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 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 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 : 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,
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.
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/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) 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.
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.
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.
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.
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, 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 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
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/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 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.
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.
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.
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