VRS Fixed Message 02 General Message Number Input Data 0 = Disable (VRS fixed message will not be played.) 1 = Enable (VRS fixed message will be played.) 0 ~ 100 (0 = No General Message Service) Description Enable (1) or disable (0) the system ability to play the fixed VRS messages (such as You have a message). This item assigns the VRS message number to use for the General Message. Default 0 Related Program 0 03 04 05 06 07 08 VRS No Answer Destination VRS No Answer Time Park and Page Repeat Timer (VRS Msg Resend) Set VRS Message for Private Call Refuse (VRS Msg Private Call) Set VRS Message for Caller ID Refuse (VRS Msg CID) Call Attendant Busy Message 0 ~ 25 (Incoming Ring Group Number) 0 ~ 64800 seconds 0 ~ 64800 seconds 0 = No Message Played 1 ~ 100 = VRS Message 1 ~ 100 101 = VRS Fixed message (Message will only play if PRG 40-10-01 is enabled.) 0 = No Message Played 1 ~ 100 = VRS Message 1 ~ 100 101 = VRS Fixed message (Message will only play if PRG 40-10-01 is enabled.) 0 ~ 100 (0 = No message) This item assigns the transferred Ring Group when the VRS is unanswered after Call Forwarding with Personal Greeting Message. If an extension has Personal Greeting enabled and all VRS ports are busy, a DIL or DISA call to the extension waits this time for a VRS port to become free. If a Park and Page is not picked up during this time, the Paging announcement repeats. This item assigns the VRS Message number to be used as Private Call Refuse. When Fixed message is set, VRS message guidance is: “Your call cannot go through.” This item assigns the VRS Message number to be used as Caller ID Refuse. When Fixed Message is set, VRS message guidance is: “Your call cannot go through.” 0 (No Setting) 0 0 0 0 0
While a caller waits for a UCD group member to answer (in queue), several queued call operations are possible. These include the following. • No Answer – Member advancement. Each UCD group has a No Answer Timer. You can set this timer to advance the call from one UCD group member to the next when the ringing member does not answer the call within the time allotted. • Overflow 1 Destination programming. Each UCD group has an overflow 1 timer and destination. You can set this timer to determine how long calls will remain in queue, before being routed to the overflow 1 destination. The destination can be an extension responsible for handling calls that remain in queue too long, or a voice announcement device. You can use recorded announcement devices to play recorded messages to callers waiting in queue, for example, “All agents are still busy - please continue to hold.” The system plays the overflow 1 destination only once. For high traffic scenarios, you can use a recorded voice announcement UCD group to play the same message to multiple callers. • Overflow 2 Destination programming. Each UCD group has an overflow 2 timer and destination. You can set this timer to determine how long calls will remain in queue, following the overflow 1 timer, before being routed to the overflow 2 destination (overflow 1 timer + overflow 1 destination recorded message time + overflow 2 timer.) The destination can be an extension responsible for handling calls that remain in queue too long or a voice announcement device. You can use recorded announcement devices to play recorded messages to callers waiting in queue, for example, “Please continue to hold to reserve your place in queue.” For high traffic scenarios, you can use a recorded voice announcement UCD group to play the same message to multiple callers. • Overflow Count programming. Each UCD group has an overflow count that is associated with the overflow 2 timer. You can set this timer to allow a specific number of times that the system may repeat the overflow 2 timer. For each cycle of the overflow 2 timer, the system plays the overflow 2 destination recording. If a call remains in queue so long that the overflow count counter expires, the system routes this call to the programmed reroute destination. • Reroute Destination programming. Each UCD Group has a reroute destination. You can program this with an extension number the system uses to remove the call from UCD group queue. The system then routes the call for immediate handling. Other features whose programming may affect UCD programming includes the following. • CO Line Ring Assignment • UCD Reroute Destination • UCD Voice Announce Group • UCD Agent Log On/Log Off • Voice Mail - Digital Integration • Recorded Announcement Devices (RADs)
Receive Format Delimiter Dial Code Route Setup of Receive Dial 0 = Address 1 = * ANI 2 = * * DNIS 3 = * ANI * * Address 4 = 5 = ( * ANI * DNIS * DNIS * ANI * = Delimiter Code) 1 ~ 9, 0, #, * 0 = Fixed Route (Item 08) (No Routing) 1 = Routes on Received DNIS or Address Data 2 = Routes on Received ANI Data COS 01 = 0 COS 02 ~ 15 = 0 * * * Use this option to specify the format of the ANI/DNIS data received from the Telco. Make sure your entry is compatible with the service the Telco provides. The character * indicates a delimiter.If Program 34-01-02 is selected to 2 (MF), this Program works only as 4 = *ANI *DNIS *. This option defines the character Telco uses as a delimiter (see entries 1 ~ 5 in Item 1 above). Valid entries are 0 ~ 9, #, and *. This option specifies the source of the data the system uses to route incoming ANI/DNIS calls. If option 2 is selected, refer to Program 34-09-04
1. Place line 4 (743) into CO group2 (800). 2. From the Fax machine, go off hook and dial 800 (be sure you are accessing line 4). Then dial a valid number. You must dial out on that line /(line group) before programming the hot line (dial 740 for line 1, 741 for line 2, 742 for line 3 etc.…). 3. Program a System Speed dial bin with a pause. To do this, you can use PCDBA or program using the phone. Using PCDBA Go to the DX80 PC-DBA System Resource SPD. No. Programming Tenant 1screen in PCDBA. Choose a system speed dial bin (DIR#s 500-699), and insert a P for pause in its SPD. No. field.
60-Key Expansion Console, B.3, I.8, I.9 60-Key Second Expansion Console, B.3, I.8, I.9 Analog ports, I.7 Battery. See Cautions Cabinets Expansion, F.2 Cautions, E.1 Battery, E.1 Fuse, E.1 Power supply, E.1 CO lines Capacities. See System capacities Connecting, I.5 Console, B.3, I.8, I.9 ESI Cordless Handsets. See Phones ESI Presence Management, D.1 Expansion Cabinet, F.2 Expansion Console, B.3, I.8, I.9 Fuse. See Cautions Grounding, F.2, I.1, I.3 Hardware installation, E.2–F.15 LED functions, F.15, G.6, H.6 Main board, A.2 Memory Module, A.3 Installation or replacement, F.5–F.8, G.5, H.5 Mirrored Memory Module (M3), A.3 Installation, F.9–F.13 MOH, I.3 NSP (Network Services Processor), A.7 Overlays, B.4 Paging, I.4 Phones Digital Feature Phones, B.1 ESI Cordless Handsets, B.2, B.3 IP Phones, B.2 VIP Softphone, B.4 Port cards Capacities, A.4 Charts, I.13–I.20 Installation, G.3–G.4, G.3–G.4 Installation, F.2 Port card adapter, F.3 Power, I.1 Power Distribution Shelf, A.3 Power supply. See Cautions Transformers, wall-mount, A.3 PRI, I.5 Regulatory information (U.S. and Canada), E.2 Ringer equivalence number (REN), E.2 Serial ports, I.3 Site location, F.1 SMDR, I.3 System capacities, D.1 T1, I.5 UPS (uninterruptible power supply), I.1 VIP Softphone.
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 (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.
Registration Expire Timer Input Data 60 ~ 65535 seconds Description The Expires value of the REGISTER message which received from DR700 terminal is out of range or when the Expire value is not set up, in case it assigns the effective time to the DR700 terminal. The timer for supervising whether DR700 terminal is connected or not. Default 180 02 03 04 05 06 07 08 09 Subscribe Expire Timer 60 ~ 65535 seconds Session Expire Timer 60 ~ 65535 seconds Minimum Session Expire Timer Invite Expire Timer Signal Type of Service Error Display Timer Digest Authorization Registration Expire Timer 60 ~ 65535 seconds 60 ~ 65535 seconds 0x00 ~ 0xFF (0 ~ 9, A ~ F) 0 ~ 65535 seconds 0 ~ 4294967295 seconds Temporally Password Read Only: Maximum 16 characters (0 ~ 9, a ~ f, A ~ F) The subscribe Expire timer to transmit and receive the terminal operation instructions between the Main Device and DR700 terminal. Set effective time for supervising the Voice Path. Set minimum value of effective time for supervising the Voice Path. Set effective time for Incoming/Outgoing call when the Expire value is not set in the INVITE message received from DR700 terminal. Set Type of Service value which applied to send SIP Message Pac
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
LED functions The unit's various LEDs are designed to provide visual feedback as follows: Power LED The Power LED is located on the right side of the Base Cabinet, and is illuminated when power is being applied to the system. This LED blinks periodically to indicate that the main processor is operational. Port LEDs The Port LEDs are located above their respective connectors on each installed port card. Each LED is illuminated when any port on its associated port card is in use. Note: Disconnecting a connector when its respective LED is lit will disconnect any of its ports that are in use. Upon power-up, approximately five minutes are required for the system to configure. The Power and Port LEDs will blink three times to indicate that the power-up sequence has been completed. Note: When the LED on a DLC1 is . . . • . . . blinking, the T1/PRI circuit is out of service. • . . . not lit at all, the T1/PRI circuit is in service but is idle. • . . . lit solidly, the T1/PRI circuit and/or a station on the card are in use. ESI Presence Management installation For information on installing ESI Presence Management, see its Installation Manu
Number of G.711 Audio Frame 02 G.711 Silence Detection (VAD) Mode Input Data 1 = 10 ms 2 = 20 ms 3 = 30 ms 4 = 40 ms 0 = Disable 1 = Enable Description Maximum number of G711 Audio Frames. When the voice is encoded using the PCM (Pulse Code Modulation) method, a unit is a frame of 10ms. Select whether to compress silence with G.711. When there is silence, the RTP packet is not sent. Default 2 Related Program 0 03 04 05 06 07 08 09 10 11 12 14 G.711 Type G.711 Jitter Buffer - Minimum G.711 Jitter Buffer - Minimum G.711 Jitter Buffer - Maximum G.729 Audio Frame G.729 Silence Compression (VAD) Mode G.729 Jitter Buffer - Minimum G.729 Jitter Buffer - Standard G.729 Jitter Buffer - Maximum Number of G.723 Audio Frame G.723 Jitter Buffer - Minimum 0 = A-law 1 = μ-law 0 ~ 255 ms 0 ~ 255 ms 0 ~ 255 ms 1 ~ 6 (1 = 10 ms, 2 = 20 ms, etc.) 0 = Disable 1 = Enable 0 ~ 300 ms 0 ~ 300 ms 0 ~ 300 ms 1 = 30 msec 2 = 60 msec 0 ~ 300 ms Set the type of G.711. Set the minimum value of the G.711 Jitter Buffer. Set the average value of the G.711 Jitter Buffer. Set the maximum value of the G. 711 Jitter Buffer. Maximum number of G729 Audio Frames. G.729 assumes the audio signal made by a specimen by 8 kHz and the frame of 10 ms is assumed to be a unit to 8 kbps by the encoding compressed method. Select whether to compress silence with G.729. When there is silence, the RTP packet is not sent. Set the minimum value of the Jitter Buffer of G.729 is set. Jitter is the variation in the time between packets arriving and the buffer allows this variation to be absorbed. Set the average G.729 Jitter Buffer. Set the maximum G.729 Jitter Buffer. Maximum number of the G.723 Audio Frame. Set the minimum value of the G.723 Jitter Buffer.
Bounce Protect Time 0 = No setting 1 ~ 15 = 100 ms ~ 1.5 sec Description Specify a time for detection of a valid offHook indication that is long enough to prevent an unintentional bounce of the receiver from being detected as a new Off-Hook indication from a Single Line Telephone. Default 3 (300 ms) 02 03 HookFlash Start Time 0 = 40 ms 1 ~ 15 = 90 ms ~ 790 ms HookFlash End Time 0 = HST + 0 ms 1 ~ 15 = HST + 100 ms ~ HST + 1500 ms (HST = Hookflash Start Time) Conditions None Specify the minimum hookflash time from a Single Line Telephone or analog Voice Mail system before it is detected as the beginning of a valid hookflash. Specify the maximum hookflash duration from a Single Line Telephone to receive a second dial tone.
Remote connection to the DX-80 system is possible via modem. The optional modem may be purchased allowing remote administration of the DX-80 system database and maintenance operations. The default directory number of the modem is 199. Some working knowledge of modem operation and connection is useful. 220.127.116.11 F10-RS232C This function key is used to setup the PC COM port. To successfully setup the PC COM port you must know how the PC hardware is configured. In this utility you must select the COM Port number (PCDBA supports COM 1 or 2 only) and the baud rate that will be used for the connection. At default PCDBA is setup to use COM Port 1 at 9600 bps. (9600 bps matches the default baud rate set for the DX80 CPM – PC-DBA Port.) While using a modem connection, it is best to set the COM port baud rate at 2400 bps since this is the speed of the DX-80 optional modem; setting this speed can expedite the modem negotiation process since compression link choices will not be attempted.
System Maintenance Maintaining the Comdial DX-80 digital telephone system is a combination of customer database changes, facilities and apparatus moves, adds and changes. These requirements are accomplished by practicing the techniques, illustrations and step-by-step instructions listed in the previous sections of this manual. When properly installed, the Comdial DX-80 is relatively maintenance-free. From time to time the digital telephone instruments may become dirty or dusty and require cleaning. We suggest the use of a clean, dry cotton (or other soft, absorbent) cloth to wipe the instrument clean. The use of chemicals to clean the telephone plastics is NOT recommended since some chemicals can cause permanent damage to the telephone finish. If deep soiling conditions exist, many specialized telephone cleaning solutions will provide satisfactory results. When trying any cleaner for the first time, apply the cleaner to a small sample area on the underside of the instrument. If the expected results are achieved, proceed with cleaning the remainder of the telephone.
Connecting a Serial Cable for SMDR SMDR (Station Message Detail Recording) can be output from the DX-80 system for use with serial printers of collection in call accounting devices. Connection of the SMDR device to the DX-80 is accomplished through the serial data port on the CPM labeled “SMDR.” Connection to serial printers may require customization of the serial cable used to make the connection. CPM“SMDR” serial port connector (straight- (Dedicated means that this AC outlet has no other equipment connected on this circuit breaker. Whenever a call accounting system (third-party device) is deployed, connection is often no more complicated than using a straight-through, 9-pin, female to male, serial cable. (Use Radio Shack model 26117B for good results.) Connection to the PC/call accounting system is made via an available 9-pin serial port connector that is designated as COM1 or COM2 in the PC configuration. Once the cable is linked between the collection device and the DX-80 CPM-SMDR port, the collection device must be programmed for compatible link protocol (baud rate). At default the SMDR port baud rate is set at 9600 bps.
Assigning a Password to the Extension All extensions of the DX-80 system have an associated user password. Passwords are required to use Phone Lock, Call Forward Remote, and Attendant features. Note: Extension passwords can be changed at the extension only by using the Phone Lock feature. Passwords can be from four to eight characters in length. You may want to program the overall system password length before programming individual extension passwords. See Section 4.4, Setting User Password Lengths, Setting Passwords for System Directory Numbers 101, 102, and 108 for more details. Changes to the length of the user password affect existing passwords—the system adds or subtracts one default character to the end of the password. That is, if you extend the length of the password, the system increases all passwords by adding a 0 in the right-most position. If you reduce the length of the password, the system truncates all passwords by one character in the right-most position.
The system prompts “Save Current Setting?” Press Y to save your changes. 9. The system then prompts “File Exists, Override?” Press Y to continue with the save operation. 10. Press Esc to return to the Uniform Call Distribution: Parameter Processing: Hunt Group 24 menu. Press Esc three times more to return to the Database Programming menu. 11. Next, program the voice mail hunt group. Highlight Voice Mail Table, and press Enter. Choose the tenant group you want to program, and press Enter. The system displays the Extension Application—Voice Mail Table: Tenant x menu.
You can add one AAM to the KSU1. The AA Module adds automated attendant functionality to the DX-80 system with 10 integrated announcements. The AAM does not provide voice mail functions. For details on how to program the AAM and its announcements, refer to the DX-80 Technical Manual, Volume II, Programming. Install the AAM onto designated connectors located on the CPM. To install the AAM perform the following steps. 1. Be sure that the entire system is turned off. 2. Remove the KSU cover (four screws at each corner). 3. Connect a static discharge wrist strap to a suitable earth ground. Be sure that the strap is touching bare skin.
Programming Third Party VM, Analog Ports You can connect the DX-80 system to a third-party voice mail system using spare analog ports. Doing so occupies these ports and therefore reduces the number of system ports that you can use for the telephones, FAX machines, modems, etc. Comdial recommends using the DX-80 DX-SO voice mail system, because it is a digital integration and therefore does not occupy valuable analog port space of the DX-80 system. For further details, see Section 9.2, Programming Optional Internal VM, Digital. Using voice mail greatly enhances the use of the DX-80 system. Features accessible when voice mail is installed vary depending upon the third-party product connected. Features that the DX-80 system accommodates include the following. • Automated Attendant • Extension unique voice mailboxes • Call Forward to extension voice mailboxes • Answering Machine Emulation • UCD Queue Announcements • Menu Routing • Voice Record • Automatic Voice Record • Specific CO Line Greetings on Automated Attendant The following conditions apply to analog voice mail. • Voice mail feature operation is limited only by the ancillary voice mail system. • When voice mail ports are used heavily (high call traffic), the system updates message indications notably slower than usual. • When VM messages are waiting, the system flashes the auxiliary lamp green.
Adding a COM4 You can add one COM4 to KSU1 and one COM4 to the KSU2. These modules expand the DX-80 system CO line interface capacity to a maximum of eight CO line ports in each KSU (16 total CO line ports when used in KSU 1 & 2). When adding a COM4 to either KSU, connect to the 408M/E via the COM4 module ribbon cable. This ribbon cable has the logical system address of SLOT 5 in both cabinet 1 (KSU1-408M) and cabinet 2 (KSU2-408E). COM4 modules are installed using four 1.5 cm brass-color standoffs. Always install the COM4 module beneath the CPM module (in KSU1) by first removing the CPM, installing the COM4, and then re-installing the CPM. This is required whenever you install the InSkin Voice Processor.
The 408M is equipped with a heartbeat LED that indicates processing activity on the PCB. (The 408M peripheral processor is operating when the heartbeat LED is flashing.) The KSU1 operation LED (located next to the power switch) is tied to the 408M heartbeat LED. Therefore, when the LED next to the power switch is flashing, the 408M is active. The KSU1-408M has three ribbon cables: • located at the upper right and oriented in a horizontal position, the J1 cable is used to interface a COM4 if required to expand the system CO line capacity. • located at the upper center and oriented in a vertical position, the J4 cable is used to interface the DX-80 CPM. • located at the upper center and oriented in a vertical position, the J5 cable connects to the standard APM4 installed in KSU1. Each CO line circuit incorporates over-voltage protection, ring detector, loop detector, loop/pulse-dial relay, current sink circuit, coupling/isolation transformer (impedance 600:600), hybrid circuit, CODEC & filter, polarity guard circuit and Radio Frequency noise filter. The fourth CO line port is equipped with CNG Fax Tone Detection circuitry. When programmed as a “FAX” line, this circuit will automatically engage the FAX Tone detector. If FAX tone is detected, the system routes the call to the analog port designated as the destination for fax calls.
CPM (Central Processor Module) The CPM module is equipped standard in KSU1. This board contains all circuitry required to control the fully equipped DX-80. The system uses the CPM to perform all digital voice switching and call processing data switching. The CPM has one ribbon cable connector for connection to the KSU1 408M and five (5) connector sockets for connection of the system built-in modem, voice processor, and second cabinet (KSU2). Since the CPM comes installed inside of KSU1 the CPM ribbon cable is already in place and connected to the KSU1-408M J5 socket. Assuming the orientation of the KSU1 cabinet is installed on the wall; the two horizontal connector sockets in the upper right corner of the CPM are for the MDM (Modem Module). The connector socket labeled “2nd Cabinet” is for connection to the KSU2-408E if that expansion is required. The remaining two connector sockets on the CPM, one at the left side, the other at the right side are for the voice processor solution. Note: The voice processor solution can be any of three possible choices: AAM, 7271C, or 7270C.
Call ForwardAll/No Answer/ BothRing 0 = Call Forwarding off 1 = Call Forwarding with Both Ringing 2 = Call Forwarding when No Answer 3 = Call Forwarding All Call Up to 8 digits Input Data Description Read Only: Indicates Call Forward-All/No Answer/BothRing setting statsus per extension. Default Related Program None 11-11-01 11-11-03 11-11-04 11-11-05 02 03 04 05 06 07 08 09 Call Forwarding Destination for Both Ring, All Call, No Answer Call ForwardBusy Call Forwarding Busy destination Call Forwarding– Follow-Me Call Forwarding Follow-Me destination Do Not Disturb 0 ~ 9, *, #, P, R, @ (Up to 36 digits) 0 = Call Forward-Off 1 = Call Forward-Busy or No answer 2 = Call Forward-Busy 0 ~ 9, *, #, P, R, @ (Up to 36 digits) 0 = Disable 1 = Enable Extension Number (Up to 8 digits) Read Only: Indicates Call Forward-All/No Answer/BothRing destination number set per extension. Read Only: Indicates Call Forward-Busy setting status per extension. None None Read Only: Indicates Call Forward-Busy destination number set per extension. Read Only: Indicates Call Forward-Follow-Me setting status per extension. Read Only: Indicates Call forwarding follow-me extension number set per extension. 0 = No Setting 1 = DND External 2 = DND intercom 3 = DND Transfer 4 = DND All Message Waiting (Set) Message Waiting (Rec) Extension Number (Up to 8 digits) Extension Number (Up to 8 digits) Read Only: Indicates DND setting status per extension. Read Only: Indicates extension number which you set Message Waiting. Read Only: Indicates extension number when left Message Waiting.
KSU1 COMPONENTS The CPM (Central Processor Module) is installed inside of KSU1 to the 408M ribbon cable J4 (also labeled “To CPM”). The CPM contains: • Two serial ports, • Two music ports, • One control contact (external paging/LBC/gate control), • One external page equipment interface connection, and • Socket connectors for the KSU2 (labeled “2nd Cabinet”), MDM, and VP modules (AAM, 7271C, and 7270C). The standard 408M (part of 7201) contains: • 4 CO line ports, • 1 power failure transfer port for the first CO line circuit, • 8 digital ports, • a ribbon cable (J1, also labeled “COM4”) for connection to the COM4 module (PN 7210) in KSU1, and • ribbon cable sockets (J2 and J3, also labeled “To DPM8/APM4”) for connection of DPM8 (PN 7220) or APM4 (PN 7230) modules. The standard APM4 (part of 7201) contains four analog device ports (installed on ribbon cable J5, also labeled “APM4”). 1.2.2 KSU2 COMPONENTS The standard 408E contains: • 4 CO line ports • 1 power failure transfer port for the first CO line circuit • 8 digital ports • a ribbon cable (J1, also labeled “COM4”) for connection to the COM4 module (PN 7210), • ribbon cable sockets (J2 and J3, also labeled “To DPM8/APM4”) for connection of DPM8 (PN 7220) or APM4 (PN 7230) modules, and • a shielded cable (J4) for connection to CPM socket JP2 (also labeled “2nd Cabinet”) in KSU1.
Memory Module installation or replacement
Note: The Memory Module has a proprietary formatting scheme — do not attempt to install a non-ESI drive.
Contact ESI for a replacement Memory Module, if needed.
Adding or replacing the Memory Module will require that the ESI-100 be taken out of service.
All of the ESI-100’s configuration data and customer recordings are stored in the Memory Module.
Replacing it, therefore, requires re-programming and re-recording, unless you have previously performed a backup
using ESI System Programmer software. (Prompts stay intact, however.)
Note: Be sure to observe all proper procedures regarding the prevention of electrostatic discharge (ESD) when
performing the following procedures; otherwise, circuit boards may suffer damage.
Install the CompactFlash Memory Module
1. Open the lid of the cabinet (you must remove
the screw on the top that secures the lid).
2. Power down the system.
3. Plug the Memory Module into the J14 connector
on the main board (see diagram, right).
4. Secure the lid to the KSU
: Service Code Setup (for System Administrator) to customize the Service Codes for the System Administrator. You can customize additional Service Codes in Programs 11-11 ~ 11-16. The following chart shows: • The number of each code (01 ~ 50). • The function of the Service Code. • The type of telephones that can use the Service Code. • The default entry. For example, dialing item 26 allows users to force a trunk line to disconnect. Input Data Item No. 01 Item Night Mode Switching 0~9, Input Data *, # Maximum of 8 digit Description Terminal: MLT, SLT Default 718 Related Program 12-xx 20-07-01 03 04 05 06 07 08 09 11 12 Setting the System Time Storing Common Speed Dialing Numbers Storing Group Speed Dialing Numbers Setting the Automatic Transfer for Each Trunk Line Canceling the Automatic Transfer for Each Trunk Line Setting the Destination for Automatic Trunk Transfer Charging Cost Display by the Supervisor Entry Credit for Toll Restriction Night Mode Switching for Other Group 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit 0~9, *, # Maximum of 8 digit Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT Terminal: MLT 728 753 754 733 734 735 No Setting No Setting 618 24-04-01 24-04-01 24-04-01 12-xx 2
System Numbering to set the system numbering plan. The numbering plan assigns the first and second digits dialed and affects the digits an extension user must dial to access other extensions and features, such as service codes and trunk codes. If the default numbering plan does not meet the site requirements, use this program to tailor the system numbering to the site. Caution! Improperly programming this option can adversely affect system operation. Make sure you thoroughly understand the default numbering plan before proceeding. If you must change the standard numbering, use the chart for Table 2-1 System Numbering Default Settings on page 2-57 to keep careful and accurate records of your changes. Before changing your numbering plan, use PC Pro to make a backup copy of your system data. Changing the numbering plan consists of three steps: Step 1 : Enter the digit (s) you want to change You can make either single or two digit entries. In the Dialed Number column in the Table 2-1 System Numbering Default Settings on page 2-57, the nX rows (e.g., 1X) are for single digit codes. The remaining rows (e.g., 11, 12, etc.) are for two digit codes. • Entering a single digit affects all the Dialed Number entries beginning with that digit. For example, entering 6 affects all number plan entries beginning with 6. The entries you make in step 2 and step 3 below affect the entire range of numbers beginning with 6. (For example, if you enter 3 in step 2 the entries affected are 600 ~ 699. If you enter 4 in step 2 below, the entries affected are 6000 ~ 6999.) • Entering two digits lets you define codes based on the first two digits a user dials. For example, entering 60 allows you to define the function of all codes beginning with 60. In the default program, only * and # use 2-digit codes. All the other codes are single digit. If you enter a two digit code between 0 and 9, be sure to make separate entries for all the other two digit codes within the range as well. This is because in the default program all the two digit codes between 0 and 9 are undefined.
ACD department programming ACD departments can be programmed to route calls based on several optional parameters. Each ACD station can be logged onto as many as two ACD departments at the same time. Also, each ACD station can be a member of up to 20 ACD departments (i.e., up to 20 log-on keys may be assigned to each ACD station.) Each ACD agent must have a Digital Feature Phone1, IP Feature Phone II, Digital Cordless Handset, or VIP Softphone. Agent log-on keys will be automatically assigned to the lower left programmable feature keys for the stations listed in ACD departments (with wrap keys automatically assigned above them; see “Feature keys,” page G.37). Note: Line keys can’t be used to answer calls ringing Attend departments set in the live-ring list (in Functions 211, 2121, and 2131). ACD overflow Incoming calls that are holding (queued) for an available agent can be automatically forwarded to a new destination if ACD overflow is assigned. ACD overflow can be initiated by exceeding a maximum number of queued calls or by an individual exit timer set for each ACD department. If ACD overflow parameters aren’t assigned, the default action will be to overflow calls based on the system default ACD exit timer only. ACD overflow parameters are: • Queue exit threshold — If the number of calls in queue matches a predetermined queue exit threshold, all subsequent calls to that ACD department will immediately follow that department’s call-forward destination. • ACD exit timer — When a call has been held in queue for a predetermined duration specified for that ACD department, the call will follow the department call-forward destination. If the department’s ACD exit timer isn’t assigned, the system-wide default (for the current operation) will be used. ACD agent priority ACD agents who are simultaneously logged into two departments can have calls to one department take precedence over the other department’s calls. When the ACD station is assigned to each ACD department, the “baseline” priority is set for calls that are directed to that station from that department. ACD call escalation (priority override) An incoming call that has been in an ACD department queue the longest can be forced to ring at the next available agent, regardless of the priority setting of that agent’s station department log-in. A timer that’s set in ACD department programming (see page G.27) triggers this ACD escalation.
Network Keep Alive Setup to set the interval and retry count of the AspireNet networking keep alive message. The keep alive is used for ISDN and IP networking. The keep alive message is automatically responded to by the destination system, if the response is not received the retry count will start. If a response is not received within the number of retries, the networking link will be taken out of service. When the link is taken out of service: • Any calls that are in progress will be released. • Park Hold orbits will be released. • No further Park Hold information will be sent until the link is active. The link will automatically become active when the next keep alive response is received.
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
Select the trunk based off the Trunk Route Priority (0) or based off the trunk that has not been used in the longest time (1). Default 0 Related Program 14-05 14-06 02 03 04 05 06 07 08 09 Intercom Interdigit Time Trunk Interdigit Time (External) Dial Tone Detection Time Disconnect Time when Dial Tone not Detected Dial Pause at First Digit Toll Restriction Override Time Preset Dial Display Hold Time Ringdown Extension Timer (Hotline Start) 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds 0 ~ 64800 seconds When placing Intercom calls, extension users must dial each digit in this time. The system waits for this time to expire before placing the call in a talk state (Call Timer starts after time expires, Voice Over and Barge-In is not allowed until after time expires). If dial tone detection is enabled, the system waits this time for the Telco to return dial tone. When the time expires, the system assumes dial tone is not present. To disable this time (and have the system wait continuously), enter 0. If 14-02-11 is enabled, the system skips over a trunk if dial tone is not detected. This option pertains to calls placed using Speed Dial, ARS, Last Number Redial or Save Number dialed. It does not pertain to line key or Direct Trunk Access calls. After dialing the Toll Restriction Override codes, the system removes Toll Restriction from the extension for this time. A Ringdown extension automatically calls its programmed destination after this time.
To achieve optimum operation from your ESI Cordless Handset II Repeater: • Place the Repeater at least six feet off the ground so it has a clear line-of-sight. • Make sure the Repeater has good reception from the base station (or Repeater to which it is daisy-chained). • Make sure the Repeater location is close to a standard 120 VAC power outlet. • Never install electrical cords across traffic areas where they can cause a tripping hazard (additionally, such cords, if damaged, may create fire or electrical hazards). • Allow at least 35 feet between Repeaters. If you install Repeaters across multiple floors, be sure to allow 35 feet vertically, too. • Install the Repeater away from sources of electrical interference. Examples include audio systems, office equipment, and microwave ovens. • Install the Repeater away from heat sources and direct sunlight. • Install the Repeater away from items that can interfere with radio signals. Examples include metal doors, thick walls, niches, and cupboards.
An Avaya System SD card must be present in this slot at all times. This card holds copies of the IP Office firmware and configuration and is used as the IP500v2 control units non-volatile memory. · Each Avaya System SD card has a unique Feature Key serial number which is used for generating and validating licenses entered into the IP Office configuration. · The card stores the prompts for embedded voicemail operation and acts as the message store for embedded voicemail messages. · Prior to any planned shutdown or restart of the IP Office system, the current configuration running in the IP Office system's RAM memory is copied to the /primary folder on the System SD card and to the systems nonvolatile memory. · Following a restart, the software in the /primary folder is loaded by the IP500v2 control unit. If the required software is not present or valid a sequence of fallback options is used, see Booting from the SD Cards 155 for full details. · Following a restart, if present, the configuration file in the /primary folder is loaded by the IP500v2 control unit. If no file is present the system will check for a file in its internal non-volatile memory. If no copy is found it will generate a default configuration file. See Booting from the SD Cards 155 for full details. · Once each day (approximately between 00:00 and 00:30) the IP Office will copy the current configuration running in its RAM memory to the /primary folder on the card. · Configuration changes made using IP Office Manager are first written to the copy of the configuration file on the card and then merged with the configuration running in the IP Office system's RAM memory. · The write lock setting on cards in the System SD card slot is ignored. · Optional SD Card A card does not have to be present in this slot for normal IP Office operation. The slot can be used for various maintenance actions.
60-Key Expansion Console, B.3, I.8, I.9 60-Key Second Expansion Console, B.3, I.8, I.9 Analog ports, I.7 Battery. See Cautions Cabinets Expansion, F.2 Cautions, E.1 Battery, E.1 Fuse, E.1 Power supply, E.1 CO lines Capacities. See System capacities Connecting, I.5 Console, B.3, I.8, I.9 ESI Cordless Handsets. See Phones ESI Presence Management, D.1 Expansion Cabinet, F.2 Expansion Console, B.3, I.8, I.9 Fuse. See Cautions Grounding, F.2, I.1, I.3 Hardware installation, E.2–F.15 LED functions, F.15, G.6, H.6 Main board, A.2 Memory Module, A.3 Installation or replacement, F.5–F.8, G.5, H.5 Mirrored Memory Module (M3), A.3 Installation, F.9–F.13 MOH, I.3 NSP (Network Services Processor), A.7 Overlays, B.4 Paging, I.4 Phones Digital Feature Phones, B.1 ESI Cordless Handsets, B.2, B.3 IP Phones, B.2 VIP Softphone, B.4 Port cards Capacities, A.4 Charts, I.13–I.20 Installation, G.3–G.4, G.3–G.4 Installation, F.2 Port card adapter, F.3 Power, I.1 Power Distribution Shelf, A.3 Power supply. See Cautions Transformers, wall-mount, A.3 PRI, I.5 Regulatory information (U.S. and Canada), E.2 Ringer equivalence number (REN), E.2 Serial ports, I.3 Site location, F.1 SMDR, I.3 System capacities, D.1 T1, I.5 UPS (uninterruptible power supply), I.1 VIP Softphone. See Phones
For tie lines, enable or disable the ability to absorb (ignore) the first incoming digit. Use this to make the tie trunk compatible with 3- and 4-digit tie line service. This option does not apply to DISA.
This option enables or disables a DISA or tie trunk caller ability to dial 9 for Trunk Group Routing or Automatic Route Selection (ARS/F-Route).
Virtual Extension Ring Assignment to assign the ringing options for an extension Virtual Extension Key or Virtual Extension Group Answer Key which is defined in Program 15-07. You make an assignment for each Night Service Mode. Assign extension numbers and names to virtual extension ports in Program 15-01. Program Virtual Extension keys in Program 15-07 (code *03). There are 50 Virtual Extension Ports.
Speed Dialing Trunk Group to define the trunk group to be seized for each Speed Dialing number. If this program has an entry of 0 (no setting), then seizing a line follows the trunk access group routing of the caller’s extension (refer to Program 14-06). This setting is available only in External Speed Dialing Mode
(with Esi-Link) CO line 1 answer ring destination First ring — Line 1 (optionally named “SALES”) rings at operator’s extension. Third ring — Extension 112 at Location 702 is added. Fifth ring — Extension 100 and Location 702 extension 112 stop ringing; and Location 703 extension 101 starts ringing. Ninth ring (or no available Esi-Link channels for Ring 5) — Call is answered by auto attendant. CO line 2 answer ring destination First ring — Line 2 (optionally named “MFG.”) rings at extensions 118–119. Third ring — Line 2 rings at Department 290 in Esi-Link Location 702. Fifth ring — (In this example, Ring 5 isn’t programmed. If an Esi-Link connection to Location 702 is available, the call routing will follow the call forwarding for Department 290.) CO line 3 answer ring destination First ring — Line 3 (optionally named “TECH”) is answered by auto attendant (branch ID 4) in home location. Branch ID 4 is assigned as a GoTo: Remote branch to an ID branch at Locati
Telephone line powered • Weather resistant • Adjustable microphone and speaker volume (preset for optimal operation) • Call time out, to limit prank calls and false alarms (approximately 30 seconds) • Responds to forward disconnect • Auto-answer feature allows remote monitoring (can be disabled) • Two way handsfree communication • Vandal resistant brushed stainless steel face plate with mounting gasket • Flush mountable using included rough-in box
Service Code Setup, Administrative (for Special Access) to customize the special access Service Codes which are used by the administrator in the Hotel/Motel feature. You can customize additional Service Codes in Programs 11-10 ~ 11-14 and 11-16. The following chart shows: • The number of each code (01 ~ 14). • The function of the Service Code. • What type of telephones can use the Service Code. • The default entry. • Programs that may be affected when changing the code.
Activation Key Code and Key Management System To obtain additional activation keys, you need to purchase the appropriate activation key codes and access the Key Management System. You can download the activation keys as an activation key file from the Key Management System. To download the activation keys, enter the MPR ID number shown on the IPCMPR card in the PBX, and activation key number and registration ID provided on each activation key code. For information about the type of activation key codes available, refer to "Additional Activation Keys in the SD Memory Card (Activation Key Files)". Note • You can only download the activation key file once using the activation key number and registration ID provided on the activation key code. • Up to 8 activation keys can be downloaded as one activation key file. • Up to 100 activation key files can be installed on the SD Memory Card. • It is possible to send the activation key file to a specified e-mail address at the same time as downloading it to a PC. • Make sure to backup the downloaded activation key files on your PC. • In the event of a system malfunction, you need a temporary activation key for maintenance purposes. The temporary activation key can only be used for a limited time period, and can be downloaded from the Key Management System in the same way as downloading activation key files.
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. (See “Worksheet” wiring charts,
System Numbering to set the system numbering plan. The numbering plan assigns the first and second digits dialed and affects the digits an extension user must dial to access other extensions and features, such as service codes and trunk codes. If the default numbering plan does not meet the site requirements, use this program to tailor the system numbering to the site. Caution! Improperly programming this option can adversely affect system operation. Make sure you thoroughly understand the default numbering plan before proceeding. If you must change the standard numbering, use the chart for Table 2-1 System Numbering Default Settings on page 2-57 to keep careful and accurate records of your changes. Before changing your numbering plan, use PC Pro to make a backup copy of your system data. Changing the numbering plan consists of three steps: Step 1 : Enter the digit (s) you want to change You can make either single or two digit entries. In the Dialed Number column in the Table 2-1 System Numbering Default Settings on page 2-57, the nX rows (e.g., 1X) are for single digit codes. The remaining rows (e.g., 11, 12, etc.) are for two digit codes. • Entering a single digit affects all the Dialed Number entries beginning with that digit. For example, entering 6 affects all number plan entries beginning with 6. The entries you make in step 2 and step 3 below affect the entire range of numbers beginning with 6. (For example, if you enter 3 in step 2 the entries affected are 600 ~ 699. If you enter 4 in step 2 below, the entries affected are 6000 ~ 6999.) • Entering two digits lets you define codes based on the first two digits a user dials. For example, entering 60 allows you to define the function of all codes beginning with 60. In the default program, only * and # use 2-digit codes. All the other codes are single digit. If you enter a two digit code between 0 and 9, be sure to make separate entries for all the other two digit codes within the range as well. This is because in the default program all the two digit codes between 0 and 9 are undefined.Defining codes based on more than 2 digits require a secondary program (Program 11-20) to define the codes. Step 2 : Specify the length of the code you want to change After you specify a single or two digit code, you must tell the system how many digits comprise the code. This is the Number of Digits Required column in the Table 2-1 System Numbering Default Settings on page 2-57. Step 3: Assign a function to the code selected After entering a code and specifying its length, you must assign its function. This is the Dial Type column
Network Keep Alive Setup to set the interval and retry count of the AspireNet networking keep alive message. The keep alive is used for ISDN and IP networking. The keep alive message is automatically responded to by the destination system, if the response is not received the retry count will start. If a response is not received within the number of retries, the networking link will be taken out of service. When the link is taken out of service: • Any calls that are in progress will be released. • Park Hold orbits will be released. • No further Park Hold information will be sent until the link is active. The link will automatically become active when the next keep alive response is received.
WHAT THE SOFTKEY DISPLAY PROMPTS MEAN When using a display telephone in programming mode, various Softkey options are displayed. These keys will allow you to easily select, scan, or move through the programs. Softkey Display Prompts Softkey Display Prompts If you press this Softkey ... back select The system will ... Go back one step in the program display. You can press Cursor Key (UP) or Cursor Key (Down) to scroll forward or backward through a list of programs. Scroll down through the available programs. Scroll up through the available programs. Select the currently displayed program.- 1 + 1 Move the cursor to the left. Move the cursor to the right. Move back through the available program options. Move forward through the available program options.
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.
Group listen enable/disable With this feature disabled: if a station user presses SPEAKER while on a call, the Feature Phone immediately turns off the handset and switches to hands-free mode. If enabled, the group listen feature is available system-wide. If disabled, it is no longer available. Default: Disabled. Field 2: Privacy release enable/disable With this feature enabled: if a station user presses a CO line key that is in use (lit red), the user will be immediately conferenced with the call in progress on that line. With this feature disabled: pressing an in-use CO line key has no effect. Default: Disabled.
Grounding instructions System grounding (supplemental ground) is as follows: • The conductor wires can be no smaller than the ungrounded branch-circuit supply conductors (usually 16-gauge or higher). • Acceptable wire: bare or covered with green (or green-and-yellow-striped) jacket. • Conductors (and power receptacles) shall connect to earth ground at the service equipment (usually a cold water pipe or copper ground rod). • The supplemental ground must: be used regardless of power cord ground, be connected to the ground lug on the bottom of the cabinet, and retain ground connection when the power supply module is unplugged. • Connect the grounding lugs of all units to system ground
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.
Registration The CO line telephone numbers, FCC registration number, and ringer equivalence number (REN) of this equipment must be provided to the telephone company before installation. (See below for FCC registration number and ringer equivalence number.) FCC Part 15 This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses and can radiate radio frequency energy and — if not installed and used in accordance with the instruction manual — may cause harmful interference to radio communications (in which case, the user will be required to correct the interference at his/her own expense). FCC Part 68 This equipment complies with Part 68 of the FCC Rules. On the bottom of this equipment is a label that contains, among other information, the FCC Registration Number and Ringer Equivalence Number (REN) for this equipment. You must, upon request, provide this information to your telephone company. The REN is helpful to determine the quantity of devices you say connect to your telephone line and still have all of those devices ring when your telephone number is called. In most, but not all, areas, the sum of the RENs of all devices connected to one line should not exceed five (5.0). To be certain of the number of devices you may connect to your line, as determined by the REN, you should contact your local telephone company to determine the maximum REN for your calling area. If your telephone equipment causes harm to the telephone network, the telephone company may discontinue your service temporarily. If possible, the telephone company will notify you in advance but, if advance notice is not practical, you will be notified as soon as possible. You will be informed of your right to file a complaint with the FCC. Your telephone company may make changes to its facilities, equipment, operations or procedures that could affect the proper functioning of your equipment. If so, you will be notified in advance, to give you an opportunity to maintain uninterrupted telephone service. If you experience trouble with this telephone equipment, the telephone company may ask that you disconnect this equipment from the network until the problem has been corrected or until you are sure that the equipment is not malfunctioning. This equipment may not be used on coin service provided by the telephone company. Connection to party lines is subject to state tariffs. Installation: The device is equipped with a USOC connector. Registration Number: 1T1MF08B33727. Ringer equivalence number (REN): 0.8 Hearing-aid compatibility This equipment, utilizing telephone station equipment manufactured by ESI, meets all FCC requirements for hearing-aid compatibility.