Room  number  (1~9). 4.  If  desired,  enter  a password  for  the Conference Room  (must  be exactly  5 digits). 5.  Press  [SAVE]  button  to establish the Room. To join a Conference  Room 1.  Dial  59,  the Conference  Room  entry  code. 2.  Dial  the Conference Room  Number. 3.  Dial the Conference Room  password.

HOLD/SAVE

Differential Ring Signals When multiple phones in a small area ring, it can be difficult to  tell  which are  ringing.  The iPECS  Phone has 14  Ring Tones  available for differentiating  ring from one phone to another.  Four of the tones are stored in the  phone’s  permanent memory; the remaining ten tones  are  in the system’s memory.  Four of  these ten can be downloaded into the phone’s memory for use as the 5th  to 8th Ring Tone. OPERATION To download a Ring Tone from System memory:   Lift Handset   Press  TRANS/PGM,   Dial 1 for Ring Tones,   Dial 5 for Ring Tone download,   Dial the memory location  to receive the tone (5-8),   Dial tone number 0-9, tone  is heard,   Press  HOLD/SAVE  to download. To select a Ring Tone  from phone memory:   Press  TRANS/PGM,   Dial 1 for Ring Tones,   Dial 1 or 2 for Internal or EDifferential Ring Signals When multiple phones in a small area ring, it can be difficult to  tell  which are  ringing.  The iPECS  Phone has 14  Ring Tones  available for differentiating  ring from one phone to another.  Four of the tones are stored in the  phone’s  permanent memory; the remaining ten tones  are  in the system’s memory.  Four of  these ten can be downloaded into the phone’s memory for use as the 5th  to 8th Ring Tone. OPERATION To download a Ring Tone from System memory:   Lift Handset   Press  TRANS/PGM,   Dial 1 for Ring Tones,   Dial 5 for Ring Tone download,   Dial the memory location  to receive the tone (5-8),   Dial tone number 0-9, tone  is heard,   Press  HOLD/SAVE  to download. To select a Ring Tone  from phone memory:   Press  TRANS/PGM,   Dial 1 for Ring Tones,   Dial 1 or 2 for Internal or External ring,   Dial tone number 1-8,   Press  HOLD/SAVE  to make  the selection. 2.4  Answering  Calls at Night In the Night  mode,  Loud Bell  Control  may be used  to send ring  signals to external bells.  You  may  then answer these calls with  Universal Night Answer (UNA). OPERATION To answer  a call during Night mode ringing over an external bell:   Lift the handset,  Dial the UNA code 567(iPECS-Lik)/587(iPECS-MG).   to make  the selection. 2.4  Answering  Calls at Night In the Night  mode,  Loud Bell  Control  may be used  to send ring  signals to external bells.  You  may  then answer these calls with  Universal Night Answer (UNA). OPERATION To answer  a call during Night mode ringing over an external bell:   Lift the handset,  Dial the UNA code 567(iPECS-Lik)/587(iPECS-MG).

Define the extension

Define the extension/virtual extension name.Upto12CharactersExt.101~184=Nosetting02OutgoingTrunkLinePref-erenceUsethisoptiontosettheextensionoutgoingTrunkLinePreference.Ifenabled,theextensionuserreceivestrunkdialtonewhentheyliftthehandset.TheuserhearstrunkdialtoneonlyifallowedbyTrunkAccessMappro-gramming(Programs14-07and15-06).RefertotheLinePreferencefeatureformoredetails.

Dial code to reroute call after answer

Dial code to reroute call after answer.  578 578 578 578 578 10 ACD REROUTE QCALL NO AN ENTER NEW #:579 Dial code to reroute call prior to answer.  579 579 579 579 579 11 CAMP-ON ANSWER ENTER NEW #:600 Dial code to answer a Camped On call.  600 600 600 600 600 12 CALL PARK LOCATIONS START&END#:601-610 Dial code to place/retrieve a call in a Park location.  601~610601~610 601~619 601~699601~699 13 STA GRP PILOT NUMBER START&END #:620-659 Station group pilot numbers.  620~631620~659 620~667 620~667401~500 14 STA USER VSF FEATURES ENTER NEW #:66 VSF feature access dial code.  66 66 *66 *66 *66 15 CALL COVERAGE RING ENTER NEW #:67 Code for Call Coverage button.  67 67 67 67 76 16 DIRECT CALL PICK-UP ENTER NEW #:7  Dial code to activate Directed Call Pick-up.  7 7 7 7 *77 17 ACCESS CO GROUP FEAT START&END:801-820 Dial code to access a CO Line or IP channel from a CO/IP group.  801~820801~820 801~872 801~872n/a 18 ACCESS IND CO/IP FEAT START&END:88 Dial code to access a specific CO Line.  8801 ~8805 8801  ~8842 88001 ~ 88200 88001 ~88400 88001 ~ 8860019 ACCESS HELD CO/IP FEAT ENTER NEW:8* Dial code to access last held CO Line or IP channel from Hold.  8* 8* 8* 8* 8* 20 ACCESS HELD IND CO/IP ENTER NEW #:8# Dial code to access a specific CO Line/IP channel from Hold.  8# 8# 8# 8# 8# 21 ACCESS CO IN 1ST CO GRP ENTER NEW #:9 Dial code to access the 1st available CO Line in any accessible group.  9 9 9 9 9 22 ATTENDANT CALL ENTER NEW #:0 Dial code to call an Attendant.  0 0 0 0 0 23 VM MSG WAIT ENABLE ENTER NEW #:*8 Dial code for external Voice mail to activate Message Wait indication.  *8 *8 *8 *8 *8 24 VM MSG WAIT CANCEL ENTER NEW #:*9 Dial code for external Voice Mail to deactivate Message Wait

DID

DID calls are subject to Group Call Pick-up and Directed Call Pick-up. 5. If a VMIM/VSF announcement is defined as the destination in the Flexible DID Destination Table, a Caller Controlled Routing Table for the announcement can be defined. iPECS can be configured to drop (disconnect) the call after playing the recorded announcement. Programming STATION 1 SIP User ID Table (PGM 111-Btn 19)  2 Station SIP Attributes 2 (PGM 126-Web only) BOARD 1 H323 VoIP Attributes  2 SIP Gateway Attributes CO/IP 1 CO Service Type (PGM140)  2 ISDN DID Remove Digit Count (PGM 143-Btn 5)  3 DID Conversion Type (PGM 145) SYSTEM 1 DID/DISA Busy Destination (PGM 167-Btn 1)  2 DID/DISA Error Destination (PGM 167-Btn 2)  3 DID/DISA No Answer Timer (PGM 181-Btn 2) TABLES 1 CCR Audio Text Tables (PGM 228)  2 Flexible DID Table (PGM 231

B Channel

B channels B channels are the bearer channel.  They are used to carry voice or data information and have speeds of 64 kbps. Since each ISDN line (BRI or PRI) has more than one B-channel, more than one transmission can occur at the same time, using a single ISDN line. D channels The standard signaling protocol  is  transmitted  over a dedicated data channel called the D-channel. The D-channel carries call setup and feature activation information to the destination. This channel has speeds of 16 kbps (BRI) and 64 kbps (PRI). Data information consists of  control and signal information and packet-switched data such  as credit card verification.

ISDN layers

ISDN layers ISDN layers refer to the standards established to guide the manufacturers of ISDN equipment. The layers include both physical connections, such as  wiring, and logical  connections, which are programmed in computer software. When equipment is designed to  the ISDN standard for one of the layers, it works with equipment for the layers above and below it. There are three layers at work in ISDN for Norstar. To support ISDN service, all three layers must be working properly.

UCS Mobile Client

IPECS applications Taking Business Communications to A New Level UCS Desktop Client  UCS Desktop Client enables presence, instant messaging, file sharing, multi-party conferences, internal and external SMS, video conferencing, call recording, application sharing, white board and desktop sharing, MS Outlook integration and much more. UCS is designed as a single-server solution with straight-forward installation and management with simple licensing that results in low total-cost-of-ownership (TCO).  UCS Mobile Client No need to be in the office to take that big meeting.  UCS Mobile Client brings the power of iPECS UCS to your smartphone.  Available for Android, iOS and Windows.    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 are optimized to help you meet your needs while your business continues to transform.  Unifying Your Business iPECS UCS is a complete Unified Communications solution designed for the SMB that keeps people communicating when they are in the office or away. 

ISDN

This chapter provides you with  some background information about ISDN, including information about: • analog vs. ISDN • type of ISDN service •ISDN l ayer s • ISDN bearer capability • services and features  for ISDN PRI and  BRI • ISDN hardware • ISDN standards compatibility Integrated Services Digital Network (ISDN) technology provides a fast, accurate, and  reliable means of sending and receiving voice, data, images,  text, and other information through the telecom network. ISDN uses existing analog telephone wires. The signal on the wire gets divided into separate digital channels, which dramatically increases the bandwidth. ISDN uses a single transport to  carry multiple information types. What once required separate networks for voice, data, images, or video conferencing is now combined on to one common high-speed transport.

iPECS sBG-1000

iPECS sBG-1000 Integrated office in a box The iPECS SBG-1000 is the ideal converged business communication services platform for small businesses, multi-site and branch office environments.   With voice, data and IT services built-in, the SBG-1000 is a unique platform that offers the best price-to-feature ratio, all in a single appliance. Enhanced security, VPN and mobility features allow businesses to confidently extend and manage access and services to remote and mobile workers in a secure environment.

intercom call hold

INTERCOM  CALL HOLD Description While on an active ICM  Call,  users  of  iPECS  Phones  can place the ICM  Call  on hold.  The held station will  receive the assigned Music-on-Hold.  The call  is  placed on Exclusive Hold and recalls  to the holding  station after  expiration of  the  Exclusive Hold Recall  Timer. Operation iPECS Phone To place  an active  ICM  call  on hold 1.  Press  the  [HOLD]  button,  the  [ICM]  button LED  will  flash at  the exclusive hold  rate. ICM  dial  tone is  received. To retrieve  the held ICM  call 1.  Press  the  [ICM]  button  or  the  {DSS/BLF}  button associated  with the held station, the  [ICM]  button LED  illuminate and the ICM  call  connected. Conditions 1.  Only  one ICM  call  may  be placed on hold  at  a station. Programming Related  Features   MOH (Music-On-Hold)   Intercom  Call (ICM  Call)   Exclusive  Hold   Hold Recall Hardware   iPECS Phone

Intercom call hold

INTERCOM  CALL HOLD Description While on an active ICM  Call,  users  of  iPECS  Phones  can place the ICM  Call  on hold.  The held station will  receive the assigned Music-on-Hold.  The call  is  placed on Exclusive Hold and recalls  to the holding  station after  expiration of  the  Exclusive Hold Recall  Timer. Operation iPECS Phone To place  an active  ICM  call  on hold 1.  Press  the  [HOLD]  button,  the  [ICM]  button LED  will  flash at  the exclusive hold  rate. ICM  dial  tone is  received. To retrieve  the held ICM  call 1.  Press  the  [ICM]  button  or  the  {DSS/BLF}  button associated  with the held station, the  [ICM]  button LED  illuminate and the ICM  call  connected. Conditions 1.  Only  one ICM  call  may  be placed on hold  at  a station. Programming Related  Features   MOH (Music-On-Hold)   Intercom  Call (ICM  Call)   Exclusive  Hold   Hold Recall Hardware   iPECS Phone

User Agent A SIP User Agent

Network  Elements   1.1   User Agent A  SIP  User  Agent  (UA)  is  a logical  network  end-point  used to  create  or  receive  SIP  messages  and thereby  manage a SIP  session.  A  SIP  UA  can  perform  the role  of  a  User  Agent  Client  (UAC),  which  sends  SIP  requests,  and the  User  Agent Server  (UAS),  which receives  the requests  and returns  a SIP  response.  These roles  of  UAC  and UAS  only  last  for  the duration  of  a SIP  transaction. 1.2   Proxy server An  intermediary  entity  that  acts  as  both a server  (UAS)  and a  client  (UAC)  for  the  purpose  of  making requests  on behalf of  other  clients.  A  proxy  server  primarily  plays  the  role  of  routing,  which means  its  job is  to  ensure that  a request  is  sent  to another  entity  "closer"  to  the targeted  user.   1.3   Registrar A  server  that  accepts  REGISTER  requests  and  places  the  information  it  receives  in those requests  into the  location service for  the  domain  it  handles  which  registers  one  or  more IP  addresses  to  a certain  SIP  URI. SIP  registrars  are  logical  elements,  and are commonly  co-located with SIP  proxies.  But  it  is  also possible and  often  good for  network  scalability  to  place this  location service  with a redirect  server. 1.4   Redirect server A  user  agent  server  that  generates  3xx  (Redirection)  responses  to requests  it  receives,  directing the client  to contact  an alternate set  of  URIs.  The redirect  server  allows  proxy  servers  to direct  SIP  session invitations  to external  domains. 1.5   Session border controller Session border  controllers  serve as  middle boxes  between  UA  and  SIP  server  for  various  types  of  functions,  including network  topology  hiding,  and assistance  in  NAT  traversal. 1.6   Gateway  Gateways  can  be  used to  interface a  SIP  network  to other  networks,  such as  the  public  switched telephone network, which use different protocols or technologies. 1.7   Application Layer Gateway  ALG  is  a  SIP  aware monitoring  device  commonly  contained  in Routers  and  or  Firewalls.  SIP  ALGs  can  have  the capabilities of changing SIP Messages and should be disabled if any issues are experienced with SIP Calls.

C-3000

This entry system is designed to utilize a building’s existing telephone wiring and to address a variety of apartment entry applications.  Viking’s  C-3000 Entry System provides secure multi-tenant building access, without the need for any C.O. lines; the only requirement is that each tenant have a touchtone telephone set wired to their apartment.  The  C-3000 performs equally as well in installations where C.O. lines are present. An  individual  C-3000 module has a 12-tenant capacity; for larger applications, a maximum of eight (8)  C-3000 modules can be interconnected, allowing a total system capacity of up to 96 tenants.  A C-3000 system consists of at least one Master  C-3000 module, and as many as seven Slave modules. From either of two entry doors, visitors may ring a tenant's telephone with a distinctive cadence, alerting the tenant of their arrival.  The  C-3000 provides call waiting tones if the tenant’s line is in use.   The tenant may then converse freely with the visitor via their existing telephones. Once identified, the tenant can let the visitor in by entering a single touch-tone on the keypad of the tenant's telephone set. The  C-3000 keyless entry feature supports unique entry codes for each tenant.  As a measure of security, the codes can be easily changed as tenants move in and out of the building. A postal lock feature is also supplied as an added convenience. A

LIK-MFIM50B Multi-Function Interface gateway

LIK-MFIM50B Multi-Function Interface gateway Module, 50 ports w/ISDN-BRI 1 MICRO Multi-Function Interface gateway Module, 31 ports 2 AC/DC Adapter  AC/DC Adapter for LIK-MFIM50A/B module (12VDC, 1.5A) 3 AC/DC Adapter –K- AC/DC Adapter for LIP Phones and DSS Console (48VDC, 0.3A) 4 DHLD Desk mount Holder for module 5 DHE Desk mount Holder Extender, one (1) required for each Module  6 WHLD Wall mount Holder for module 7 1U-RMB 1U Rack mount Bracket 8 LIP-7004N LIP Phone, Basic 4 button no display 9 LIP-7008D LIP Phone, 8 button and basic 2-line display 10 LIP-7016D LIP Phone, 16 button, 3-line display w/Menu, Soft & Nav. buttons 11 LIP-7024D LIP Phone, 24 button, 3-line display w/Menu, Soft & Nav. buttons 12 LIP-7024LD LIP Phone, 24 button, Large display w/Menu, Soft & Nav. buttons 13 LIP-7048DSS LIP DSS Console with 48 buttons 14 LIP-8002 LIP Phone, 4 button and 1-line display, LAN 1 port 15 LIP-8004D LIP Phone, 4 button and 1-line display, LAN 1 port 16 LIP-8012D LIP Phone, 12 button, 3-line display w/Menu, Soft & Nav. buttons 17 LIP-8024D LIP Phone, 24 button, 4-line display w/Menu, Soft & Nav. buttons 18 LIP-8040L LIP Phone, 10 button, 9-line display w/Menu, Soft & Nav. buttons 19 LIP-8048DSS LIP DSS Console with 48 buttons 20 LIP-8012DSS LIP DSS Console with 12 buttons 21 LIP-8012LSS LIP DSS Console with 12 buttons, w/12-line LCD button label 22 LIP-7004WMK Wall Mount Kit for LIP-7004N 23 LIP-7008WMK Wall Mount Kit for 7008D 24 LIP-7024WMK Wall Mount Kit for LIP-7016D, 7024D & 7024LD 25 WIT-300HE/400H iPECS WLAN Phone 25 WIT-400H iPECS WLAN Phone 26 GDC-400B DECT Base Station 27 GDC-600B DECT Base Station 28 GDC-400H DECT Handset 29 GDC-450H DECT Handset

Power supply

Power Supply The  Total Access 900e Series products have a 90  to  125 VAC power supply with  an  IEC connector. The  appropriate three-prong cable  is included  in the shipment. Battery Backup Connection An  optional  battery backup system is available  for the  Total Access 900e Series (P/N 1175044L1 or L2). The  connection  port is  labeled  BATT. Refer to  the documentation  available for your specific battery backup  unit  for more information on this  connection, or refer to  Battery Backup Unit  on page 25 for  a more details. CRAFT Interface The  CRAFT  interface is an  EIA-232  serial  port (DCE) that  provides for local management  and configuration  (via a DB-9  female  connector). Table A-5  on  page 31  shows  the  CRAFT  port pinouts.

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