H323 Setup Mode

1 H323 Setup Mode 0~1 0 0: Fast/1: Normal 2 H323 Tunneling Mode 1: ON, 0: OFF OFF 0:Off/1:On 3 H323 DTMF Path 0~1 MFIM: Out VOIM: Inband 1:Out/0:In 4 H323 DiffServ Pre tagging 00~63 4 5 RAS Usage 1: ON, 0: OFF OFF 6 RAS Multi-cast IP Address IP address 7 RAS Multi-cast IP port Port number 1718 8 RAS Uni-cast IP Address IP address 9 RAS Uni-cast IP port Port number 1719 10 RAS Keep-alive Timer 001 ~ 999 120 1 second increments 11 RAS Numbering Plan prefix 24 digits 12 RAS Gateway Id 128 characters Web Admin only 13 RAS Light RRQ 1: ON, 0: OFF OFF 14 TCP Keep Alive 1: ON, 0: OFF ON 15 FAIL OVER USAGE 1: ON, 0: OFF ON 16 17 18 Q931 START PORT 00001-65535 2048 TCP Port 19 Q931 END PORT 00001-65535 2559 TCP Port 20 H245 START PORT 00001-65535 2560 TCP Port 21 H245 END PORT 00001-65535 3071 TCP Port 22 RAS START PORT 00001-65535 2048 UDP Port 23 RAS END PORT 00001-65535 3071 UDP Port 24-1 MEDIA START PORT 00001-65535 6000 UDP Port 24-2 MEDIA END PORT 00001-65535 8800 UDP Port 24-3 DATA START PORT 00001-65535 8500 UDP Port 24-4 DATA END PORT 00001-65535 8548 UDP Port
PGM Code: 131 -T1/PRI Attributes
1 T1 Setup Mode 0~1 D4 0:D4/1:ESF 2 T1 Line Mode 0~1 B8ZS 0: B8ZS/1:AMI 3 PRI Line Mode 0~1 TE 0:NT/1:NT 4 PRI CRC Check 1: ON, 0: OFF ON 5 E1 R2 DSP Check 1: ON, 0: OFF ON 6 DCO PX Type


Your LAN Administrator will provide these parameters. These parameters can only be installed into the PARTNER or MERLIN Messaging system using the module’s RS-232 serial interface and a PC-based communication application, such as HyperTerminal. The HyperTerminal application is available on any PC running a Windows (95, 98, NT, 2000, or XP) operating system. The PC must have its serial port connected to the RS-232 labeled port on the front panel of the messaging system module. This will require a D8W cable (up to 25-feet), a 355AF adapter, and a male DB-25-to-female-DB-9 connector. For more information regarding this serial connection, see the section “Establishing a session” in the “Terminal Based Reporting, Diagnostics and Maintenance” section of the MERLIN Messaging Release 4.0 or PARTNER Messaging Release 7.0 Installation, Programming, and Trouble Shooting Guide Online contained on the Library CD.
The following steps describe how to install the assigned Static IP, Subnet Mask and Default Gateway addresses on the messaging system:
1. Connect the RS-232 cable from a PC to the RS-232 serial port on the messaging module.
2. Start a HyperTerminal session on the PC. The Connection Description dialog box appears.
3. Enter a name that describes the connection (for example, MERLIN Messaging or PARTNER Messaging), and click the OK button. The Connect To dialog box appears.
4. From the Connect using box, select the COM port to which the RS-232 cable is connected, and then click the OK button. The COM Properties dialog box appears.
5. Set the following options:
ƒ Bits per second: 38400 (MERLIN Messaging) or 19200 (PARTNER Messaging) ƒ Data bits: 8 ƒ Parity: None ƒ Stop bits: 1 ƒ Flow control: None
6. Click the OK button. The messaging system prompts you to login.
7. Type sysadmin, and then press the ENTER key. The Password prompt appears.
8. Enter the administrator password, and then press the ENTER key. The Options menu appears.

fractional PRI

Working with fractional PRI Installer password required
When you purchase PRI from your service provider, you can request the number of B-channels that are allocated for you to use. For example, you may want to use only 12 B-channels instead of 23 B-channels. If this is your situation, you should disable all the B-channels that you do not need.
It is recommended that the number of lines that are deprovisioned on a DTI card (configured as PRI) be the same as the number of B-channels that are disabled. For example when B-channels 13-23 are disabled, you should deprovision

Automatic Route Selection

Automatic Route Selection (ARS/F-Route) provides call routing and call restriction based on the digits a user dials. ARS gives the system the most cost-effective use of the connected long distance carriers.
ARS is an on-line call routing program that you can customize (like other system options) from a display telephone. ARS accommodates 400 call routing choices - without a custom-ordered rate structure database. With ARS, you can modify the system routing choices quickly and easily. This is often necessary in the telecommunications world of today where the cost structure and service choices frequently change.
The ARS feature can add or delete digits and route calls according to pre-determined levels. ARS Feature Summary ARS provides: Call Routing
ARS can apply up to 36-digit analysis to every number dialed. For programming, ARS provides separate 8-digit and 36-digit tables. Each table can have up to 250 numbers.
• Dialing Translation (Special Dialing Instructions)
ARS can automatically execute stored dialing instructions (called Dial Treatments) when it chooses a route for a call. The system allows up to 15 Dial Treatments. The Dial Treatments can:
- Insert or delete an area code (NPA)
- Add digits (such as a dial-up OCC number), pauses and waits to the dialing sequence
- Require the user to enter an authorization code when placing a call (refer to PRG 44-03)
• Time of Day Selection
For routing purposes, ARS provides 10 different day selections (called Time Schedule Patterns). Each Time Schedule Pattern can provide up to 20 time intervals which are assigned to one of the eight day/night modes. The Time Schedule Patterns are then assigned to a day of the week (Monday~Friday, Saturday, Sunday or Holiday).
• Hierarchical Class of Service Control
ARS allows or denies call route choices based on an extension ARS Class of Service. This allows lower Classes of Service (e.g., 1) to access routes unavailable to higher Classes of Service (e.g.,
16). The system provides up to 16 (0=unrestricted, 1~16) ARS Classes of Service.
• Separate Routing for Selected Call Types
To provide unique control, you can program separate routing instructions for:
- Directory assistance calls
- Emergency calls
Basic ARS Operation
When a user places an outside call, ARS analyzes the digits dialed and assigns one of 400 Selection Numbers to the call. The Selection Number chosen depends on which digits the user dialed. ARS then checks the time of day, the day of week and the extension ARS Class of Service. Based on these call routing options, ARS selects a trunk group for the call and imposes the Dial Treatment instructions (if any).

Trunk or Station Module down

Trunk or Station Module down

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


The battery on the CPU retains the Clock/Calendar when the CPU encounters a power loss. With a fully charged battery, the settings are retained for approximately three years.
The system programmed memory (Customer Database) is stored in Nonvolatile Memory and can be erased only by performing a First Initialization.
For additional storage time and the database can be copied to the Compact Flash card on the CPU.
• The battery on the CPU should be removed during long term storage but must be installed (protection against loss of power) just before ETU installation to provide battery backup for System Memory.
• When fully charged, the battery retains System Memory for approximately three years.
• You should replace the CPU battery every three years.
• During normal operation, the battery is continually recharged using a built-in charging circuit from the CPU.
• Battery backup on the CPU does not protect the following:
- Callback
- Off-line Status (for programming system or station assignments)
- Repeat Redial
- Trunk Queuing/Camp-On

CSU stats

CSU stats

Each DTI is equipped with an internal channel service unit (CSU). When enabled, the internal CSU monitors the quality of the received T1 signal and provides performance statistics and diagnostic information.
DTIs must be individually programmed to establish parameters for collecting and measuring transmission performance statistics by the CSU.
Statistics Installer password required
The CSU provides both performance and alarm statistics. Three performance parameters are accumulated:
• errored seconds (ES-P)
• severely errored seconds (SES-P)
• unavailable seconds (UAS-P)
These parameters are defined as per TIA-547A. Errored seconds are enhanced to include control slip (CS) events.
The parameters are stored for the previous 15-minute interval, the 15-minute intervals in the last 24 hours, and the previous 24-hour interval. Only near-end performance data is recorded.
The internal CSU continuously monitors the received signal and detects four types of transmission defects:
• any active carrier failure alarms (CFA) — loss of signal LOS, out of frame OOF, alarm indication signal AIS, remote alarm indication RAI
• the number of bipolar violations that occurred in the last minute

digital phone

7000 digital phone This telephone has a no display. It does have four memory buttons. (Note this telephone is only supported on systems running Profile 2, 3, or 4).
7100 digital phone: A telephone with a single line display and one programmable memory button without an indicator.
7208 digital phone: A telephone with a single-line display and eight programmable memory buttons with indicators. This telephone has a separate mute key and supports a headset
7316 digital phone: A telephone that has a two-line display, three display buttons, 16 programmable memory buttons with indicators, and 12 memory programmable buttons without indicators. This telephone has a separate mute key and supports a headset.
7316E digital phone: This telephone has the same functionality as the 7316 digital phone, with some additional features, such as a separate handsfree key, special display icons (when running on a MICS 6.1 or newer system), and CAP capability by adding KIMs.
7406 digital phone: The 7406 base station desk set can support three handsets, which function like the other digital phones on the system. It has six programmable memory keys.
7420/7430: These Nortel Networks Digital Mobility phones allow you to set up an extended cell of base stations that allow users to contact the system from a variety of locations within a site.
AbsorbLength: A setting that determines how many of the digits in a destination code will not be dialed by the system. AbsorbLength is assigned under Destination codes in Services.
access code: Different sequences of characters used to gain access to these Norstar features: Line pools, Call park, external lines, Direct-Dial telephone, Auto DN, and DISA DN.
alarm code: A number that appears on the alarm telephone display, informing you that the ICS has detected a fault in the system.
alarm telephone: A telephone that is designated to receive reports of Norstar system problems. This function is usually assigned to a prime telephone, but this can be changed under Feature settings in Sys prgrammng.
Analog Terminal Adapter (ATA): A device that permits analog telecommunication devices such as fax machines, answering machines, and single line telephones to be connected to the Norstar system. Programmed defaults for the ATA are automatically assigned by the Norstar system.
ANSI: American National Standards Institute.
Answer button: A telephone button with an indicator that is used to monitor another telephone. The answer button indicates incoming calls destined for the other telephone. Someone working at a telephone with answer buttons (an attendant, for example) can receive all

Programming an Operator’s Extension

Programming an Operator’s Extension 7
One of the initial programming duties in programming extensions is to set up operator extensions. To set up an operator’s extension, you must program call handling options, backup answering options, and buttons used for specific operator features.
Call Handling Options 7
If you set up a centralized telephone answering position at extension 10, use the following settings to customize it:
■ Call Answering. If the operator should answer all calls, use Line Assignment (#301) to assign all lines to extension 10. Set Line Ringing for all lines at extension 10 to the desired number of Rings; set the lines assigned at each user’s extension to Delayed Ring or No Ring.
In Hybrid mode, Immediate Call Answering is the factory setting. (Lines are assigned as individual line buttons on the telephone at extension 10 and all pool buttons assigned to users’ extensions are set to No Ring.)
■ Backup Call Answering. If the operator should answer some lines only when a user does not pick up, set Line Ringing for those lines at extension 10 to Delayed Ring; set the lines or pools assigned at each user’s extension to Immediate Ring.
■ No Answering. If some lines should not be picked up by the operator at all, either set Line Ringing for those lines at extension 10 to No Ring, or simply use Line Assignment (#301) to remove those lines from extension 10. In either case, set Line Access Restriction (#302) to No Access for those lines at extension 10 to prevent the operator from using Direct Line Pickup to access those lines.

Features from 6.1MR addendum

Features from 6.1MR addendum
The following features were introduced with the MICS 6.1MR (Maintenance Release) software in the Modular and Compact ICS 6.1 Maintenance Release (MR) Documentation Update addendum (P0609198 02):
• The enhanced Call log feature allows you to log all calls to a telephone, or to gather logs for specific lines assigned to a telephone. The Call log set feature allows you to determine which assigned lines will collect logs. Refer to Call log set on page 358 and Call log on page 436. (all profiles)
• The second dial tone table allows the user to enter up to 10 one to four-digit numbers that, when dialed, will cause the system to produce a second dial tone, at which time the user can enter the remaining call digits. Refer to Configuring the second dial tone table on page 403. (all profiles)
• Profiles 1 and 4, PRI: The Send Name Display feature allows you to specify if you want the business name and OLI to be transmitted over specific PRI lines.

Programming ISDN BRI resources

Programming ISDN BRI resources
Some steps will not be necessary depending on the service you are providing.
More detailed information is included under the individual headings and settings in the Programming and Maintenance sections.
For complete card and cartridge installation instructions and safety precautions, see Installation on page 227.
1. Collect the information supplied by your service provider to support your ISDN package. This includes network service profile identifiers (SPIDs) and Network DNs. If you are supporting a point-of-sale terminal adapter, you also need one or more terminal endpoint identifiers (TEIs).
2. Make sure a Combination Fiber 6-port Services Cartridge, or a Services Cartridge has been installed in the ICS.
3. Install the BRI card in the ICS, Trunk Module. Refer to Installing the cartridges on page 231 for information about BRI card placement. If you are not using a BRI card, determine which type of card you will preprogram the ICS to use in each slot.

Data Solutions

Data Solutions
Examples of ISDN Scenarios
For information about various ISDN scenarios that may help you to decide on the data solution that is best for you, refer to the following web site: http://www.nortelnetworks.com/ support and perform a Search for TIPS. You will need your user name and access code.
If you do not have a user name and access code, the site provides information about how to get access to this site.
ISDN applications
ISDN terminal equipment delivers a wide range of powerful business applications:
• Video conferencing and video telephony: Video conferencing offers instant visual and audio contact between distant parties using either studio-based or desktop ISDN terminals.
• Desktop conferencing: ISDN allows computer users in distant locations to share and edit any image, data or text file on their own computer screens while they discuss the information.
• File transfer: The ISDN network allows you to transfer files containing data, text, images, data, or audio clips, faster and cheaper than with a conventional modem.
• Telecommuting: Convenient retrieval, processing and storage of files is possible for the employee working at home by using ISDN lines to give high-speed access to information resources at the office.

Working with ISDN

Working with ISDN
Planning your ISDN network
Consult ISDN hardware on page 64 and ISDN programming on page 76 to determine a configuration of ISDN trunks and terminal equipment (TE) for the Modular ICS, then order the appropriate ISDN capability package from your ISDN service provider.
For ISDN BRI service, your service provider supplies service profile identifiers (SPIDs), network directory numbers (Network DNs), terminal endpoint identifiers (TEIs), and other information, as required, to program your Modular ICS, TE, and other ISDN equipment.
Modular ICS does not support any package with EKTS (Electronic Key Telephone System) or CACH (Call Appearance Call Handling). EKTS is a package of features provided by the service provider and may include features such as Call Forwarding, Link, Three-Way Calling, and Calling Party Identification.
Ordering ISDN PRI
When you order ISDN PRI, order two-way DID because it simplifies provisioning and provides efficient use of the PRI bandwidth.
Ordering ISDN PRI service in Canada
In Canada, order Megalink™ service, the trade name for standard PRI service and set the Norstar equipment to the supported protocol that is identified by your service provider, either DMS-100 or NI-2.


Using the Attendant Station Program Codes, the Attendant can print SMDR and Traffic reports on-demand, assign Authorization Codes, control certain user features, record VMIM/VSF announcements, enable/disable Auto Service Mode Control, etc. Items are available using the Program Code directly or scrolling the multi-level display menu. The following indicates the menu displays, including the digit for selecting the item, the item description and further required entries. The various levels of the display menu are indicated by indentation. For additional information, reference Appendix C of the iPECS Admin & Programming Manual.
Note also, some Program Codes are only available to the System Attendant or stations allowed access to the Attendant Program code “0”.

IP Trunk - (SIP) Session

IP Trunk - (SIP) Session Initiation Protocol
Version 2.0 or higher software provides;
• When + is added to the country code of an incoming SIP trunk call, it is recognized as an
international call, simplifying outgoing calls from the incoming call list.
The SL1100 IP Trunk SIP package sends the real time voice over the corporate LAN or WAN. The voice from the telephone is digitized and then put into frames to be sent over a network using Internet protocol.
Using VoIPDB equipment at a gateway (a network point that acts as an entrance to another network), the packetized voice transmissions from users in the company are received and routed to other parts of the company Intranet (local area or wide area network) or they can be sent over the Internet using CO lines to another gateway.
Depending on the requirements and resource allocation in the LAN/WAN/Internet, the VoIPDB - SIP can be configured to use any of the following voice compressions:
• G.711 μ-law - Highest Bandwidth
• G.729 (a) - Most often used
• The LAN/WAN or Internet connection is provided by a 10 Base-T/100 Base-TX Ethernet.
For a list of vendors that have successfully completed interoperability certification go to http:// www.necntac.com and refer to Technical Documentation.
SIP Trunk E.164 Support (V2.0 or higher)
With the SIP Trunk E.164 Support enabled the PBX is able to support SIP configurations where the number presentation within the SIP messages is formatted using the E.164 international numbering scheme. Specifically the system is able to handle the ‘+’ digit when required as the International Access Code.
For example a normal international SIP call could be dialed and presented from the system as follows; Number dialed = 00441202223344
Request-URI: Invite sip: 00441202223344@ SIP/2.0
However with SIP Trunk E.164 Support enabled the SIP call could be presented once dialed as below; Request-URI: Invite sip:+441202223344@ SIP/2.0
This presentation can be a requirement of certain SIP ITSPs (Internet Telephony Service Providers) so it is necessary the PBX can handle these calls and modify any SIP messages to the correct format accordingly.
Below is the full list of SIP header fields used by this feature: Request-URIToFromP-Asserted IdentityP-Preferred Identity
SIP Trunk E.164 CLIP Enhancement (V2.0 or higher)
With the SIP Trunk E.164 CLIP Enhancement enabled, when an incoming SIP call from an external ITSP is presented at the system with a “+” in the From header field as the international access code, it is recognized and displayed as an international call at the terminal display and also logged in the terminals incoming caller history, allowing any outbound calls made from a multiline terminals caller history possible using this numbering scheme.
This presentation can be a requirement of certain SIP ITSPs (Internet Telephony Service Providers) so it is necessary the PBX can handle these calls and modify any SIP messages to the correct format accordingly.


Users can transfer an active Intercom call to other stations in the iPECS system. Intercom calls can be transferred after announcing the call (screened) or without announcing the call (unscreened).
The Intercom station is placed on Exclusive Hold. The Transfer Recall Timer is initiated and, if this timer expires before the Intercom call is answered, the call will recall the transferring station until answered or abandoned.
iPECS Phone
To perform an Screened ICM transfer, while on an ICM call
1. Press [TRANS] button.
2. Dial the station to receive call.
3. At answer or Splash tone, announce call.
4. Hang-up, return to idle. Or,
1. Press {DSS/BLF} button for the desired station.
2. At answer or Splash tone, announce call.
3. Hang-up, return to idle.
While on a Intercom call, Unscreened call transfer
1. Press [TRANS] button.
2. Dial Station to receive call.
3. Hang-up, return to idle. Or,
1. Press {DSS/BLF} button for the desired station.
2. Hang-up, return to idle.

Confirm that the system is correctly set, perform the following tests

Confirm that the system is correctly set, perform the following tests:
 If you run into an issue with any of these tests, refer to Table 3 Troubleshooting Guide. Test an outgoing call to a local number. Check for ringback, 2-way audio and quality.
1. Test an outgoing call to a long distance number. Check for ringback, 2-way audio and quality.
2. Test an outgoing call to an international number. Check for ringback, 2-way audio and quality.
3. Test a outgoing call lasting more than 15 minutes.
4. Test multiple call concurrences on outgoing calls. Setup multiple calls to PSTN.
5. Test an outgoing call to an Operator ‘0’.
6. Test an outgoing call to directory assistance ‘411’.
7. Test a 911 call.

Identify to the operator that this is a TEST!
8. Test an incoming call to an internal DID. Check for ringback, 2-way audio and quality.
9. Test an incoming call to an auto-attendant. Check DTMF and audio quality.
10. Test transferring calls off-site.
11. Test an outgoing call to an auto-attendant and verify DTMF.

Setting the Time

Setting the Time
Users can set the time on the server using NTP to set the time automatically or users can adjust the time and date manually. NTP is the preferred setting, if the server has internet access.
To set the time:
1. Navigate to Maintenance > Time.
2. Click the modify link in the Action column. The Time page displays.
3. Click the radio button to:
a. Use NTP to set time automatically. In the NTP server field, specify an SNTP server IP address or a domain name. In the Poll Period, specify the number of minutes between polls. OR
b. Set time manually - specify the time in hours, minutes, seconds, and date in the respective fields. .
4. Go to the Time Zone section and select the appropriate time zone from the drop-down list.
a. Check the Automatically adjust clock for Daylight Saving Time box underneath the selected time zone to enable the system to update the time automatically.
b. Click the Get Time button to update the clock to the current time when using an SNTP server to specify the time. Click the Set Time button to save changes. . .

Remote system access

Remote system access
The remote access feature allows callers elsewhere on the private or the public network to access a Norstar system by dialing directly into the system without going through an attendant. Once in the system, the remote user can use some of the system resources. The remote access must be enabled in programming before callers can use it.
Norstar systems support remote system access on the following trunk types, which may require the remote caller to enter a COS password for direct inward system access (DISA):
• auto-answer loop start trunks
• auto-answer E&M trunks
• DID trunks, by means of the DISA DN
• PRI trunks, by means of the DISA DN
The system resources, such as dialing capabilities, line pool access and feature access, that a remote user may access depends on the Class of Service (COS) assigned to the user. See Class of Service on page 103, COS pswds on page 412 and the Modular ICS 7.0 System Coordinator Guide for more details.