sip server

SIP Server Information Setup  to define the SIP Proxy setup for outbound/ inbound. The 10-29 commands are not used in non-registration mode.If entries are made in Program 10-29-xx for a SIP Server and the SIP Server is then removed or not used, the entries in Program 10-29-xx must be set back to their default settings. Even if 10-29-01 is set to 0 (off), the system still checks the settings in the remaining 10-29 programs.

Daylight Savings Setup

Daylight Savings Setup  to set the options for daylight savings. As the telephone system is used globally, these settings define when the system should automatically adjust for daylight savings as it applies to the region in which the system is installed

Pre-ringing

Pre-Ringing Setup  to enable or disable pre-ringing for trunk calls. This sets how a trunk initially rings a telephone. With pre-ringing, a burst of ringing occurs as soon as the trunk LED flashes. The call then continues ringing with the normal ring cadence cycle. Without pre-ringing, the call starts ringing only when the normal ring cadence cycle occurs. This may cause a ring delay, depending on when call detection occurs in reference to the ring cycle.

To enter programming mode

To enter programming mode : 1. Go to any working display telephone.In a newly installed system, use extension (port 1). 2. Do not lift the handset. 3. Press  Speaker. 4. # *  # *. 5. Dial the system password +  Hold. Refer to the following table for the default system passwords. To change the passwords,

PRI applications

T1/PRI For  T1 or  PRI  applications  (only  PRI on  the  ESI-50; it doesn’t support  T1),  an  ESI  Communications  Server can use a  compatible  digital  line  card  (DLC)1: •  ESI-1000,  ESI-600,  ESI-200,  ESI-100 —  DLC  and  DLC12,  each  for  either  T1  or  PRI. •  ESI-50  —  DLC82  for  only  PRI. Depending  on  how  you configure it,  each  supports  either  (a.)  a  single  T1 circuit  at  24  DS0 channels or  (b.)  a PRI  circuit  supporting  23  “B”  (bearer)  channels  and  one “D”  (data link)  channel.  The DLC12 and  DLC82 each also  support  12  digital  stations.  The  T1 or  PRI  line is  connected  via the  last  two pairs  of  the industry-standard 50-pin  amphenol  cable  connector  on  the front  of  the  DLC. Each  ESI  Communications  Server  has  a  different  maximum  number  of  system-wide  DLCs  (see  “Port  card options,”  page  A.4).  Partial  T1  or  PRI  applications  are supported  through  line  programming. Each DLC has built-in  CSU  functionality.  The  integrated  CSU  can be enabled  or  disabled  via system programming2.  The following functionality is  provided:  line,  payload,  DTE  and none  (normal  operation) loopback  modes  with  the  ability  to  respond  back  controlled  via  system  programming;  alarm  conditions,  and both  ANSI  T1.403  and  TR  54016  performance messages  for  ESF only.  Important:  On  the ESI-50,  the DLC82 may be installed in  only  slot  2. If  you’re  installing more  than one  T1  or  PRI,  the  DLC  in the lowest  number  slot  will synchronize (“slave”) the system  with  the public  network.  The system will  synchronize  to  only one clock  source.  Therefore,  ESI strongly  recommends  that  the first  DLC  in the  system be connected to  the  T1  or  PRI  that’s connected either  to  the local  CO  or  the nationwide long-distance provider,  either  of  which typically will provide  veryhigh-accuracy  clocking (Strata 3).  The DLC  doesn’t  provide master  or  sub-master  clocking for  privatenetwork  T1  spans.

ESI System Programmer

The  ESI  Communications  Server  supports  the  48-Key  IP  Feature  Phone  II,  ESI  IP  Cordless  Handsets,  VIP Softphone,  and  SIP  phones.  (See “System  capacities,”  page  B.1,  for  the maximum  number  of  IP  phones that your  specific  ESI Communications  Server  will  support.) The ESI-50 has  a built-in IVC12.  It  can  support  up to  12 IP  channels,  which can  be  a  combination of  local  IP, remote  IP,  and Esi-Link  channels.  The channels  are  activated in  blocks  of  four  for  local  IP,  singles for  remote IP,  and four  or  twelve  for  Esi-Link.  Here  is  an  example of  some  possible  ESI-50  IVC12 channel  combinations: •  12 all  Esi-Link. •  12  all local IP. •  Eight Esi-Link, four  local IP. •  Four  Esi-Link, four  local IP, four  remote  IP. When  two  or  more  Intelligent  VoIP  Cards  (IVCs)1  and the  necessary licensing are installed in  an  ESI Communications  Server, the  first IVC  (lowest-numbered  slot) will be  designated  as  the  primary  IVC,  which acts  as  a “go-between”  to  associate a  station to  its IVC.  To  each IVC,  the  system  automatically  allocates  24 sequential  extension  numbers,  as  defined  in  the  dial  plan selected  in  Function 169.2  Therefore,  the  primary  IVC must be connected  to  the  same  network  as  all  of  the  other  IVC  station cards. If  an IVC  supports  12  IP  stations,  only  the first  12  extension  numbers  can be  assigned to  IP  stations. Programming  IP  stations  is  similar  to  programming  digital  stations,  except  that  additional,  IP  networking parameters are required for  the former. There  are three ways  IP  networking parameters can  be assigned to  IP  stations in  an  ESI  Communications Server: •  Via  Function  31, as  described  in  the  following  pages. •  Using  ESI  System  Programmer. • Via “setup mode” at an ESI IP Feature Phone II.

Centrex

Centrex/PBX  access code If the  system  is  to  be  used  behind  Centrex  or  another PBX,  you  must  list  the  dial  access  code  used  to  gain access  to  a CO  line from  Centrex  or  the PBX,  so that  toll  restriction  can ignore  the  access  code  digit(s).  Users must  dial  the access  code  after  accessing  a line  by  either: (a.) Dialing 9, 8, 71, 72, 73, 74, 75, or 76.  or (b.)  Pressing  a  line key (if  programmed). The access  code can be one or  two digits  —  e.  g.,  9,  81,  etc.  —  and must  be programmed for  each line group. Default:  0.  Note:  You must  set  the flash duration in Function 151 (page  E.3)  for  the requirements  of  the host  switch. Function  222: Toll restriction exception tables The  system’s  toll  restriction  is  based  on  outbound  calls  being  defined  as  either  toll calls  (i.e.,  calls  in  the  deny table) or non-toll calls (calls in the allow table). Four tables exist for this purpose: 1.  Allow exception  table  (programmable).  Up to  100 entries;  no  entry  can exceed  26  digits. Default: No entries. 2.  Deny exception  table  (programmable).  Up to  100  entries;  no  entry  can exceed  26  digits. Default:  No  entries. A number listed  in  the  allow  exception  table  —  e.g.,  a  branch  office  or  vendor’s  location  —  will  be  allowed  to all  stations,  regardless  of  how they’re  set  in  Function  32  (see  page  G.19).  Conversely,  a number  listed  in  the deny exception table (e.g., a “1-900” number) will be denied to all stations. 3.  Fixed allow  table  (not  programmable). Default:  1800,  1888,  1877,  1866,  1855,  1844,  1833  and  1822.  4.  Fixed  deny  table (not  programmable). Default:  976,  1976,  1xxx976,  900,  1900,  1xxx900,  555,  1555,  1xxx555,  0,  10,  411,  1411  and   11+-digit  restriction. In  extension feature  authorization (Function  321;  see page  G.19),  each  extension is  set  to  be  toll-restricted one of  two  ways:  TOLL  CALLS = Y  (yes) or  TOLL  CALLS  = N  (no).

Background Music Description Background Music (BGM)

Background Music Description Background Music (BGM) sends music from a customer-provided music source to the Speakers of the Multiline Terminal when the station is idle.  Each 084M-B1 unit has 2 Audio In jacks on board and J431 (BGM) is used for BGM. As system can have 1 BGM input, effective BGM port needs to be determined at PRG 10-60-01. B Conditions • Background Music stops while the Multiline Terminal is in use. • Originating a call, answering a voice announcement, a ringing call, or internal paging interrupts Background Music. • Background Music is not available on Single Line Terminals. • Refer to Analog Communication Interface (ACI) for detail settings.

Repeat Redial

Sets how many times a Repeat Redial automatically repeats if the call does not go through. Default 3 02 03 04 Repeat Redial Interval Time Repeat Dial Calling Timer Time for Send Busy Tone for ISDN Trunk Conditions None 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds Set the time between Repeat Redial attempts. After dialing the trunk call, Repeat Redial maintains the call after this time. After this time, the system terminates the call, waits the Repeat Redial Time (Timer 02) and tries again. Sets the time (sec) to send out Busy Tone with an ISDN line, when called party is busy.

Connects up to six analog loop-start

CS-684, E2-684 — Connects up to six analog loop-start CO lines, eight Digital Feature Phones and four analog station ports. The CO line ports support standard CO and Centrex loop-start lines (but not ground-start CO lines). The analog ports provide a standard 24-volt, two-wire connection to fax machines, courtesy phones, modems, etc. Only one device can be connected to each analog station port. This card uses 12 station ports and six CO ports. • CS-612, E2-612 — Provides circuits to connect up to six analog loop-start CO lines and 12 Digital Feature Phones. Ground-start CO lines are not supported. This card uses 12 station ports and six CO ports. • CS-6ALC, ESI-6ALC — Similar to the CS-612 and E2-612, but connects only up to six analog loop-start CO lines (and no stations). • E2-A41 — Connects up to four analog devices (only), such as fax machines and cordless phones. This card uses four station ports and no CO ports. Each port provides a standard 24-volt, two-wire phone connection. Only one analog device can be connected to each port. • CS-A12, E2-A12 — Connects up to 12 analog devices (only), such as fax machines and cordless phones. This card uses 12 station ports and no CO ports. Each port provides a standard 24-volt, two-wire phone connection. Only one analog device can be connected to each port. • CS-D12, E2-D12 — Connects up to 12 Digital Feature Phones (only). This card uses 12 station ports and no CO ports. • CS-DLC12, E2-DLC12 (Digital Line Card) — Provides either a T1 interface supporting 24 DS0 channels and 12 digital stations or an ISDN PRI interface supporting 23 B (bearer) channels, one D (datalink) channel, and 12 digital stations.  A jumper on this card must be plugged onto pins 7 and 8 of J3 to enable ISDN PRI functions. Any (or all) of the available channels of the T1/PRI span (24 on T1, 23 on PRI) can be assigned, and the card supports loop-start, ground-start, E&M and DNIS/DID trunk types with immediate, wink-start or dial-tone-start signaling. This card is equipped with a built-in CSU that can be connected directly to a network interface unit, SmartJack, or ISDN PRI. Up to 12 Digital Feature Phones can be connected to the card.  All 24 CO ports are allocated (regardless of whether they are assigned or used). • CS-DLC, ESI-DLC — Similar to the CS-DLC12 and E2-DLC12, but supports only a T1 or PRI circuit (and  no phones). • CS-IVC, IVC (Intelligent VoIP Card) — Supports standards-compliant IP telephony service and features, including VoIP to the desktop and Esi-Link. It features highly configurable DSP technology that manages the flow of traffic among the port cards and converts IP packets into PCM (pulse-code modulation) traffic for transmission over the PSTN. The physical connection is a 10/100Base-T, RJ-45 Ethernet® interface that allows the system to connect to an IP-based local area network (LAN).  The IVC is offered in three versions:  •  IVC 24R — Provides 24 IP stations (local or remote).2  •  IVC 24EL — Provides 24 channels for Esi-Link.  •  IVC 12R12EL — Provides 12 IP stations (local or remote) and 12 Esi-Link channels; does not      support SIP phones.  Each ESI Communications Server model has a specific maximum of each type of IVC (see the table on page A.4). The system automatically designates the first IVC station card (lowest-numbered slot) as the primary IVC — which acts as the “master” that, when an IP Phone first comes on line, identifies the IVC station card to which the IP Phone connects (IVC Esi-Link cards are excluded from this operation). Licensing is required to support each IP Feature Phone or SIP phone. The following table shows the maximum number of IP Phones and Esi-Link channels for each system.

Toll Restriction enabled

Deny Restriction Table This option lets you program the Restrict Code Tables. If the system has Toll Restriction enabled, users cannot dial numbers listed in these tables. There are four Restrict Code Tables, with up to 60 entries in each table. The system restricts calls exactly as you enter the code. PBX Access Code Use this option to enter the PBX Access Code. When the system is behind a PBX, this is the code users dial to access a PBX trunk. Toll Restriction begins after the PBX access code. For PBX trunks (Program 14-04) the system only Toll Restricts calls that contain the access code. Always program this option when the system is behind a PBX, even if you don’t want to use Toll Restriction. PBX Access Codes can have up to two digits, using 0-9, #, *  and LINE KEY 1 (don’t care). When using Account Codes, do not use an asterisk in a PBX access code. Otherwise, after the Tables 1 ~ 4 = No Setting [caption: table] 1 ~ 4 (table) 1 ~ 60 (Entry) [caption: table] 1 ~ 4 Dial (Up to 12 digits) Dial (Up to two digits) Tables 1 ~ 4 = No Setting *, the trunk stops sending digits to the central office. Entries 1~4 correspond to the 4 PBX Access Codes. Each code can have up to two digits.

transfer calls

Use this option to prevent or allow extensions to Transfer calls to busy extensions. If disabled, calls transferred to busy extensions recall immediately. Use this option to enable or disable MOH on Transfer. If enabled (0), a transferred caller hears MOH while their call rings the destination extension. If disabled (1), a transferred caller hears ringback while their call rings the destination extension. Default 1 Related Program 1 (V1.5 Changed) 03 04 05 07 08 Delayed Call Forwarding Time Transfer Recall Time Message Wait Ring Interval Time Trunk-to-Trunk Transfer Release Warning Tone Delayed Transfer Time for all Department Groups 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds If activated at an extension, Delayed Call Forwarding occurs after this time. This also sets how long a Transferred call waits at an extension forwarded to Voice Mail before routing to the called extension mailbox. An unanswered transferred call recalls to the extension that initially transferred it after this time. For Single Line Telephones (SLTs) without message waiting lamps, this is the time between intermittent ringing. If this value is set to 0, the system rings once. Time starts when a trunk begins talking with another trunk (for example : trunk-to-trunk transfer, outgoing from trunk, Tandem Trunking). When this time expires, a warning tone is heard. If Program 24-02-10 is set, the conversation disconnects after time expires. This time is set again when the external digit timer expires. One of the trunks used must be an analog trunk (or leased line).

Mounting the cabinet(s) If wall-mounted

Mounting the cabinet(s) If  wall-mounted,  the  system  and  supporting  components  should  be  mounted to  a half-inch (or  thicker)   plywood  backboard.  To  wall-mount  a Base  Cabinet  or  Expansion Cabinet,  use the five  tabs  located at  the rear  of the cabinet.  The  center  tab has an enlarged  hole  and slot,  to  allow you  to  fix  the screw on  the  wall  before hanging the cabinet  onto  the  screw.  Once  you’ve  done  so,  fasten the  other  screws into  the four  remaining  holes to  finish  securing the  cabinet  onto  the wall. To  rack-mount  a Base  Cabinet  or  Expansion Cabinet,  use the forward-facing  screw  holes on  the  sides  of  the cabinet.  Only  two  screws  are  needed  per  side  (in  fact,  on  most  server  racks,  you  can’t  use  all  four  screws  on each side). Allow room  for  installation  of  the Expansion  Cabinet  either  now  or  in  the future;  the Expansion Cabinet  must  be installed  directly  below the Base  Cabinet.  Allow about  two inches  of  clearance  between the  units,  for  cabling.   Attach the  power  transformer  to  the wall  or  rack,  allowing  sufficient  length  in  both  cords to  reach the  power connector  on the front  side  of  the  cabinet  and to  reach a UPS  or  a dedicated  110  VAC outlet.

VRS/DISA

VRS/DISA One-Digit Code Attendant Setup  to set up single digit dialing through the VRS. This gives VRS callers single key access to extensions, the company operator, Department Calling Groups and Voice Mail. For each VRS message set to answer outside calls (refer to Programs 25-04 and 25-05), you specify: • The digit the VRS caller dials (0 ~ 9, *, #). Keep in mind that if you assign destinations to digits, outside callers cannot dial system extensions. • The destination reached (Maximum eight digits  ) when the caller dials the specified digit. The destination can be an extension, a Department Calling pilot number or the Voice Mail master number. A one-digit code can be assigned for each Automated Attendant message. Example: Message Number = 01, Destination = 2, Next Message Number = 0, Dial = 399 In this example, when 2 is dialed by an outside caller, the system transfers the call to 399. This means that extension 200~299 cannot receive calls from VRS/DISA users during/after VRS

transferred caller hears MOH

Use this option to prevent or allow extensions to Transfer calls to busy extensions. If disabled, calls transferred to busy extensions recall immediately. Use this option to enable or disable MOH on Transfer. If enabled (0), a transferred caller hears MOH while their call rings the destination extension. If disabled (1), a transferred caller hears ringback while their call rings the destination extension.

Universal answer

Universal Answer/Auto Answer  to assign trunk routes (set in Program 14-06) to extensions for Universal Answer. If the call ringing the paging system is in an extension assigned route, the user can dial the Universal Answer code (#0) to pick up the call. You can also use this program to let an extension user automatically answer trunk calls that ring other extensions (not their own). When the user lifts the handset, they automatically answer the ringing calls based on Trunk Group Routing programming (defined in Program 14-06). The extension user ringing calls, however, always have priority over calls ringing other co-worker extensions. Refer to the Line Preference feature in the  SL1100  Features and Specifications Manual for more information.

DID Translation Number Conversion

DID Translation Number Conversion  to specify for each Translation Table entry (800). • The digits received by the system (eight maximum) • The extension the system dials after translation (36 digits maximum) • The name that should show on the dialed extension display when it rings (12 characters maximum) • The Transfer Target - 1 and 2If the Transfer Targets are busy or receive no answer, those calls are transferred to the final transfer destination (Program 22-10). • Operation Mode Use the following chart when entering and editing text for names. Press the key once for the first character, twice for the second character, etc. For example, to enter a C, press 2 three times.

ISDN trunk

Enter the number of digits the table expects to receive from the Telco. Use this program to make the system compatible with 3- and 4-digit DID service. If ISDN trunks, we analyze the last digits that are set here. If it is T-1 or analog DID, it analyzes the first digits that are assigned here.

T1 service type

When connecting to T1 trunks, after changing Program 22-02-01 to match the Telco connected T1 service type, the T1 cable or the T1 unit must be unplugged and then reconnected for the T1 unit to sync. • When the trunk type is set to 3 (DID), the DID Transfer to Destination in 22-11-04 for each DID feature is not supported. This feature is supported only for DID trunks when assigned as VRS. • When the trunk type is set to 3 (DID), the DID Intercept Destination feature for each DID is not supported. This feature is supported only for DID trunks assigned as VRS.

Restriction

Use this option to set how the system Toll Restricts calls over PBX trunks. If you enable PBX Toll Restriction, the system begins Toll Restriction after the PBX access code. The user cannot dial a PBX extension. If you disable PBX Toll Restriction, the system only restricts calls that contain the PBX access code. The system does not restrict calls to PBX extensions. Refer to the PBX compatibility feature. Make sure Program 21-05-04 (Maximum Number of Digits Table Assignment) allows for PBX Toll Call Dialing (normally 12 digits). It chooses w