Trunk or Station Module down 1. Run a Maintenance session to ensure that the Trunk Module is not disabled. See Module status on page 537. 2. Disable the module using the Maintenance heading Module status. 3. Enable the module using the Maintenance heading Module status. 4. For Trunk Module Check the external line by terminating a single-line telephone directly on the distribution block, or equivalent, which connects to the Trunk Module. 5. For Station Module If the Station Module is still down, power down, then power up the ICS. If the problem persists 1. If AC power is present and the LED indicator on the Trunk Module is off, replace the Trunk Module. 2. Replace the fiber cable. 3. Replace the Trunk Cartridge. 4. Replace the Expansion Cartridge. 5. Replace the ICS.
Receive Alarm
Receive Alarm — yellow LED on indicates a problem with the digital transmission being received. This halfduplex link is unusable. • Receive Error — yellow LED indicates a minor error as a result of degraded digital transmission. Possible causes are an ohmic connection, water ingress, or too long a loop. • Transmit Alarm — red LED on indicates an inability to transmit. Alarm indication signal (AIS) is being transmitted to the terminating switch. This half-duplex link in unusable. • Transmit Error — yellow LED on indicates a remote alarm indication (RAI) carrier failure alarm (CFA) is being sent to the terminating switch. If the Transmit Alarm is not on, this indicates a far-end or cable problem. • In service — a flashing green LED indicates that the T1 or PRI trunks are out of service because of a running loopback test, or because the DTI is being initialized. • Loopback test — red LED on while a continuity loopback test is running. • All LEDs flashing continuously — the DTI is being initialized. 3. Run a Maintenance session and any loopback tests as appropriate. 4. Check the pinout of the cable that connects the DTI to the termination point from the T1 or PRI service provider or the external channel service unit, and check that the cable is properly connected. 5. Check with your T1 or PRI service provider to see if through-fed repeaters are used on the T1 or PRI span. The DTI does not provide the DC conne
External paging
External paging 1. Use the Button Inquiry feature (≤•‚) to verify the feature of a programmable memory button. 2. Check the wiring between the 50-pin connector and the paging amplifier or between the connections shown in the external paging wiring chart. External paging wiring Feature Page out (Tip) Page out (Ring) Pin 40 (Black-Slate) 15 (Slate-Black) Page (Make) Page (Common) 41 (Yellow-Blue) 16 (Blue-Yellow) 3. Test external paging ≤fl¤ to ensure that it is working. The nominal output signal from the Norstar ICS is 100 mV across 600 Ω. Music on Hold/Background Music trouble Although Music on Hold and Background Music are separate features, they share the same wiring and customer-supplied music source. 1. Ensure that the proper feature access code (≤°fl) is turned on. Adjust the volume using the volume control bar. 2. Use the Button Inquiry feature (≤•‚) to verify the feature on a programmable memory button. 3. If there is trouble with Music on Hold or Background Music, check Featr settings in System prgming.
Selected line pool displays: No free lines
Selected line pool displays: No free lines If the user selects a line pool and the telephone displays No free lines, refer to this section for solutions. Possible problem If this happens often, there are not enough lines in the line pool to serve the number of line pool users. Solution If the line pool contains loop start trunks, enter programming and move under-used loop start trunks from other line pools into the deficient line pool. OR If the line pool contains E&M trunks, order more trunks from the telephone company or private network vendor. Install
Dial tone absent
Dial tone absent on external lines 1. Use Button Inquiry (≤•‚) to display the number of the external line you are testing. 2. Check for dial tone using a test telephone at the connections for the external line on the distribution block. 3. Make sure that a Trunk Cartridge for the line is properly installed in the ICS or Trunk Module. 4. Make sure that the Trunk Module fiber cable is properly connected to the Expansion Cartridge on the ICS. Refer to Problems with trunk cartridges service on page 611 and Trunk or Station Module down on page 619. 5. Run a Maintenance session to ensure that the line is not disabled. See Port/DN status
password entry
Programming Password Setup to set the system passwords. For password entry, the system allows eight users to be defined. Each user can have a: • Unique alphanumeric name (up to 10 alphanumeric characters) • Password entry of up to eight digits (using0~9,#and*) • Password level The IN level password is used by the System Installer for system programming. The SA or SB level password cannot access the IN level programs. The reverse type (white on black) just beneath the Description heading is the program access level. You can only use the program if your access level meets or exceeds the level the program requires. (SA level password can access to SA or SB programs, and SB level password can access to SB programs only.)
Outgoing trunk preference
OutgoingTrunkLinePref-erenceUsethisoptiontosettheextensionoutgoingTrunkLinePreference.Ifenabled,theextensionuserreceivestrunkdialtonewhentheyliftthehandset.TheuserhearstrunkdialtoneonlyifallowedbyTrunkAccessMappro-gramming(Programs14-07and15-06).RefertotheLinePreferencefeatureformoredetails.
Multi-Party Voice Conference
Multi-Party Voice Conference Description The system allows multiple internal and external parties to be connected on a call, conference. An unlimited number of 3-party conferences may be established using iPECS Phones. In addition, each MCIM (Multi-party Conference Interface Module) up to 32 parties with the g.711 or g.729 codec and 24 with the g.723 codec may be connected in a single voice conference. The MCIM will support any combination of parties and conferences to the maximum total number of parties in conference. Multiple MCIMs, see chart below, are installed to support multiple multi-party conferences with a maximum of 32 parties in any single conference. iPECS iPECS iPECS iPECS iPECS Operation ----Micro 50 300 600 1200 iPECS Phone 1 MCIM unit maximum 2 MCIM units maximum 4 MCIM units maximum 8 MCIM units maximum 30 MCIM units maximum To establish an ad-hoc conference 1. Establish first call. 2. Press the [CONF] button. The LED will light, the connected party is placed on exclusive hold and the user receives dial tone. 3. Place second call. 4. When connected, press [CONF], new call is placed on exclusive hold. 5. Repeat steps 3 and 4 above to add additional conference parties. 6. Press [CONF] button to establish conference. To place a conference on hold 1. Press the [HOLD] button, the [CONF] button LED will flash. To retrieve held conference 1. Press [CONF] button, all parties reconnected.
Conference Room
Conference Room Description In addition to ad-hoc conferencing, users may establish a Conference Room. Other internal and external parties are invited to the conference and can join the conference without further action by the user who established the Conference Room. A user can transfer an active call to a Conference Room. A Conference Room can be password protected so that only parties that enter the password are allowed to join the Room. Up to 9 Conference Rooms can be set-up and each can support a maximum of 32 parties with the g.711 or g.729 codec or 24 parties with the g.723 codec. Conference Rooms employ channels from an MCIM (Multi-party Conference Interface Module). Each MCIM supports up to 32 parties and multiple MCIM units may be installed as shown in the chart below. iPECS-Micro iPECS-50 & 100 iPECS-300 iPECS-600 iPECS-1200 Operation Attendant Phone To view Room participant list 1 MCIM unit maximum 2 MCIM units maximum 4 MCIM units maximum 8 MCIM units maximum 30 MCIM units maximum 1. Press the [PGM] button. 2. Dial “054”. 3. Dial Room number (1~9). To delete a Conference Room 1. Press the [PGM] button. 2. Dial “055”. 3. Dial Room number (1~9). iPECS Phone To set-up a Conference Room 1. Press the [PGM] button. 2. Dial 53 to create a Conference Room. 3. Dial the desired Conference Room number (1~9). 4. If desired, enter a password for the Conference Room (must be exactly 5 digits). 5. Press [SAVE] button to establish the Room. To join a Conference Room 1. Dial 59, the Conference Room entry code. 2. Dial the Conference Room Number. 3. Dial the Conference Room password.
HOLD/SAVE
Differential Ring Signals When multiple phones in a small area ring, it can be difficult to tell which are ringing. The iPECS Phone has 14 Ring Tones available for differentiating ring from one phone to another. Four of the tones are stored in the phone’s permanent memory; the remaining ten tones are in the system’s memory. Four of these ten can be downloaded into the phone’s memory for use as the 5th to 8th Ring Tone. OPERATION To download a Ring Tone from System memory: Lift Handset Press TRANS/PGM, Dial 1 for Ring Tones, Dial 5 for Ring Tone download, Dial the memory location to receive the tone (5-8), Dial tone number 0-9, tone is heard, Press HOLD/SAVE to download. To select a Ring Tone from phone memory: Press TRANS/PGM, Dial 1 for Ring Tones, Dial 1 or 2 for Internal or EDifferential Ring Signals When multiple phones in a small area ring, it can be difficult to tell which are ringing. The iPECS Phone has 14 Ring Tones available for differentiating ring from one phone to another. Four of the tones are stored in the phone’s permanent memory; the remaining ten tones are in the system’s memory. Four of these ten can be downloaded into the phone’s memory for use as the 5th to 8th Ring Tone. OPERATION To download a Ring Tone from System memory: Lift Handset Press TRANS/PGM, Dial 1 for Ring Tones, Dial 5 for Ring Tone download, Dial the memory location to receive the tone (5-8), Dial tone number 0-9, tone is heard, Press HOLD/SAVE to download. To select a Ring Tone from phone memory: Press TRANS/PGM, Dial 1 for Ring Tones, Dial 1 or 2 for Internal or External ring, Dial tone number 1-8, Press HOLD/SAVE to make the selection. 2.4 Answering Calls at Night In the Night mode, Loud Bell Control may be used to send ring signals to external bells. You may then answer these calls with Universal Night Answer (UNA). OPERATION To answer a call during Night mode ringing over an external bell: Lift the handset, Dial the UNA code 567(iPECS-Lik)/587(iPECS-MG). to make the selection. 2.4 Answering Calls at Night In the Night mode, Loud Bell Control may be used to send ring signals to external bells. You may then answer these calls with Universal Night Answer (UNA). OPERATION To answer a call during Night mode ringing over an external bell: Lift the handset, Dial the UNA code 567(iPECS-Lik)/587(iPECS-MG).
Define the extension
Define the extension/virtual extension name.Upto12CharactersExt.101~184=Nosetting02OutgoingTrunkLinePref-erenceUsethisoptiontosettheextensionoutgoingTrunkLinePreference.Ifenabled,theextensionuserreceivestrunkdialtonewhentheyliftthehandset.TheuserhearstrunkdialtoneonlyifallowedbyTrunkAccessMappro-gramming(Programs14-07and15-06).RefertotheLinePreferencefeatureformoredetails.
Dial code to reroute call after answer
Dial code to reroute call after answer. 578 578 578 578 578 10 ACD REROUTE QCALL NO AN ENTER NEW #:579 Dial code to reroute call prior to answer. 579 579 579 579 579 11 CAMP-ON ANSWER ENTER NEW #:600 Dial code to answer a Camped On call. 600 600 600 600 600 12 CALL PARK LOCATIONS START&END#:601-610 Dial code to place/retrieve a call in a Park location. 601~610601~610 601~619 601~699601~699 13 STA GRP PILOT NUMBER START&END #:620-659 Station group pilot numbers. 620~631620~659 620~667 620~667401~500 14 STA USER VSF FEATURES ENTER NEW #:66 VSF feature access dial code. 66 66 *66 *66 *66 15 CALL COVERAGE RING ENTER NEW #:67 Code for Call Coverage button. 67 67 67 67 76 16 DIRECT CALL PICK-UP ENTER NEW #:7 Dial code to activate Directed Call Pick-up. 7 7 7 7 *77 17 ACCESS CO GROUP FEAT START&END:801-820 Dial code to access a CO Line or IP channel from a CO/IP group. 801~820801~820 801~872 801~872n/a 18 ACCESS IND CO/IP FEAT START&END:88 Dial code to access a specific CO Line. 8801 ~8805 8801 ~8842 88001 ~ 88200 88001 ~88400 88001 ~ 8860019 ACCESS HELD CO/IP FEAT ENTER NEW:8* Dial code to access last held CO Line or IP channel from Hold. 8* 8* 8* 8* 8* 20 ACCESS HELD IND CO/IP ENTER NEW #:8# Dial code to access a specific CO Line/IP channel from Hold. 8# 8# 8# 8# 8# 21 ACCESS CO IN 1ST CO GRP ENTER NEW #:9 Dial code to access the 1st available CO Line in any accessible group. 9 9 9 9 9 22 ATTENDANT CALL ENTER NEW #:0 Dial code to call an Attendant. 0 0 0 0 0 23 VM MSG WAIT ENABLE ENTER NEW #:*8 Dial code for external Voice mail to activate Message Wait indication. *8 *8 *8 *8 *8 24 VM MSG WAIT CANCEL ENTER NEW #:*9 Dial code for external Voice Mail to deactivate Message Wait
DID
DID calls are subject to Group Call Pick-up and Directed Call Pick-up. 5. If a VMIM/VSF announcement is defined as the destination in the Flexible DID Destination Table, a Caller Controlled Routing Table for the announcement can be defined. iPECS can be configured to drop (disconnect) the call after playing the recorded announcement. Programming STATION 1 SIP User ID Table (PGM 111-Btn 19) 2 Station SIP Attributes 2 (PGM 126-Web only) BOARD 1 H323 VoIP Attributes 2 SIP Gateway Attributes CO/IP 1 CO Service Type (PGM140) 2 ISDN DID Remove Digit Count (PGM 143-Btn 5) 3 DID Conversion Type (PGM 145) SYSTEM 1 DID/DISA Busy Destination (PGM 167-Btn 1) 2 DID/DISA Error Destination (PGM 167-Btn 2) 3 DID/DISA No Answer Timer (PGM 181-Btn 2) TABLES 1 CCR Audio Text Tables (PGM 228) 2 Flexible DID Table (PGM 231
B Channel
B channels B channels are the bearer channel. They are used to carry voice or data information and have speeds of 64 kbps. Since each ISDN line (BRI or PRI) has more than one B-channel, more than one transmission can occur at the same time, using a single ISDN line. D channels The standard signaling protocol is transmitted over a dedicated data channel called the D-channel. The D-channel carries call setup and feature activation information to the destination. This channel has speeds of 16 kbps (BRI) and 64 kbps (PRI). Data information consists of control and signal information and packet-switched data such as credit card verification.
ISDN layers
ISDN layers ISDN layers refer to the standards established to guide the manufacturers of ISDN equipment. The layers include both physical connections, such as wiring, and logical connections, which are programmed in computer software. When equipment is designed to the ISDN standard for one of the layers, it works with equipment for the layers above and below it. There are three layers at work in ISDN for Norstar. To support ISDN service, all three layers must be working properly.
UCS Mobile Client
IPECS applications Taking Business Communications to A New Level UCS Desktop Client UCS Desktop Client enables presence, instant messaging, file sharing, multi-party conferences, internal and external SMS, video conferencing, call recording, application sharing, white board and desktop sharing, MS Outlook integration and much more. UCS is designed as a single-server solution with straight-forward installation and management with simple licensing that results in low total-cost-of-ownership (TCO). UCS Mobile Client No need to be in the office to take that big meeting. UCS Mobile Client brings the power of iPECS UCS to your smartphone. Available for Android, iOS and Windows. Unleash the full power of iPECS, and experience the next generation in business communications technology. With a full range of powerful software applications, the true potential of your iPECS voice platforms can be realized. As your business grows and your technology needs become more sophisticated, Ericsson applications leverage your current iPECS investment. Applications deliver features that empower your employees to be more productive, more mobile and more collaborative. They can also enhance your business’ ability to deliver a more responsive and superior customer service experience beyond traditional voice-only customer contact. Whether your business is faced with the continued adoption and resulting challenges of “bring-your-own-device” (BYOD), increased need for Unified Communications (UC) or the complexity of managing a geographically dispersed and disparate voice and data network, iPECS applications are optimized to help you meet your needs while your business continues to transform. Unifying Your Business iPECS UCS is a complete Unified Communications solution designed for the SMB that keeps people communicating when they are in the office or away.
ISDN
This chapter provides you with some background information about ISDN, including information about: • analog vs. ISDN • type of ISDN service •ISDN l ayer s • ISDN bearer capability • services and features for ISDN PRI and BRI • ISDN hardware • ISDN standards compatibility Integrated Services Digital Network (ISDN) technology provides a fast, accurate, and reliable means of sending and receiving voice, data, images, text, and other information through the telecom network. ISDN uses existing analog telephone wires. The signal on the wire gets divided into separate digital channels, which dramatically increases the bandwidth. ISDN uses a single transport to carry multiple information types. What once required separate networks for voice, data, images, or video conferencing is now combined on to one common high-speed transport.
iPECS sBG-1000
iPECS sBG-1000 Integrated office in a box The iPECS SBG-1000 is the ideal converged business communication services platform for small businesses, multi-site and branch office environments. With voice, data and IT services built-in, the SBG-1000 is a unique platform that offers the best price-to-feature ratio, all in a single appliance. Enhanced security, VPN and mobility features allow businesses to confidently extend and manage access and services to remote and mobile workers in a secure environment.
intercom call hold
INTERCOM CALL HOLD Description While on an active ICM Call, users of iPECS Phones can place the ICM Call on hold. The held station will receive the assigned Music-on-Hold. The call is placed on Exclusive Hold and recalls to the holding station after expiration of the Exclusive Hold Recall Timer. Operation iPECS Phone To place an active ICM call on hold 1. Press the [HOLD] button, the [ICM] button LED will flash at the exclusive hold rate. ICM dial tone is received. To retrieve the held ICM call 1. Press the [ICM] button or the {DSS/BLF} button associated with the held station, the [ICM] button LED illuminate and the ICM call connected. Conditions 1. Only one ICM call may be placed on hold at a station. Programming Related Features MOH (Music-On-Hold) Intercom Call (ICM Call) Exclusive Hold Hold Recall Hardware iPECS Phone
Intercom call hold
INTERCOM CALL HOLD Description While on an active ICM Call, users of iPECS Phones can place the ICM Call on hold. The held station will receive the assigned Music-on-Hold. The call is placed on Exclusive Hold and recalls to the holding station after expiration of the Exclusive Hold Recall Timer. Operation iPECS Phone To place an active ICM call on hold 1. Press the [HOLD] button, the [ICM] button LED will flash at the exclusive hold rate. ICM dial tone is received. To retrieve the held ICM call 1. Press the [ICM] button or the {DSS/BLF} button associated with the held station, the [ICM] button LED illuminate and the ICM call connected. Conditions 1. Only one ICM call may be placed on hold at a station. Programming Related Features MOH (Music-On-Hold) Intercom Call (ICM Call) Exclusive Hold Hold Recall Hardware iPECS Phone
User Agent A SIP User Agent
Network Elements 1.1 User Agent A SIP User Agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction. 1.2 Proxy server An intermediary entity that acts as both a server (UAS) and a client (UAC) for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. 1.3 Registrar A server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles which registers one or more IP addresses to a certain SIP URI. SIP registrars are logical elements, and are commonly co-located with SIP proxies. But it is also possible and often good for network scalability to place this location service with a redirect server. 1.4 Redirect server A user agent server that generates 3xx (Redirection) responses to requests it receives, directing the client to contact an alternate set of URIs. The redirect server allows proxy servers to direct SIP session invitations to external domains. 1.5 Session border controller Session border controllers serve as middle boxes between UA and SIP server for various types of functions, including network topology hiding, and assistance in NAT traversal. 1.6 Gateway Gateways can be used to interface a SIP network to other networks, such as the public switched telephone network, which use different protocols or technologies. 1.7 Application Layer Gateway ALG is a SIP aware monitoring device commonly contained in Routers and or Firewalls. SIP ALGs can have the capabilities of changing SIP Messages and should be disabled if any issues are experienced with SIP Calls.
C-3000
This entry system is designed to utilize a building’s existing telephone wiring and to address a variety of apartment entry applications. Viking’s C-3000 Entry System provides secure multi-tenant building access, without the need for any C.O. lines; the only requirement is that each tenant have a touchtone telephone set wired to their apartment. The C-3000 performs equally as well in installations where C.O. lines are present. An individual C-3000 module has a 12-tenant capacity; for larger applications, a maximum of eight (8) C-3000 modules can be interconnected, allowing a total system capacity of up to 96 tenants. A C-3000 system consists of at least one Master C-3000 module, and as many as seven Slave modules. From either of two entry doors, visitors may ring a tenant's telephone with a distinctive cadence, alerting the tenant of their arrival. The C-3000 provides call waiting tones if the tenant’s line is in use. The tenant may then converse freely with the visitor via their existing telephones. Once identified, the tenant can let the visitor in by entering a single touch-tone on the keypad of the tenant's telephone set. The C-3000 keyless entry feature supports unique entry codes for each tenant. As a measure of security, the codes can be easily changed as tenants move in and out of the building. A postal lock feature is also supplied as an added convenience. A
LIK-MFIM50B Multi-Function Interface gateway
LIK-MFIM50B Multi-Function Interface gateway Module, 50 ports w/ISDN-BRI 1 MICRO Multi-Function Interface gateway Module, 31 ports 2 AC/DC Adapter AC/DC Adapter for LIK-MFIM50A/B module (12VDC, 1.5A) 3 AC/DC Adapter –K- AC/DC Adapter for LIP Phones and DSS Console (48VDC, 0.3A) 4 DHLD Desk mount Holder for module 5 DHE Desk mount Holder Extender, one (1) required for each Module 6 WHLD Wall mount Holder for module 7 1U-RMB 1U Rack mount Bracket 8 LIP-7004N LIP Phone, Basic 4 button no display 9 LIP-7008D LIP Phone, 8 button and basic 2-line display 10 LIP-7016D LIP Phone, 16 button, 3-line display w/Menu, Soft & Nav. buttons 11 LIP-7024D LIP Phone, 24 button, 3-line display w/Menu, Soft & Nav. buttons 12 LIP-7024LD LIP Phone, 24 button, Large display w/Menu, Soft & Nav. buttons 13 LIP-7048DSS LIP DSS Console with 48 buttons 14 LIP-8002 LIP Phone, 4 button and 1-line display, LAN 1 port 15 LIP-8004D LIP Phone, 4 button and 1-line display, LAN 1 port 16 LIP-8012D LIP Phone, 12 button, 3-line display w/Menu, Soft & Nav. buttons 17 LIP-8024D LIP Phone, 24 button, 4-line display w/Menu, Soft & Nav. buttons 18 LIP-8040L LIP Phone, 10 button, 9-line display w/Menu, Soft & Nav. buttons 19 LIP-8048DSS LIP DSS Console with 48 buttons 20 LIP-8012DSS LIP DSS Console with 12 buttons 21 LIP-8012LSS LIP DSS Console with 12 buttons, w/12-line LCD button label 22 LIP-7004WMK Wall Mount Kit for LIP-7004N 23 LIP-7008WMK Wall Mount Kit for 7008D 24 LIP-7024WMK Wall Mount Kit for LIP-7016D, 7024D & 7024LD 25 WIT-300HE/400H iPECS WLAN Phone 25 WIT-400H iPECS WLAN Phone 26 GDC-400B DECT Base Station 27 GDC-600B DECT Base Station 28 GDC-400H DECT Handset 29 GDC-450H DECT Handset
Power supply
Power Supply The Total Access 900e Series products have a 90 to 125 VAC power supply with an IEC connector. The appropriate three-prong cable is included in the shipment. Battery Backup Connection An optional battery backup system is available for the Total Access 900e Series (P/N 1175044L1 or L2). The connection port is labeled BATT. Refer to the documentation available for your specific battery backup unit for more information on this connection, or refer to Battery Backup Unit on page 25 for a more details. CRAFT Interface The CRAFT interface is an EIA-232 serial port (DCE) that provides for local management and configuration (via a DB-9 female connector). Table A-5 on page 31 shows the CRAFT port pinouts.
Posts
