Frequently Asked Questions

Support :: Frequently Asked Questions

We've compiled a list of Frequently Asked Questions and grouped them by category to make it easy for you to find the answer you need.

Technical Support

Q: What is the difference between a non-INL telephone and an INL telephone?
A: Non-inline phones provide all the basic functionality and benefits of the other Meridian sets. These phones require a power board and transformer when adding an add-on module. Inline power phones have an enhanced display and built in power board. You no longer need to purchase a power board or transformer if you require an add-on module or display.
Q: My Meridian 1 or Norstar phone used to work, but now has a problem. What should I do?
A: Switch the bad phone with another one in your office of the same type. (Make sure to include base cord that goes from wall jack to phone when changing.) If new phone works ok, then try changing base cord, handset and coil (cord that connects handset to phone) with bad phone to try and make good phone fail. If it does fail, replace the appropriate component.
Q: My M2XXX or M3XXX was unplugged when I came in this morning, so I plugged it in, but now it's dead. Why?
A: The PBX system runs a routine daily (usually overnight) that checks each digital phone. If it doesn't see a phone hooked up, it disables the port. You can either wait until it runs the next routine, or call your service person that can log into your system and re-enable your port.
Q: A key on my Norstar phone stopped working. Why?
A: Chances are you had a power outage or spike that caused the phone system to default, or someone made a programming change on your phone. Press feature button and then * and 0 key on your touch-tone pad. Then press button that is not functioning. The display will show what the system has programmed on that button. This probably will not match what you're trying to use it for, so you will need to have someone re-program it. Check with your manager to see who does the programming for your system.
Q: I can't see the display on my Norstar phone. What should I do?
A: Press feature button and then * and 7 on your touch-tone pad. Then press 1,2,3,4 on your touch-tone pad to change the display contrast. Then press the Rls button. (Note: some phones can use 5,6,7,8 and 9 on the touch-tone pad also to change the contrast)
Q: I can't hear the ring on my Norstar or M2XXX phone. What should I do?
A: Have someone call your phone, and while it's ringing, press the volume up bar under the touch-tone pad. This should increase the volume on the phone.
Q: I just installed my 22-button to my M2616 phone but can't get any of the keys to show up in programming. Why?
A: You need to change the prompt AOM in LD-11 from 0 to 1 for your phone. Then key 16-37 will show up for programming.
Q: I just added a 48-button KLM to my M7324 phone but it doesn't work. Why?
A: Make sure you plug the base cord from the phone into the KLM and then the base cord from the KLM into the wall where you had your M7324 phone plugged in. When you first plug it in, the lights on the KLM should flash slowly along with the lights on the M7324. If the M7324 lights flash, but the KLM lights do not, check that a 24 vdc saps power transformer was hooked to the black/yellow wires of your jack.
Q: I just upgraded my phone from an M2XXX or M3XXX to a M2XXX or M3XXX. I can get dial tone on key 0, but none of the other keys function. Why?
A: The port needs to be re-programmed in LD-11 at the TYPE prompt to match the new phone that you installed.
Q: I programmed key 6 on my M2008 HF phone as a call forward, or some other key, but can't get it to work. Why?
A: Key 6 on the M2008HF phone is automatically the handsfree/mute key and will not allow any other function.
Q: Can I get caller ID info on an analog phone in my Meridian 1 PBX?
A: Yes. You need to be running release 23 or later software and use an NT5D60 card. You also need an analog port, and a phone capable of caller ID.
Q: What does In-line power on a Meridian 1 phone do?
A: This enables the display on a M2008 phone to work with out external power, and also allows an M2616 phone to use a 22-button add-on module without external power. (Note: If you add a data adapter to any phone, it always requires a power supply and a power board.)
Q: What button can I program, as a user, on my Norstar phone?
A: All, except line, intercom, answer, or handsfree/mute. To program a feature onto a button: press feature button and * and 3 on your touch-tone pad. Press the button you want to program. Press the feature button and enter the feature code you want to program. (Note: A partial list of feature codes should be listed on the index card under the handset on your phone.) To program an external auto-dial button use feature * 1 instead of feature * 3 above, for internal auto-dial button programming use feature * 2.
Voice Mail
Q: How do I hook a Startalk, Flashtalk, or Norstar voice mail into the system?
A: The voice mail will have modular jacks in it similar to a wall jack. These need to be connected via a base cord to any free station ports on your system, then power up the unit.
Q: My Norstar Startalk, Flashtalk, or voice mail says my mailbox is not accepting any more messages. I deleted all my messages, but callers still cannot leave a message. Why?
A: Check with your voice mail administrator. Chances are another Mailbox on the system used up all the message storage space. Usually it's the general delivery mailbox that hasn't been checked for messages on a regular basis.
Q: My Norstar display shows "Message for you." I've checked my voice mail several times, but have no messages. Why?
A: Someone probably turned on the internal message indicator for your phone by accident. You can press the feature button and then 6 and 5 on your touch-tone pad to see who left you the message, or just press the feature button and then # and 1 on your touch-tone pad to cancel the message.
Q: Where do I hook music into my CICS or MICS system?
A: On the CICS, use the violet/green pair of the trunk amphenol connector- on the MICS use the yellow/orange pair of the 2nd station amphenol connector.
Q: Lines 1-4 on the MICS work fine, but I can't get the lines on the 2nd LS/DS trunk cartridge to work. Why?
A: You need to use lines 25-28 in programming for the 2nd LS/DS card slot in the Main KSU. Lines 1-24 are dedicated to the 1st card slot in case T-1 is used, therefore the starting line for programming the card slot is line 25.
Q: Does my Norstar System support T-1 with PRI?
A: Yes, if you have an MICS system. But you also require a keycode to enable the PRI function.
Q: Can I hook up a standard phone to my Norstar system?
A: Yes, but you need to use an ATA that will plug into a station port on your system. Your standard phone then plugs into the ATA.
Q: Can I use an "old style" IPE shelf on my Meridian 1 system?
Why would I need a "new style" shelf?
A: The "old style" shelf only cables out 16 ports per card slot which will run most of Nortel's current cards. If you are using a card that has 24 ports, you will need a "new style" shelf that cables out 24 ports per card slot.
Q: Can I use ground start lines on a Norstar System?
A: No. The Norstar system is only compatible with loop start lines.
Q: Can I get CDR output from my Norstar system?
A: Yes. By adding an optional NT8B95 SMDR unit. The unit hooks up to a station port, and gives line-by-line output of call records through its 9 pin serial connector. This can be hooked to a PC or some other data collection device to make use of the information.
Q: Does the Norstar system hold it's programming through a power outage?
A: Yes, a battery backup is supplied on all Norstars that will hold programming info through a power outage up to 3 days, provided the unit has been plugged in for 3 days or more to charge. (Note: The software cartridge, which contains the battery backup, only has life span of about 5 years. So if your system has been installed longer than that, you might want to think about replacing the software cartridge.)
Q: How many expansion cabinets can be added to an Option 11 Meridian 1 system?
A: For an Option 11 (NTAK01 CPU card) you can add 1 expansion cabinet. For an Option 11E (NTBK45 CPU card) you can add 2 fiber expansion cabinets, for an Option 11C (NTDK20 CPU card) you can add 2 fiber expansion cabinets, but if you are running release 24 or higher software, you can add 4 fiber expansion cabinets.
Q: How do I use DID analog or T-1 trunks to call directly to a phone on the Norstar system?
A: You need to program a target line on each phone that you want a DID call to go to. The target line is a dummy line number in the Norstar numbering scheme. Then you program the DID digits the central office is going to send you in the received number field for each target line. (Note: Don't forget to set the received number length under your system data to match the expected number of digits from the central office.)
Q: I can't log in on my Option X system terminal and am sure my password is correct. I keep getting OVL400 output. Why?
A: Either the parameters for the terminal you are using are set incorrectly, or someone has tried to log in on that terminal with an invalid password and exceeded the logon attempts threshold. You will have to verify terminal setup and/or wait until the lockout timer has expired and try again. The lockout timer can be set anywhere from 0-270 minutes.
Q: Can I hook an external ringer or bell to the auxiliary ringer hook up on my Norstar KSU?
A: No. This is only a set of contacts meant to control a relay unit that in turn controls a bell ringer.
Q: How can I tell what release of software I'm running on my Meridian 1 PBX?
A: If you know how to log into your system terminal, you can go to LD-22 and at the REQ prompt, type in "ISS". This will output the release info for the system. Otherwise, you can look at a spare copy of your system diskettes, or pull the one from the system drive to check it and then put it back in. Both types of diskettes will have release info printed on them.
Q: Can I use my computer's modem on my Meridian 1 PBX?
A: Yes. You will need to have a jack set up with an analog port hooked to either an NT8D03 or NT8D09 card in the PBX. Make sure when programming in LD-10 to make CLS-WTD to stop the warning tone of a 2nd call from interrupting your modems communication.
Q: What card do I need to add to my Meridian 1 PBX to hook up more outside lines?
A: You need the NT8D14 card to hookup any type of analog line except E&M tie-lines. They require the NT8D15 card.

Tip & Ring

Meridian Business Set (MBSII) solutions equipment
Q: Do the MBSII sets require multiple line cards in the central office?
A: Although the sets have multi-line capability, they require only one line card.
Q: What central office switching systems are compatible with the MBSII sets?
A: With a 70%+ digital Centrex market share, Nortel's DMS-100 family of switches is the driving force behind the sets.
Q: Will they work behind the Meridian SL-100?
A: Indeed! The MBSII set is a versatile terminal.
Q: Are the user guides available in other languages?
A: They are available in English, Spanish and French.
Q: Do the MBSII sets offer the convenience of on-hook dialing?
A: Absolutely! Each set provides on-hook monitoring of calls while dialing, on hold, etc.
Q: Is 2-way speakerphone capability available?
A: The M5316 provides excellent quality, third generation, and full duplex technology.
Q: What is the loop length recommendation (distance from the Central Office to the customer premises)?
A: Roughly 3 miles.
Q: Does the red triangle in the upper right hand corner of the MBSII set provide visual voice mail indication?
A: The red triangle is a visual ringing indicator. Voice mail indication is through the black diamond appearance next to your message key.
Q: So if my ringer volume is turned completely down, can I still tell when I'm receiving an incoming call?
A: Yes. With the exception of the M5008, you can still catch that important phone call even with the ringer turned all the way down.
Q: Can I use a headset with all the MBSII sets?
A: Although you can use a headset with any of the MBSII sets, the M5216 was especially designed with a headset jack built right in. It is perfect for high volume, call handling attendant applications or in customer service Automatic Call Distribution (ACD) environments. The other sets require an adapter to accommodate a headset.
Q: My customer is interested in branding the MBSII sets with company logos. Can Nortel customize that market requirement?
A: Without a doubt! MBSII sets can be customized with brand/logo inserts.
Q: Do customers have a choice in colors?
A: The MBSII sets are available in ash, black, or gray.
Automatic Call Distribution (ACD)
Q: What is ACD?
A: The basic concept behind Automatic Call Distribution is as simple as the lineup at your local bank branch. In the bank, rather than wait at individual teller positions for varying lengths of time, customers form a single queue. As soon as a teller becomes available, the person at the head of the line moves forward to be served. Similarly, Basic ACD processes telephone calls on a first come, first served basis. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the next available call center agent. When an agent becomes free, he or she services the first caller in the queue.
Meridian 1 ACD does far more than simply process calls in sequence. A system can offer different kinds of treatment to different callers. People calling long distance, for example, can be given priority handling. Or customers placing orders can be distinguished from those seeking technical support.
However your system is configured, the ultimate goal is to serve every caller quickly and efficiently in order to meet customer service expectations. Those who do wait in a queue can be reassured with recorded announcements and/or music, and thresholds can be programmed into the system to minimize delays and divert calls to alternative queues to ensure that no one waits too long.
Internet Call Waiting
Q: What is Nortel Networks Internet Call Waiting?
A: It is an internet telephony application that allows Internet users to be notified of incoming phone calls while they are online, and it also gives control over the handling of the call.
Q: What do you sell?
A: Nortel sells a turnkey Internet Call Waiting product that is composed of a highly reliable industrial grade server and application software.
Q: Who is a customer for this product? What is the channel to market?
A: ICW can be implemented either by Telcos or Internet service providers (ISPs). The end user will buy the service from a provider.
Q: With ICW, does the customer lose their Internet connection in order to accept a call?
A: The user has an option to accept the incoming call via VoIP, which will allow the user to maintain the Internet connection, or via a regular phone which will terminate the Internet session.
Q: How does ICW work?
A: When the user is connected to the Internet, her incoming voice calls are redirected to the ICW server. The server extracts the caller and subscriber information from the call-signaling message and generates a pop-up window via the Internet to the subscriber's PC. The pop-up window offers the subscriber a number of options such as take the call, route to voice mail, or play a message.
Q: What server platform is used?
A: A high-reliability industrial-grade Windows NT server.
Q: What standards does ICW support?
A: Wherever feasible, we comply with industry standards. The platform has been certified by UL, FCC Class A/B, and CE. SNMP is used for OAM functions.
Q: How does the server communicate with the switch?
A: The server connects to the CO switch via a standard ISDN PRI interface.
Q: Can the system interface with non-DMS switches?
A: The system uses a standard ISDN PRI interface to the switch, which will allow inter-operability with other switch vendors.
Q: How does the system interface with the Internet?
A: The interface to the Internet is via a 10/100 Base-T Ethernet facility.
Q: What is the capacity of a server?
A: The server capacity depends upon the traffic characteristics of the end users. Our traffic studies have indicated that the current server capacity is in the range of 24,000 subscribers.
Companion System
Q: How does Companion differ from cellular?
A: The Companion System provides wireless communications for users at walking speeds in buildings. Cellular is better suited for wide area needs or users in high-speed vehicles.
Other advantages offered by the Companion System include:
*no airtime charges
*excellent voice quality
*security, anti-fraud and privacy features
*full integration with phone system
*networking capabilities
*full-feature functionality
*long battery life
*simple deployment
*host billing system interface
Q: Does the Companion System require special environmental conditions?
A: The Controller and Base Stations operate in a normal office environment with a temperature range of 32-122 F.
Q: Will the Companion System cause, or be susceptible to radio interference?
A: Unlike ISM-band systems, the Companion System operates in a dedicated radio band, reducing the likelihood of interference from other radio systems. Also, since this system operates at very low power levels, it is unlikely that it will interfere with other electronics.
Q: What happens to the Companion System in a power failure?
A: If the Controller or Base Station connects to a battery back-up system, the Companion Wireless Communications System will continue to operate without interruption. Otherwise the system will maintain administrative programming integrity for at least 72 hours.
Q: What is the extent of a Companion System's coverage?
A: The Companion System provides coverage for up to 10 million sq. ft., depending on a building's physical characteristics. For instance, elevators, walls and other physical barriers will influence the extent of the coverage area. Radio coverage provided by a Base Station is 3-dimensional. Therefore, a Base Station on one floor may be able to provide some coverage for the floors above and below.
Q: What happens if a user moves out of the coverage area?
A: When a user reaches the outer limits of the coverage area while on a call, the Portable Telephone will beep and the quality of the conversation will diminish. Walking back into the coverage area remedies the problem. If a user continues to move out of range, the Companion System will drop the call and transfer the user to a preprogrammed number, such as the attendant. If a user attempts to place a call outside the range of a Base Station, the system will not be able to establish the call.
Q: Does using a Companion Portable lose any PBX or Centrex functions?
A: Most standard 2500-type features are available to Companion Portables but some, such as a visual message waiting indication and calling party name display, depend on the host system's ability to provide them.
Q: Will a Base Station provide coverage outdoors?
A: Companion Base Stations often provide outdoor coverage because radio waves can travel through windows and doors. In some situations, however, it may require a separate, optional, external antenna mounted to its casing with coaxial cable.
Q: How does the Companion System support more portables than channels?
A: The same principles of traffic engineering for PBX/key system trunks apply to radio channels. Ordinarily, all Portables are not in use at the same time nor at the same place. Also, the System reuses frequencies. Its ability to reuse frequencies means that the same channel may be used for conversations taking place on two different floors because the cells are separated by enough distance to prevent interference.
Q: Can outsiders access the system?
A: No. Only Portables registered on the system have access.
Q: How does the audio quality compare to that of a wired phone?
A: The Companion system is a fully digital wireless system that achieves near wire-line sound quality. Users are often surprised to discover that its sound quality is indistinguishable from that of a conventional desk phone.
Q: Is there a charge for airtime?
A: Users within the coverage area do not incur any additional costs.
Q: Are they hearing aid compatible?
A: Currently, they are designed for use with hearing aids via the over-the-head headset.
Q: What is CallPilot?
A: CallPilot is a customer-developed unified messaging solution that brings together the core functionality of Meridian Mail voicemail with the ability to integrate email and fax to create a personalized, feature-rich messaging system. It makes use of leading-edge technology and standards such as advanced speech recognition and IMAP4 to solve real customer problems.
Q: What are its benefits?
A: The advanced capabilities translate into significant business benefits for end users. Starting with a complete voice messaging system, it can be easily enhanced to provide features including a single in-box for voice and fax messages. There are also three options for message management including touch-tone, speech recognition and PC access via popular email clients such as Microsoft Outlook and Lotus Notes. Simple voice commands can manage messages using the speech-activated user interface and voice. Fax and email messages can be managed from one convenient desktop interface, which leads to increased productivity. Additionally, it can be used with existing infrastructure and can save money by using VPIM networking technology to send voice messages over data networks and the Internet.
System Administrators will find it easy to manage and maintain. With Windows 95 graphical user interfaces for management, administration and reporting, this product takes advantage of client/server architecture to enable access to single or multiple systems without the hassle of being in the switch room or dialed in via modem. Application Builder also helps administrators build powerful messaging applications such as voice menus and fax-on-demand quickly and easily.
Q: What are some of the features?
A: Starting with the core call answering functionality of Meridian Mail, CallPilot adds Speech Activated Messaging (Speech Recognition), standards-based unified messaging that supports more email programs than any other commercially available product, including MAPI and IMAP4 Web clients, enhanced management interfaces including Application Builder to create voice processing applications, and Windows 95/98/NT administration and reporting for easier use.
Q: Who is the ideal customer for this product?
A: Customers who desire advanced applications such as Speech Activated Messaging or Unified Messaging, utilize multiple multimedia applications, and have large messaging networks with multiple messaging systems are in the best position to take advantage of the features.
Q: What is Speech Activated Messaging?
A: Speech Activated Messaging is the latest application of speech recognition technology. It enables management of voice and fax messages via telephone with speech commands instead of touch-tones. CallPilot changes the most commonly used touch-tone commands into intuitive speech activated commands: "play", "delete", "next message", "print", etc. This makes the user interface more intuitive and helps users to take full advantage of the features available to them including forward, reply and call sender.
Q: How does it differ from other messaging systems on the market today?
A: It incorporates standards to help reduce the total cost of ownership and enables customers to implement the features they need for their business. Standards such as VPIM (Voice Profile for Internet Mail), SNMP (Simple Network Management Protocol) and IMAP4 integrate seamlessly with our customers' existing communications network.
Unified messaging is now easy to implement. Adding fax or desktop users is as easy as enabling the mailbox, connecting to the LAN and adding the CallPilot Client software to the PC.
Q: Does it compete with or replace Meridian Mail?
A: It offers customers a choice of messaging systems. Meridian Mail is an excellent solution for customers whose messaging needs are voice only. CallPilot is ideal for customers who are seeking advanced multimedia capabilities within their messaging infrastructure.
Q: I am already using Meridian Mail. Can I just install CallPilot?
A: It can be installed on any Meridian PBX operating on X11 Release 23C. Customers have the option to bring the system on-line and move all their users at once, or have it run side by side with Meridian Mail to gradually move users to the new system. It is designed to integrate seamlessly into an existing Meridian Mail Network by offering both standards-based and proprietary networking options while keeping the same touch-tone user interface.
Q: What is the minimum release of Meridian Mail that will allow database migration to CallPilot?
A: Release 11 is the first one that supports direct database conversion to CallPilot. For customers not on release 11 or 12, CallPilot's AutoAdd feature enables you to do bulk loading directly from any tab or comma delimited ASCII file.
Q: Do I have to use a Meridian 1 to have CallPilot?
A: Initially yes, but we will be able to offer other PBX integration starting in 1999. In its initial release, CallPilot will work specifically with the Meridian 1 PBX operating on X11 Release 23C. Future connectivity options will include MSL-100, Norstar and other manufactures equipment.
Q: Why should a customer connect CallPilot to their existing data LAN?
A: It uses a connection to the data LAN to enable the delivery of voice and fax messages to multimedia computers via desktop messaging and administration of the CallPilot system from any computer on the corporate LAN. This connection can also be used to provide VPIM networking.
Q: Do users have to be on the same LAN as the CallPilot system?
A: No. Users do not have to be on the same LAN, but they must have TCP/IP connectivity via the WAN or Remote Access to the customer LAN or Virtual Private Network.
Q: What hardware and software options are available?
A: It was designed to offer our customers the ultimate flexibility both in hardware and software configuration. There are three hardware options available: the IPE, which offers up to 24 channels of messaging, and the Tower and the Rack, which support up to 96 channels of messaging.
The systems come standard with voice messaging. Customers then select the options that make sense for their business needs. Software options include fax messaging, desktop messaging, networking, and speech activated messaging. CallPilot's standards-based software enables customers to select the options their users need to communicate more effectively.
Q: What email packages does it work with?
A: It supports Microsoft Exchange, Outlook and Outlook Express, Lotus Notes, Eudora Pro and Netscape Messenger. CallPilot integrates at the client or desktop level by addressing industry standards such as MAPI and IMAP4.
Q: How are messages played and viewed with an email client?
A: It enables users to view the message header information in their familiar interface. To hear a voice message the user double-clicks on the message icon and indicates whether the message is delivered over the telephone or the TCP/IP connection to the PC. To view a fax, the user double-clicks on the message icon to invoke the Imaging for Windows application. The TIFF file is then downloaded from the Meridian Application Server and displayed on the desktop.
Q: How are messages retrieved remotely?
A: Through any telephone interface, users can access and manage their voice and fax messages through either speech-activated messaging or touch-tone commands. Users who have desktop messaging enabled and are connected to their network can use their PC to access and manage their voice, fax and email messages with desktop messaging.
Q: Can a user switch from a Speech Activated Messaging session to a DTMF session without logging off?
A: Yes. However, once they switch to DTMF they cannot switch back to Speech Activated Messaging. Also, all four MPU units required for a Speech Activated session will continue to be used.
Q: Will it network with other voice mail systems?
A: Yes. It utilizes the industry standards of AMIS and VPIM to integrate with other voice messaging systems. This allows connectivity to Norstar as well as other vendor voice mail systems.
Q: Does it work in a VPIM environment?
A: It is fully VPIM compliant, which allows it to digitally network with any VPIM Compliant messaging system including voice and fax messages.
Q: Does it support LDAP?
A: Yes. All of the unified messaging clients use LDAP (Lightweight Directory Access Protocol) to access the directory to look up users and create personal distribution lists.
Internet Voice Button
Q: What is Nortel's Internet Voice Button?
A: It is a value-added web tool, available through Internet Service Providers, that allows a Web-browsing consumer to call a company directly from its Web site while keeping his or her Internet session active. In addition to communicating by voice in real-time, the business representative and consumer may push Web pages to each other and engage in text chat.
Q: How do my customers call me from my Web site?
A: They simply click on a Voice Button placed on your Web page and within seconds their call to your business is established.
Q: How does the service work?
A: A business incorporates a Voice Button link on its Web site that, when clicked, passes information about the customer to the Voice Button server operated by the Service Provider. First time users are presented with a simple configuration screen, where they indicate whether they'd like to use a regular phone, Internet phone or request delayed connection. The parameters entered become the default settings for future uses, whether on this site or any other that incorporates the technology. The user may update settings at any time.
The Voice Button server then establishes a call to the customer using the method selected in the configuration. When the customer answers this call, the server initiates a second call to the business. The business and customer are then connected.
Q: Are my customers using a phone when making their calls?
A: If your customer has two phone lines to provide telephone and modem connections, he may choose to establish the call via regular phone. Once Voice Button is clicked, the phone will ring within seconds. If only one phone line is available to provide both voice and Internet connections, your customer has two options:
Internet phone - Placing the call using VoIP requires the person to have a multimedia PC with headset (or speakers and mic) plus Internet phone software such as Microsoft NetMeeting.
"Call me in X minutes" - Modem users with a single phone line may specify when their call will be placed (within a few seconds to a few minutes) and disconnect their current Internet session enabling the call to be taken via phone.
Both the regular and Internet phone options allow the customer to maintain an Internet connection while speaking with the business.
Q: Do I, the business, have the choice of taking calls on our existing phone system or via computer using VoIP?
A: Voice Button currently allows for the business to take calls by phone only. This makes the service as seamless as possible to the business, requiring no new hardware or software. Just take calls as usual.
Q: How good is the quality of VoIP?
A: The sound quality of VoIP calls continues to improve rapidly with new technology. However, the overall voice quality depends upon many factors including the user's PC and network connection.
Meridian HomeOffice II
Q: What is required on the PBX for Meridian HomeOffice II?
A: Meridian HomeOffice II requires an IPE module, digital trunking and support for Extended Digital Line Cards (XDLCs).
Q: What software release is required for the PBX?
A: Meridian HomeOffice II is compatible with Meridian 1 Release 17 and above and SL-100 BCS 32 and above.
Q: What are the PBX platforms for Meridian HomeOffice II?
A: Meridian HomeOffice II can be used with the following Meridian 1 or SL-100 PBXs:
*Meridian 1, Options 11C, 21E, 51C, 61C, 71C, and 81C
*SL-100, Options 111 and 211
*Older systems that have been upgraded with IPE modules
Q: What application(s) does Meridian HomeOffice II support?
A: Meridian HomeOffice II is designed for telecommuting, one of the three most common remote access applications (nomadic, telecommuting and LAN-to-LAN) in the enterprise space. Meridian HomeOffice II provides the following:
*Digital voice connectivity extended from the Meridian 1 PBX, with all features and functionality
*A LAN Ethernet connection into the corporate data network
Q: Can Meridian HomeOffice II place data network calls to multiple destinations?
A: Yes, Meridian HomeOffice II supports up to 32 user-specified locations, where each location is a distinct network. On the telephone network the Meridian HomeOffice II system is limited to on PBX connections per user.
Q: Can Meridian HomeOffice II place simultaneous data calls to different destinations?
A: Yes, but only two, and digital telephone must be in offline mode to allow this.
The two B-channels available with ISDN provide a maximum of two simultaneous connections. Meridian HomeOffice II is designed to support one data connection into a host environment when the digital set is active (that is, online to the PBX). Since the digital telephone uses one B-channel when in active mode, only one B-channel is available for a data connection. With the digital set in local mode (that is, offline from the PBX), two simultaneous data calls can be connected.
Q: Does Meridian HomeOffice II have analog phone interfaces?
A: Yes. The HomeOffice Router has one analog jack interface (port) which can be used to support analog devices (such as a fax machine) at the home office.
Q: Does Meridian HomeOffice II accept analog modem calls?
A: No. The HomeOffice Router does not have the necessary digital modems to support this type of functionality. However, you can connect a modem to the Fax port on the back of the unit to send and receive data and faxes from a PC. This would be identical to using the modem on a regular analog line.
Q: How do I attach my PC and digital telephone to the HomeOffice Router?
A: The Meridian HomeOffice II product is a bridge/router device that provides an Ethernet network connection. To connect your PC to the Meridian HomeOffice II product, install an Ethernet network interface card (NIC) and load the appropriate drivers on your PC. Use the Ethernet crossover cable (included) to connect the Ethernet card in your PC to the Meridian's Ethernet port. This provides a LAN connection between your PC and the Meridian HomeOffice II product.
The digital telephone is connected to the bridge/router device directly to the RJ11 port labled MERIDIAN on the rear of the product.
Q: How many analog devices (modems, fax machines, telephones, etc.) can be connected to the POTS (FAX) jack on the HomeOffice Router?
A: Any combination of devices that add up to a total of three Ringing Equivalency Number (REN), which usually equates to about three or four devices. Each of these devices acts as an extension off the phone jack. Only two devices and two B-channels can be active at the same time.
The greater the wiring distance, the fewer the extensions (exact specifications depend upon cable type, specification, and so on). Also, the greater the REN of the analog devices, the fewer devices the router can support off the port.
Q: What kind of devices can be connected to the HomeOffice Router?
A: The Router offers three different communications capabilities to the telecommuter: LAN Ethernet data connection, digital telephone set extension and the use of an analog device(s) connected to the telephone jack.
Q: Is bridging or routing a better solution?
A: Bridging or routing depends on your network configuration. As a rule, you must route between dissimilar network addressed LANs, and bridge between similar ones. The HomeOffice Router is especially suited for routing from a remote location into a LAN backbone. Obviously, not all network designs are the same. Talk with your network manager to determine your particular needs.
Q: Is there an easy way to configure Meridian HomeOffice II?
A: Yes, there is. The Meridian HomeOffice II is user installable versus technician installable. The telecommuter can use the friendly installation wizard, along with auto-SPID detection, to easily configure it The graphical user interface allows configuration in a point-and-click environment.
Q: What channel is used for voice calls, the B- or D-channel?
A: There are three types of voice calls that can be placed on Meridian HomeOffice II, whether it is digital voice or analog voice. They are: * The voice call made while the digital telephone set is "online" to the PBX * The voice call made while the digital telephone set is "offline" to the PBX * The voice call made via the FAX port on the HomeOffice Router. In any of these scenarios, the call is placed across one of the B-channels. The D-channel is used only for ISDN BRI B-channel call setup and tear down.

Nortel Networks authorized distributor channels can contact 1-800-4 Nortel for further assistance with questions. For more information visit

Reprinted with permission from Telecom Reseller magazine

Nortel Notes
by Phil Ruffin

Q: I need to add DID numbers to my Nortel PBX but all of the available numbers from my provider conflict with my dial plan. Can I modify the incoming numbers so they terminate on internal numbers that are different from the dialed numbers?
A: Yes, you can use IDC (Incoming Digit Conversion) to change the numbers. It will be confusing (at least it is to me!), but you can do it. In LD 49, establish an IDC table with the incoming number first on each line, followed by the internal extension number where you want to terminate the call. Then in LD 16, change IDC for the route to YES, and add the IDC table number at the DCNO prompt (for day mode) and NDNO (for night mode).
Understand though, that this may make you pull your hair out when you forget it's in place and you're trying to figure out why things don't work the way they should.
A better answer is to renumber whatever is necessary to allow available numbers to terminate on your PBX with matching internal numbers. Maybe you can renumber your trunk access codes or Park codes, or even some of the internal extension numbers. Print a DNB list (LD 22) and review the conflicting ranges of numbers to find the best match.
Q: I have three lines on my 2616 set, but calls don't come in on the third one. If the first two lines are busy, a third call will always go directly to voicemail. How can I get it to work?
A: Look at the LHK (Last Hunt Key) prompt in the station programming. You want it set to "2". While you're at it, set LPK (Last automatic line Preference Key) to "2" as well. This will allow the third line to be automatically selected, so you can answer it by lifting the receiver, if you have IRA (Incoming Ringing line preference Allowed) selected in the CLS (CLass of Service).
Q: My 2616CT cordless phone keeps losing its mind. My vendor came and reset the phone by removing the battery and unplugging the phone from power and the PBX for a few minutes. They said that's all they can do - it just has problems sometimes. Is there any way to fix it?
A: This problem sounds familiar. Look at the bottom of the set. Find the barcode label, and look for "Rls # 1" or "Rls # 2". Yours sounds like release 1. Inform your vendor that there is a free replacement program for the first release of the 2616CT sets. Tell your vendor to refer to Product Bulletin 2000-005 and "KPD # R99-H-34". That should be enough information to get results.
Q: I changed the base cord on my 2112 set to be able to move my phone farther from the jack, but now my speaker doesn't work. When I change the cord back, it works. Does my phone only work with a short cord? That doesn't make sense!
A: You are correct - that doesn't make sense. I suspect you will find the problem in the number of conductors in the cord, not its length. The 2112 set requires a special transformer and a six-conductor base cord to supply voltage to drive the speakerphone. Your new cord probably has only four conductors.
Q: Sometimes when I plug a digital phone into the wrong jack, it will make all the indicators flash together. Other times, it does nothing. Does this mean anything?
A: Yes, it means you are careless about where you plug your phones. You will be pleased to know, though, that the flashing indicators indicate that you are plugged into an analog port. This isn't 100% reliable, though. You should test with an analog phone before you declare it an analog port. I've seen defective digital ports with the same indication.
Q: Whenever I dial 181 on a phone it makes a loud beep and goes dead. The programming for the phone completely goes away. The only way I can get it to work again is to reprogram the entire phone. What can I do to make this stop?
A: I suppose you didn't think of this: sop dialing 181. No, that's too easy. It sounds like you've found a feature called ASR (Automatic Set Relocation). Actually, you only found the first half of the feature. The second half works like this: once you have ports prepared to accept the phones that have been "removed" from software using the ASR feature, you can plug one of these "removed" phones into the port, and it will magically reappear, reprogrammed on that port. Isn't that a neat feature? ASR allows you to move phones from port to port, and performs the basic programming changes for you. The feature code is SPRE (Special PREfix) + 81 + ASR Security Code. You seem to have a SPRE of 1, and no ASR Security Code.
You can make it much less likely that phones will keep disappearing from software if you program an ASR Security Code in the CDB (Customer Data Block). The prompt is SRCD (Set Relocation Security Code).
I suggest you use a combination of digits that will make it most unlikely for people to dial it accidentally. Some people use four zeros, since it would mean people would have to dial 1810000 to evoke the feature. Choose a set of digits that you can remember if you want to use the feature properly, and program it in LD 15.
Now that you have a grasp of how to use this feature, you need to know that each time you reprogrammed a phone instead of moving it by completing the ASR process, a set of programming was stored in memory, and it may still be there. Here's how to find it and remove it.
In LD 21, PRT the SRDT. This should give you a list of all the sets stored, waiting for them to be plugged into an available port and activated.
Next, use LD 50 to OUT the MTRT, followed by the old TN. Do this for each of the sets listed above in LD 21.
Is that powerful? You bet. You can move sets on the spot, without using a terminal at all. You still may need to follow up with changes in DES and any documentation you use, though.
Let me give you a little warning about this feature. Some people have experienced what they believe is memory corruption from using the feature. One story I heard came from a user who was moving a long list of phones using ASR. Just as the last phone was removed, the PBX initialized and all the programming was lost. The phones had to be restored by reprogramming them individually. I don't know about that being a memory corruption, though - I think he ran out of RAM in the PBX.
I haven't had an experience like that, but I would strongly recommend printing out all of the phones before relocating a long list, no matter what method you use. And if you do suspect a memory corruption in telephone programming, I suggest you LD 1. That will automatically run a routine to inspect the telephone configurations and (in some cases) correct memory corruptions that occur.
Q: We have a mix of 2008 sets without displays and 2616 sets with displays. Often, when a person leaves the company, his 2616 set disappears the same day. Is there an easy way to keep track of these?
A: Yes, you probably find the set when someone calls to have his display fixed. I usually check the programming first, to see if it's a 2008. If so, I go and confiscate the phone. I inform the user that he can have a 2616 set if he can get approval for the purchase, and I even provide the purchase request documents. The original 2008 will mysteriously show up within a day or two.
Q: Is it true I can use my phone to enable my TTY port?
A: Yes. Your phone must have MTA in the Class of Service. Use Key 0, and dial SPRE plus feature code 91. Using the telephone keypad, key in "LD#37##" and wait for the overlay to load. Key in "STAT#TTY##". Your TTY should be in the list, and it should show to be disabled. Note the TTY number, and key in "ENL#TTY#" (key in the TTY number here) "##". Check to see that the status changed by using "STAT#TTY##" again. Exit maintenance mode by using "****".
Q: I have several phones that are restricted from dialing local calls, but I need to allow them to dial 911. How can I do that?
A: I think this is a question many administrators and managers will (and should) be asking in the near future, with states and municipalities beginning to require 911 call capability from any public phone at business sites, and the answer is not necessarily an easy one. There are a couple of ways to do this, depending on the strategy already in place for the current restrictions, and the release of software available to you. Two dialing plans are common for 911 calls at businesses, and the lawyers have already chosen one of these for some areas. Before I get into the programming, let's talk about these two dialing plans, and you can decide which one to implement.
The choice of lawmakers in some areas (some say the FCC will require this eventually) has been to make the emergency call as easy as possible by allowing the user to simply pick up the phone and dial 911 to get emergency services, just like at home. That sounds easy, doesn't it? I have to admit that I was taken in by this at first, too. It's compelling to make it so easy anyone can just dial 911 any time from any phone without having to think about how to dial special codes, like an access code to get an outside line. And it can be easy to program in some cases. The trouble comes when the local police start to get upset at all of the calls placed to 911 in error. I discovered this problem myself when I implemented this strategy some years ago on an Option 61. Things sailed along just fine for a few days, and then people started making mistakes dialing calls. There was the manager who dialed 9-11+ instead of 9-011+ for an international call. Twice. Then there was the conference phone that sometimes dialed double digits. Several other calls were made to 911, and each time the emergency equipment arrived, and there was no emergency. Once, even I was discovered to be the culprit, testing a terminal program installation that wasn't configured properly. The police department got really irritated with us, and wanted to put a stop to all of these false emergencies.
By now, you may have figured out that this dial 911 only approach is not my favorite. While it is required in some areas, I choose not to implement it until the lawyers drag me into the switch room and give me a copy of the requirements. Until then, I program systems to dial the way most business workers expect. Dial 9 (or whatever local access code you normally use), then 911. "But wait!" some people point out to me. "What about the visitor in your building who doesn't know how you use your phones, but has an emergency and tries to dial 911?" I don't know how to answer questions like that. Maybe you need to give people 911 instructions when you issue a visitor badge. I still think most business people are more likely to dial 9-911 in an emergency at work than to dial 911. Ask your people, if you want to find out what will work best in your environment. Check your local requirements, too, to see whether the lawyers have already decided for you.
Some older systems may have the phones restricted by assigning a CLS (Class of Service) of FRE (Fully Restricted), SRE (Semi-Restricted), or some other that will restrict calls by type of connection (meaning station or type of trunk). For these systems, especially if you have software that is release 22 or older, you really should consider implementing BARS (Basic Automatic Route Selection) in order to properly allow 911 calls. It is a very flexible system and can be programmed several ways to accomplish the same result, so many people are confused by it. If reading the manual doesn't help, contact your vendor for assistance.
Also, a very capable gentleman who goes by the name GHTROUT has a great description on his web site. Go to and click on the BARS 101 link.
Beginning in release 23 (I think it was 23.55), a new way of programming this was introduced. With the new ESA (Emergency Services Access) feature, the PBX allows you to designate an emergency number for the system, and it gives that number priority over the entire dialing plan. The idea was to allow your users to dial 911 (or whatever emergency number your community uses) without any regard at all for the way the rest of the phone system is programmed. So the emergency number simply overrides all other routing or number plan information in your system. Pretty neat, I think. It also gives you a way to notify a guard or attendant when a 911 call is placed, and even prints out details on the 911 call on the system terminal. I really liked this feature, until I tried to program it so that calls could be dialed 9-911. It didn't work. Finally, someone pointed out to me that the book specifically states that you must have CAMA (Centralized Automatic Message Accounting) trunks for this to work with PRI (Primary Rate Interface) trunks. But it should work fine with all other trunks, including CO (Central Office) Ground Start Trunks. I haven't had the opportunity to check later versions of this feature to see whether Nortel has fixed the problem for PRI trunks. I expect it to work in later releases.
So, what will you do with this? Will you reprogram your system to use BARS to handle these calls? Will you use ESA to give priority to the calls? I invite all my readers to let me know what you're doing with this, and what trouble you find as a result.
Q: The 911 dialing information was helpful but I need more information for our environment. We have several locations tied together using tie lines and T1s. Some locations share a PBX and phone lines.
A: One way to handle 911 calls in a shared PBX is to use Tenant Services software (also called Multi-Tenant software). Multi-Customer software should also work. You can direct the 911 calls to dedicated local trunks (if your hardware will support them) so the call will carry the local phone number. If you need CAMA trunks to carry even more information, I think you will need to have CAMA to each PSAP (911 office). As a last resort you can use an external box that will provide the PSAP with the information it needs, but it requires more ongoing administration.
In cases where you have PBXs tied together, the solution is easier, since that PBX can send such calls out on local trunks with few programming changes.
I think that this would be a fun project to work on.
Q: What amplified handsets are compatible with the 2008 and 2616?
A: You probably don't need an amplified handset with those phones. You can use the built-in amplifiers. While talking on the handset, increase the volume by repeatedly pressing the right side of the volume bar at the bottom of the face of the set. I haven't needed any additional amplification for anyone yet.
Q: One 2616 set in our office doesn't work as a speakerphone. I swapped phones, so that isn't it. The speaker works, but the person on the other end can't hear me when I test.
A: This should be an easy one. You probably just need HFA (HandsFree Allowed) in the CLS (CLass of Service) instead of HFD (HandsFree Denied). If you have a feature programmed on key 15, it will go away. Key 15 is changed to Handsfree/Mute when you activate the Handsfree feature, just as key 7 is changed to Program when you activate the display.
Q: What can I do to allow Last Number Redial on the phones where I don't have a spare key to use?
A: On all the digital sets except the 2317 or 3000, you can use the feature without assigning a key. Just add LNA (Last Number redial Allowed) to the CLS (CLass of Service), and set LNRS (Last Number Redial Size) to the appropriate number of digits, and the feature is programmed. To use this press a line key twice, or lift the receiver and press the active line key.
Q: All our phones sound alike when they ring. How can I change the ringers so they have different sounds?
A: In the CLS (CLass of Service), you probably have DRG1 (Distinctive RinGing 1) on all the sets. This is the default - a high frequency, high-speed oscillation. You can change the programming to DRG2, DRG3, or DRG4 to apply a different ring to the phone. I usually set up a demonstration area with the 4 rings clearly identified so users can determine which ring they want, then have them email me with their preferences.
Q: I have a group of phones set up with a MCR (Multiple Call Ringing) key to share calls for software support. Still, callers sometimes get busy signals. Shouldn't it always be available for calls until the keys are busy on all the phones?
A: Yes, this is a problem. It really doesn't work like ACD (Automatic Call Distribution). I don't know a good solution for this, since it should work, but doesn't. Have you considered using real ACD? It gives you better control, plus reports.
Q: I can't get the Hot Line feature to work. I'm programming key #8 with "7 HOT D 4545".
A: You're making a common mistake. After the key number "7" and "HOT D", you must insert the number of digits that will follow before you enter the digits themselves. There are four digits in the number 4545, so that would make your command, "7 HOT D 4 4545".
Q: Is it possible to program a different function in the place of the Handsfree Mute key? I get an error whenever I try to program it. It happens when I try to replace the Program key, too.
A: Yes, it is possible, but do you really want to? The Handsfree Mute key is a requirement if the phone is to be used as a speakerphone. To eliminate that function, use HFD (HandsFree Denied) in the CLS (CLass of Service) instead of HFA (HandsFree Allowed). After you make that change, you will be allowed to program a different function on that key.
In the same way, the Program key is necessary if you have a display on the set. You can program NDD (No Digit Display) in the CLS to eliminate most of the display functions, even if you have a physical display on the phone.
Q: What is MARP, and why would I want to activate it?
A: I've heard this question many times, and it amazes me sometimes that this feature is still optional. I frequently find systems with MARP (Multiple Appearance Redirection Prime) not activated, but I can't think of a single reason NOT to activate it when the system is first installed.
MARP only applies to situations where one extension appears on more than one phone. This is commonly called a Multiple Appearance DN (Directory Number). You might find this arrangement on a boss's phone and his secretary's phone when the boss wants the secretary to answer calls for him while he's busy (or at the golf course). This is a useful strategy for some manager/secretary situations where the secretary needs to have full control over the boss's calls, or where the secretary just needs to know when the boss is on the phone.
Before MARP, it was a puzzle to figure out which phone would have control over the behavior of the shared DN. That's why MARP was invented - to make it easier to be certain what phone controls each DN.
Consider this example. The boss and secretary share extension 3000, and the boss gives the extension to his close friends, and even his wife. Today, the secretary is out, and the boss is working in a conference room, where he has the budget spread out all over the table. He forwards his phone to the conference room, and plans to answer his calls there. Uncertain of this technology, he even tests this arrangement by calling his extension to ensure it rings on the conference room phone. Only later will he learn that his shared extension 3000 does not forward with the rest of his numbers. Why? His phone is not in control of that DN on his phone. Do you see what a mess this can become in a system that doesn't have MARP activated?
Without MARP, only math and tealeaves will show you which phone is in control of a shared extension. Be prepared to list all appearances of a shared extension and look for the lowest numbered TN (Terminal Number) and the most recent date. Be aware of whether or not the DN is included in the Short Hunt (Last Hunt Key or lower). And expect to be required to deal with this every time you make a change on any of the phones with the shared DN.
When you have MARP activated, this is a simple matter. The first phone programmed with the extension is the default MARP, the one in control of the extension number. When you program an additional appearance, the system will ask you whether to make this appearance the MARP, or controlling extension. If you just return, it assumes you responded NO. If you answer YES, the new phone will control the extension.
"What do you mean by control?" you may ask. By control, I mean forwarding to another location (as in the example), or hunting (busy), or FDN forwarding (no answer).
Before MARP came along, companies would sometimes resort to programming analog phones with shared DNs, in order to control call handling. This helped clear some of the confusion, but led to higher cost from using more TNs (Terminal Numbers), and made software changes more complex.
Here's a word of warning, though. If you have a large system that does not have MARP activated, don't just turn MARP on and then leave on vacation. It's likely that some of the shared extensions will not transition properly, and you'll need to reprogram MARP on those.
Otherwise, I suggest you always use MARP from the time the system is installed. If the boss knew how much time you will save by using MARP, he would thank you (when he gets back from the golf course).
Q: I get complaints about how hard it is to set up a conference on the speakerphones. How can I make it easier, so they don't disconnect people?
A: Oh, yes, the analog conference phones. What would we do without them? Not that we wouldn't like to try.
I have a trick I've used several times that users like.
If you can associate a secretary (or someone else who can follow instructions) with the conference phone, put its DN on her phone. The secretary can use that key to establish a conference, and then have someone in the conference room pick up the call on that phone. She gets to use the conference key to establish the conference, so it's much easier.
Another advantage to using this method is when someone needs to be added to the conference after the meeting starts. The party will usually call the secretary, wanting to join, and the secretary can use the Call Join feature to put them together. She (or he) will press Conference, press the conference phone line key, then press Conference again. She can then hang up and allow the conference to continue. It's pretty slick. When you first try it, though, you may have a hard time hanging up on the conference, for fear that it will disconnect them. But as long as the conference phone has a COS (Class Of Service) setting that allows the connections you made, the call will continue on the speakerphone with no trouble.
Q: Why doesn't my Nortel phone have a Drop key like Lucent phones have?
A: You're correct that the Nortel phones don't have a Drop key. There are a couple of ways you can duplicate the function, though, and even improve on it.
If you have release 23 or later, you can drop one party (conferee?) from your conference by using the Conferee Selectable Display key (on a 39XX phone it's labeled "ConfSelDsp"). It actually works better than the Drop key, because you can drop ANY party from the conference, not just the last one added. You scroll through the list (you have to have a display to do this) and use the display information to select the one to drop.
You can also use the No-Hold Conference key instead of the regular AO6 Conference key. The No-Hold Conference key works like the Conference key you're used to, except it doesn't "set aside" or "put on temporary hold" the other party or parties while you add another one. The existing conference hears everything while you add another party. As a result, you don't have to press Conference the second time to complete the conference until you're sure you don't want to Drop that party, but everyone can talk and hear each other. I personally don't like the No-Hold Conference key for myself, because it doesn't give me the privacy I want, so I can see whether I really want to add this new party to the conference.
Q: My system won't let me delete a secretary's phone, and I think it's because of something called BFS. What are BFS keys?
A: BFS (Busy Forward Status) keys can be pretty useful, and sometimes frustrating. In a way, they operate like DSS keys on a key system. A single key allows you to see when a phone is busy, and you can transfer calls to that station by pressing Transfer, then the BFS key.
If you're a secretary and you want any calls to the boss's phone to come to your extension instead, just press the BFS key on your phone. It sets up his phone to forward all calls to you. You and anyone else who has a BFS key to it can see by the flashing BFS key that his calls are being forwarded.
The frustrating part comes in when you try to copy a set that has a BFS key, relocate one of the sets using Automatic Set Relocation, or move the set. It just won't work, and you get an error.
Secretaries often don't fully understand the BFS key a day or two after being instructed (maybe that's my fault). They invariably get calls forwarded and can't figure out why. I still like the key and recommend it for certain situations, but I expect to have a few calls that indicate training is necessary again.
Q: When I'm talking on my 2616 set, I sometimes want to mute the handset and listen while I talk with someone else in the room. I know I can put the call on the speaker and mute it, but I want to keep it on the handset. How do I do that?
A: You don't. At least, you don't with a 2616 set. If you have release 25 software, you can look at moving to the new 3904 set. It has that capability.
Q: I have seven phones set up with the same extension as SCR (Single Call Ringing), but I'm trying to change them to MCR (Multiple Call Ringing). When I try to change one, I get an error.
A: You can't have a combination of SCR and MCR for the same extension, even while changing them. You must NUL (null) all of the instances of the SCR before programming the MCR.
Before you do that, though, are you sure you want to use MCR? In North America the feature has a bug that doesn't appear in the international versions of the software. The effect is that only one call can be ringing at once on the MCR keys. The second caller hears a busy. If this is a sensitive application where you never want to get a busy signal, you may want to find another solution.
Q: When I look at the FFCs (Flexible Feature Codes) in LD 57, they're all blank. I want to know the default codes for features such as Call Park and Call Pickup, but they don't print. How can I tell what the default codes are?
A: I couldn't find a good list in the NTPs, so I got this list from another user. I'm giving you the LD 57 abbreviation, the code, and the feature name.
CDRC: *46, CDR Charge Account Code
CFWA: *21, Call Forward All Calls Activate
CFWD: #21, Call Forward All Calls Deactivate
CPAC: #76, Call Park Access Code
CPRK: *76, Call Park
C6DS: *30, Six-Party Conference
ELKA: *57, Electronic Lock Activate
ELKD: #57, Electronic Lock Deactivate
HOLD: #40, Permanent Hold
CFHO: *60*, Call Forward/Hunt Override
RPAX: *81, *82, Radio Paging Access
RPAN: #82, Radio Paging Answer
PUDN: *43, Pick Up DN
PUGR: *44, Pick Up Group
PURN: *42, Pick Up Ringing Number
RCFA: *22, Remote Call Forward Activate
RCFD: #22, Remote Call Forward Deactivate
RDLN: *80, Redial Last Number
RDNE: *53, Redial Number Erase
RDSN: #54, Redial Saved Number
RDST: *54, Redial Store
RGAA: *37, Ring Again Activate
RGAD: #37, Ring Again Deactivate
SPCC: *51, Speed Call Controller
SPCE: #50, Speed Call Erase
SPCU: *50, Speed Call User
SSPU: **, System Speed Call User
TFAS: *83, Trunk Answer From Any Station
LILO: *40, Login-Logout for 500/2500 ACD Sets
NRDY: *45, Not Ready Activate/Deactivate for 500/2500 ACD Sets
GHTA: *48, Group Hunt Termination Allowed
GHTD: #48, Group Hunt Termination Denied
Voice Mail
Q: My Meridian Mail system must be haunted. When I listen to messages or leave someone a message using my speakerphone, it interrupts me or suddenly decides to "send" the message or other actions. If I pick up the handset, it seems to work ok. What can I do?
A: For starters, you could just use the handset when you're in Meridian Mail. No, that was too easy. Your Meridan Mail is probably hearing other sounds in the room and interpreting them to be tones, as if you are dialing commands from the phone. It may be a radio, or overhead paging sounds, or any other noisemakers that you may not even notice. I had a problem like that with another brand of voicemail some years ago that interpreted the voice of one user as tones. When I explained to her what was happening, she altered the pitch of her voice talking to voicemail and the problem went away.
Q: One executive here says his Meridian Mail mailbox is haunted. Messages just disappear before he gets to hear them. I'm not getting such complaints from anyone else in the company.
A: There are two answers to this question. The quick and easy one is to have the manager change his password and DON'T TELL ANYONE the new password. Really. Chances are someone else is checking his or her messages. This may be someone who guessed the password (or saw it dialed), or it may be a secretary or family member.
The other solution is to be used in cases where you want detailed information about the voicemail settings. Log into your Meridian Mail using the password TOOLS instead of your regular one. It will prompt you for your password afterward. Then select Session Trace. You will be able to step through all activity on the mailbox for a set amount of time. Read through all the information carefully, and you can tell when each activity occurred, what happened, and in many cases what extension or telephone number placed the call. It's almost as good as listening in on the sessions.
Q: I need help building a menu in Meridian Mail.
A: Here's one I like. It allows callers to log into their mailboxes, leave messages by number or name, and dial extensions by number or name. I have to make some assumptions, so you may need to adjust several things to make it fit your situation.
Assume you have extensions in the 4xxx range. Use 4444 for this menu.
In Meridian Mail, start in 3-Voice Administration, 4-Voice Services Administration, and 4-Thru-Dial Administrations. Select Add, use Thru-Dial ID 0001, Title is 4xxx Extensions. Dial By Number, DN Length is Fixed 4 Digits, Left Pad 4, Suppress Prompt No. The Restriction/Permission Set should be one that only allows internal calls.
Add a second Thru-Dial Definition with an ID of 0002 and a Name of Names. Dial by Name, and Restriction/Permission Set again that only allows internal calls.
Next to 3-Voice Administration, 4-Voice Services Administration, 6-Voice Menu Definitions. Select Add, use Voice Menu ID 0003, Title is General Menu. Skip down to Key 1, Action VM, Comments Login Mailbox. Key 2, EM, No Mailbox ID, Comments Leave Message. Key 3, Action TS, Thru-Dial ID 0002, Comments Dial By Name. Key 4, Action TS, Thru-Dial ID 0001, Comments Dial 4 . Set all the unused keys to RP. Move your cursor to the No. that follows Greeting Recorded (Voice) and select Voice. Enter the extension number of a nearby phone, press Enter, and answer the phone. Select Record, and use a greeting like this: "Welcome to Sprockets Inc. You can dial your extension at any time. Press 1 to log into your mailbox. Press 2 to leave a message for someone. Select 3 to dial by name." Select Stop, and review your recording. Be sure to save it when you get through.
Next to 3-Voice Administration, 4-Voice Services Administration, 1-Voice Services-DN Table. Select Add, use Access DN 4444, Service MS, Voice Menu ID 0003.
In the PBX, LD 23, REQ NEW, TYPE ACD, ACDN 4444, MWC NO, MAXP 1. Skip down (keep pressing Enter) to NCFW, and enter the main number for your voicemail.
That should do it. When you dial 4444, the menu plays. Test all the choices before giving the number to anyone. Make sure callers can't reach an outside line through BARS or access codes.
You can add a toll-free number that points to 4444, if you like. You can also add hidden functions that are not given in the greeting.
Q: I'm trying to build new mailboxes in Meridian Mail. I've added a couple of new Classes of Service and they show up in the Administration screens, but when I go to build a new mailbox the only option it gives me for Class of Service is Personal. Is there something I have to do to activate the new Classes? I've already shut down and re-started the mail, but it didn't seem to do anything useful.
A: Yes. You need to tell the system which Classes of Service to offer when a mailbox is programmed. From the Main Menu, select General Administration, then General Options. Add the new Class of Service numbers in the line that says Class of Service Selection.
Q: I recently changed the primary DN (key 0) of a 2616 set. There is no voicemail box associated with the DN, nor the old DN that was on that particular phone, yet the light is lit. I reset the phone and also searched for any phones where the primary DN differs from the MWI DN and found nothing. How can I get the message light to go off?
A: There are two issues here: turning off the light and keeping it off. To turn off the light, do this. Change your personal voice mailbox so that the message light DN (Directory Number) is the number now programmed as key 0 on the phone. Call your number and leave a message for yourself. Now call voice mail and retrieve the message, and any other messages you have. The light should go off. That takes care of the immediate problem.
Next, you need to check to see whether any other mailboxes are set to turn that light on and off. You said there are no mailboxes with the current Prime DN, so (assuming you have Meridian Mail) use the Find feature to determine whether any of the mailboxes has the Prime DN listed as the Message DN for that mailbox, and if you find one, change it. That should take care of the problem for awhile.
Q: Our new Meridian 1 system will be different from the old Rolm system everyone is used to. Can I set up the system to work like the Rolm so the transition can be easier?
A: There are a couple of areas where you may be able to ease the transition for users. One of them depends on the voicemail system you have. If you elected to use Call Pilot, you may choose a set of feature codes that mimic the Rolm PhoneMail operation. If you're using Meridian Mail, you won't have that flexibility.
The operation of the phone system will never be exactly the same as the Rolm system, but you can modify the feature codes your users will access to simulate their familiar environment. For instance, you can make the Group Pickup code **3 and the Directed Call Pickup code *3, just like the Rolm systems I used to work on.
You define the Flexible Feature Code block in LD 57. Check your documentation to find the features that will relate to the practices of your users.
Q: Is there any reason to use Nortel's training instead of another company, like Global Knowledge?
A: As a matter of fact, Nortel recently outsourced its training to Global Knowledge. And yes, there is an advantage to using Global Knowledge. If you need Nortel certification, that's where you get it. If authentic certification is not necessary, other companies provide adequate training for some products.
Q: If I connect my Core card to Ethernet, can I FTP to the PBX to do administration?
A: No. There is no FTP daemon running in the PBX. Nortel wants you to buy Meridian Administration Terminal or its successor, Optivity Telephony Manager, for this.
If you connect the PBX to your Ethernet LAN, be sure to block broadcast messages from reaching the PBX by using a router. A router seems to be the only acceptable way to block those nasty broadcast messages that can take your PBX completely out of service.
Q: My dial-up connection to the PBX works fine, except for the rude interruptions that occur on the hour. I'll be typing in changes, and at the stroke of the hour, line after line of numbers spew onto my screen. It keeps me from the work I need to do. What can be done about this?
A: It sounds as if you are receiving traffic reports through your modem connection. If you don't have a device calling into your modem to receive these reports, you can turn them off for the modem port.
First, you'll need to know which TTY port is used for the modem. In later software, the system will indicate which port you've reached when you first log into the system. That information in hand, you can print the Configuration Data Block in LD 22, at Gate Opener ADAN. Look at the TTY for the modem and see which entries follow the USER prompt. TRF indicates Traffic will be sent to this port.
To change it, use LD 17 and Gate Opener ADAN. After the prompt USER, enter XTRF and whatever other entries you need to remove irritating messages you don't need.
Be aware, though, of what messages you turn off. These may come back to haunt you, so be prepared to add them back.
Q: We play classical music for music on hold. We're considering using the radio instead, but people at the radio station said we might need to pay licenses to use it. Why doesn't the radio station want us to use their material?
A: You will be disappointed with this answer. You probably are already broadcasting copyrighted music without a license. The owners of the performance on your classical recordings have probably listed them with either BMI or ASCAP. You need to either get a license to use the material, or begin using material that doesn't require a license. Contact them at or to get more information.
I want to say just a bit about this license issue. For music performers or others who receive royalties on music, this is a sensitive issue. You have chosen their music to play day after day for your customers. This implies that you receive some benefit from the playing of someone else's music for your business. You should really feel obligated to pay the royalties due to these creators of the music you provide for your customers who wait on hold. If you don't, then you should feel obligated to avoid the big fines your company will pay if you get caught.
Q: Can I have more than one kind of music on hold in my Option 61? People don't seem to agree on the music style.
A: I don't think I can help you pacify all the listeners, since few situations allow the user to listen to his or her choice of music on hold. You could only offer that choice when calls are under the control of CCR or Symposium, by adding the menu item to your script.
I have personally arrived at a choice that gets few complaints. I play orchestral CDs of John Williams conducting his movie themes and also Disney songs. I avoid any singing, since this gets many complaints (it seems everyone hates at least one style of vocal music). I also eliminated one CD that was otherwise very good, but had an extended drum solo on one of the cuts.
You can choose a different music source for each trunk route. This is easy, since the route programming includes a prompt for the music route. Just set up additional music sources that terminate into trunks in new music routes, and use those route numbers for different trunk routes.
You build new music routes in LD 16. Print out your current route in LD 21 and duplicate it when you create a new one. You'll need a trunk port for each, set up like the one assigned to your current route. You will also need a CD player or some kind of music source to connect to the trunk you built.
Q: Surely there's a better way to handle the administration of my Nortel Option 11C. My vendor installed this dumb terminal with a printer attached, but scrolling back is difficult and very limited. With some trouble I can print out information on a dot matrix printer, but it needs some Windows features!
A: I agree. The terminals don't provide a state-of-the-art interface. I have a PC connected to a serial port on mine, and use Procomm Plus to talk to it with a VT 320 emulation. I have the session defined for a HUGE scroll-back buffer, and the mouse makes it much better. I can scroll back and view days of activity at a time.
With your Option 11C, you can create a backup of the system on your hard drive through this PC, and you can connect the PC to your network to use network printers.
You can also get MAT (Meridian Administration Terminal) for a Windows-based tool. If you use Optivity for your data equipment monitoring and administration, you'll be glad to know Nortel will soon release Optivity Telephony Manager, the successor to MAT.
Q: Occasionally, I set up a 500 set to forward to an outside number so we can dial the extension and reach the outside number. So far, the only way I know to make it work is to program the 500 set, then clip a test set onto the port and dial the codes to forward calls to the outside number. Isn't there a way to do this using the administration terminal?
A: Yes, there is a much better way. Create a Phantom Loop (if you don't already have one) and program a Phantom TN for the application. Program a Default CFW for the set. The line will look like this:
FTR DCFW 11 918008795866
In the above example, the 11 tells how many digits, and the following number is the one that will be dialed. If you have Remote Call Forwarding set up, you can even change the destination number remotely.
Q: Where can I download upgrades to the firmware in my Meridian HomeOffice II routers and line cards?
A: I don't know. Nortel moved the files, and I haven't found anyone who knows where they are now. If any readers can help me locate these, please send me an email.

Phil has worked on both sides of the house, having been a field technician servicing Nortel and other PBX and key systems, and also having managed networks of PBXs. He is active in his local SL-1 Users organization and has attended INNMUG. Phil can be reached via email at

Reprinted with permission from Telecom Reseller magazine.

Definity Demystified
by Walt Medak

Trunk Card
Q: Our facility has grown outside our campus and we must expand to another location several blocks away. We have been told that we can connect our current Definity to an Expansion Node if we can run fiber-optic cable between the two sites. Our initial concerns of having the remote die if there were a major problem with the main node was put aside with the proposal of an EPN w/Survivable Remote. Are there any other options to connecting the two sites besides having to pay for the installation of a quarter-mile of fiber?
A: An Expansion Processor Node (EPN) is one way to connect two physical locations to have a single virtual office. Calls can be made between sites via extension-to-extension, one Attendant Operator can field calls for both locations, etc. A better alternative would be to install a separate Definity G3(x) at the remote site and connect them via T-1 utilizing the DCS features of the Definity. There might be more cost involved for software, but my guess is that it would be close to a wash against an EPN with Survivable Remote. It is a survivable remote in itself, and much, much easier to administer on-going and has all the benefits a Survivable Remote EPN can give.
Q: We constantly are being called by numbers of companies selling long-distance and local alternative service to our current providers. We are unsure of what to believe, and the sales people we have been talking to don't ever seem to be knowledgeable about the Definity system. Is there any rule of thumb for how many and what kind of trunks we need?
A: That's a big question with multiple answers, but mostly a fairly simple one. The Long-Distance "Fair Wars" have been going on since the '70s, and have had various incarnations as to what's the best/cheapest mode of trunking. We've gone from strictly CO trunks to WATS trunks, to Tie trunks, and on and on. For most systems in this day and age, one trunk group would suffice for Local and Long-Distance calls using a switched "Pick". Many providers are offering that L.D. service for about ten-cents per minute, making the installation of any other trunking, such as a dedicated L.D. T-1, more costly than it's worth unless your call volume is sufficiently large enough to substantiate that overhead. That's where most of the Long Distance "Agents" who are calling you are weak in expertise, and you should talk to a consultant or very knowledgeable tech. Use the "list measurements" command to determine your current trunking usage, and your call-accounting package, if you have one, to determine if you are over or under trunked. The BIG thing to be aware of is to combine all the trunk groups you have into as few as you can, such as one or two. Doing this will require fewer trunks in your system.
Q: I am installing a new ISDN-PRI two-way DID trunk group in my Definity-G3i, and am not able to get the trunks to work. The D-channel status is "in-service", and the provider says they show their D-Channel and trunks as in-service, and that we must have a problem with our switch. The Definity shows the trunks as "OOS-FE PINS". What are we doing wrong?
A: Probably nothing. I have this "thing" about network providers pontificating their absence of responsibility during events such as this. The key definition of where the problem lies is in the "FE" portion of OOS-FE PINS. FE stands for Far-End, and that may be where the problem lies, or in your not defining the protocol they should be using to match the Definity. Essentially, the Definity best uses Protocol Version "a", which is located on the DS1 form under the Country Protocol. "a" is known as "custom", and "b" is known as National (or NI-2). If you have defined the Definity as "a" (the default) and the provider is defined as NI-2, you will get exactly the results you are getting. In some cases, the provider isn't sure of what "custom" is, and will fight to get you to change the Definity to NI-2. I have had problems with this, and have always reverted to "custom", which they can usually support. If you get resistance to this, get one of our staff on line to work with the provider. They can normally convince them to see it our way.
Q: We are moving our system to our new corporate offices, and are changing our trunking in the process. Currently we just have Central Office analog trunks and a T-1 from our Long Distance provider. DID(Direct-Inward-Dial) trunks over a 2-way T-1 by our network provider was proposed to us, and though we think we would like DID, are unsure of what the heck it is that's being proposed. Will a 2-way DID trunk group work on a Definity, and what do we need to do to administer it?
A: DID is a protocol that most network providers and PBX manufacturers have agreed upon and conform to for delivering calls directly to an extension without intervention from an Operator or Attendant. It works very well on the Definity, and even the System75. It's a protocol dating back as far as processor driven telephone systems. Essentially, the call is proceeded with the last 3 or 4 digits of the telephone number from the network provider to the Definity, which then identifies them and routes them to the extension, hunt-group, etc. that has those matching digits at its number. That's one of the nice things about the design of the Definity; it uses extension numbers as its means of addressing. You have nothing to administer once the trunk group has been assigned, as it's seamless in the Definity. The 2-way part is a function of the T-1 and the network provider's switch allowing both incoming and outgoing calls to utilize any trunk for either. It is a very good service that usually has a substantial cost savings, and I would highly recommend it. This is a very brief description of the protocol, and if you are still confused, give me a call so I can better explain it in detail.
Attendant Console
Q: Our Attendant Console is not able to take any more than one call at a time. It used to be that we could put one call on hold, and the next call waiting would come in on the next loop button. Is there some option that has changed? We can't find any that seem to make a difference.
A: We have just recently been exposed to that very problem. It was the 48-volt power to the console that did the trick for us. Originally, the console was powered from the switchroom via the white-brown pair of the station wire. We disconnected that, and installed the optional 48-volt transformer with a 400-B adapter for the power source, and the problem went away.
Q: We experienced a power failure, and now our Attendant Console isn't able to forward calls for our president and sales manager as it did previously. We have looked at the software assignments for "Attendant 1" and it all looks the same as the printout we have of it when it was installed. What else can we look for?
A: You probably won't find anything wrong with the software for "Attendant 1", or anywhere else. Power failures can cause strange problems, some which you will never recover from without replacing costly hardware. In your case, however, I would bet you will recover by simply powering-down your system, waiting a minute or so, and powering it back up. Don't forget to shut down your Definity Audix first, if you have one.
Auto Attendant
Q: How do those folks with an auto-attendant that says "to dial by name, press 1" do that with an Intuity?
A: First, create a second-level auto-attendant. On the second page of that auto-attendant, in the upper left corner, is a field asking which addressing option, either "extension number" or "names". Choose "names". Then go to the third page and type an "e" in each of the destination fields for "1" through "9" and make their transfer type "transfer", meaning to send this call back to the PBX. You will need to record the greeting for the new auto-attendant as something similar to, "Please enter the first three letters of the person's first name, followed by the # sign" (if the names are entered in the mailboxes as "John Doe"; otherwise if they are entered as "Doe, John" record the greeting to say....."the first three letters of the person's last name followed by the # sign"). You then send one of the choices of the main auto-attendant to that second-level auto-attendant's extension number by utilizing the "call-answer" rather than "transfer" method. The "call-answer" routing keeps the call in the Intuity rather than transferring it back to the PBX which would then have to transfer it back to the Intuity causing an undue delay in the whole process.
Q: Our Auto-Attendant has several layers to it, and we seem to have a long delay between selections. I have listened to other Auto-Attendants, and they don't seem to have this problem. What are we doing wrong, or what do we need to correct this?
A: Assuming your Auto-Attendant is on your voice mail that's in the Audix family, it seems that on the 3rd page of the Auto-Attendant the "Treatment" for your selections might be set to "T" or "Transfer" instead of "CA" or "Call-Answer". If you are using the "T" or "Transfer" option, you need to change it, as what is happening is that you are sending all choices back to the PBX for further transfer back to Audix each time. Changing the option to "CA" or Call-Answer" will send it directly to the next Auto-Attendant box with little or no delay.
Q: We re-record our Audix auto-attendant two or three times a year, and have a need for doing it at this time. The person who always did it is no longer with the company, and we have no idea where to begin. How do we find what is needed to re-record?
A: The recording of an auto-attendant is nothing more than recording a voice mailbox in the Audix; it is done exactly the same way. If you know the extension number of the auto-attendant, you have all that's necessary to do the job, assuming you know how to change the 3rd or "selection" page of the attendant. To some, this page is abundantly clear as to its function, but to others, due to a lack of familiarity, it may not be. If you have a problem you will need to either consult your documentation, which is not usually an easy feat, or consult your service company or myself at the addresses below. If you don't know the extension number, give the command "list attendant" at the Audix command line and a list of all the auto-attendants will be displayed. If the originator of this auto-attendant was doing the job, the name of it will be readily apparent. If not, and you have a number of auto-attendants from which you cannot distinguish, you will have to start at the PBX by looking for the extension number that sends it to the Audix, the method of which, fortuitously, is answered by the next two questions in this column.
Q: Our network provider is changing our service from analog trunks to a T-1, and we are adding DID numbers at the same time. They have asked us what numbers we currently have answered by our auto-attendant that we may need as additional DID numbers to avoid losing those calls. How do we find out all of the numbers we have answered by our Intuity auto-attendant?
A: Calls come into a Definity PBX in only one way: through a trunk group. By mentioning you are adding DID numbers at this time, I assume you have no analog DID trunks currently, so therefore most probably have only Central Office (C.O.) trunks at this time. If so, that makes things easy, as on the first page of each trunk group, on the right-hand side of the page near the top, are the "incoming destination" and "night destination" fields where the information you are looking for is contained. You will also need to look at the port assignment page (either the third, fourth or fifth page, depending on your release of software) to get the list of telephone numbers assigned to this trunk group (again, assuming the person who set this trunk group up did their job and listed them properly), and to see if the "night" field has an entry. If it does, this is another destination that may be pertinent to your network provider.
If the telephone numbers aren't on the port assignment page, you will need to have the ports traced to determine the numbers in that trunk group. What your network provider is most probably looking for are any numbers your callers use to contact you, known as LDNs (Listed Directory Number) that you may have either advertised or given to your callers in mailings, business cards, etc. To continue to get those calls, you will need to have them included in your DID string or list, as they are associated with one or more of the analog trunks as the trunk's telephone number. When those trunks are removed, that number will be lost if you don't have it changed to one of your DID numbers. You will then need to build some entity, most likely an "x-port" station covering to the destination previously found in the trunk-group "incoming-destination" field, or better yet, if you have vectoring, utilizing that. There are many variables here, so if I have missed answering your question, please contact me.
Least Cost Routing
Q: Our Least Cost Routing, or ARS as Lucent terms it, seems to have some bugs in it. We make calls that return fast-busy signals or recordings that utilize the same route patterns as calls that go out successfully. Are we omitting something?
A: It sounds as if you are making the proper comparisons to successful calls, so I'll assume you know how to utilize the ARS fairly well. The fast busy's could be coming from your system or the public network. If you are certain they are coming from the public network, and never have problems other than with the area code or prefix you are trying, your problem is most likely with the programming in your local or long-distance providers' systems. I have never experienced a properly programmed ARS having problems other than on the public network, and feel that there are no bugs in your system.
Q: We have just entered our new budget year and are looking to upgrade our System 75 R1V3. The budget isn't so great that we can do this under the proposal we received for the Definity G3V9. Is there any way to upgrade in a gradual manner and not have to jump right to the latest release? Our other sites have Northern Telecom and we are able to do that with those systems.
A: Bless your heart for asking that question.....that's what the secondary market is all about! In order to upgrade through the OEM you would probably have to jump to the current release, which is the V9, but through the secondary market there are values to be had for earlier releases of the Definity. I personally like either the V4 or V6. The V4 is an exceptional value, but has the downside that it is the last release to use the older TN786B processor. Although the V6 uses a newer processor, it is only going to be usable up through the V8, as the V9 has an altogether different processor. Since I'm betting the primary reason you wish to upgrade is driven by budget availability rather than additional feature needs, my suggestion would be to upgrade to the V4 because if you want to later go to the latest and greatest you will have to replace either the V4 or V6 processor anyway. You can expect to save well over 50% on the system, and as far as the software license goes, Lucent/Avaya hasn't defined that very well other than what one client has related to me: "If you want Lucent/Avaya support, it must be re-licensed." Especially on things like this, if you need more information, we'll be more than happy to explain the reasons why the secondary market is great.
Q: Our Definity G3sV3 maximum capacity has been reached. Our answer for additional capacity has only been to upgrade. I have talked to other company's system administrators who tell me they have systems earlier than mine with three times the stations we have. What is different about our system that requires an upgrade to get the larger capacity?
A: It's the "s" in G3sV3 that indicates that your Definity has a designation of "small". The good thing is that you have, I believe, a system that was installed before the licensing of port sizing came into effect, and thus can probably expand your system to the G3iV3 (full capacity) by simply adding one circuit pack. It's the memory that defines the size of a system, and with the addition of the memory circuit pack CPP1, your system will magically become a G3iV3. The CPP1 is a daughter board that attaches to the TN786-B Processor circuit pack. If you are not sure of any of this, you should contact a knowledgeable interconnect company, or it may be available from Lucent. Or, you may contact me at any time with your questions.
Q: We are a self-maintained facility that has its own technicians and systems analysts for our Definity G3iV4. We used to be able to perform the tasks of testing, busying-out and releasing, etc. Since we have discontinued our maintenance agreement with Lucent, we have lost those permissions. How do other self-maintained Definity users cope with this?
A: This has been a thorn in the side of many, many Definity end users for over a decade. We have heard through the years many stories of what it took to get those permissions back from Lucent. In the end, it's Lucent you will have to deal with, but we have known companies that paid a one-time fee for activation of MSP (Maintenance Service Permissions), and others who claimed they were charged monthly for the privilege of doing their own maintenance. There are also different levels of MSP - one for stations, one for trunks, and one for processor/common-control. I have witnessed end users with combinations of some or all of them, and have heard from some that they were not able to get all of them. My observation has been that the larger the company, and logically, the larger the technical staff, the more likely it is to be granted the processor/common-control MSP. This will have to be taken up with Lucent, however, and it seems the permissions and answers vary from region to region of the country, and is possibly dependent on size, or perhaps tenacity, of your company.
Q: I can no longer call forward my calls. Instead of confirmation tone, I get another dial tone part way through the process. What has gone wrong with my system?
A: Probably nothing. Usually this is caused by having "Console Permissions" added to your Class of Service. Check your COS and see if you have this service. If so, you will need to dial the call forwarding feature access code, then the extension number you want to forward, and then the number to which you want to forward. This feature allows an individual to forward or cancel forwarding for extensions other than just themselves. It is usually reserved for System Administrators.
Q: We have a phone that's exposed to the public on which we need to be able to call an outside pager, but don't want it to be used for any other calls outside of our system. Is this something our Definity can do?
A: Yes, that's a task the Definity can easily perform. Here is a little-used but nice way of restricting phone calls. First, you need to establish a COR that has "All-Toll" as its Calling-Party-Restriction. That will bring up a field about half-way down the left column called "Unrestricted Call List". The number entered here will need to be defined as the one allowed by the Toll-List. It's entered much the same as the ARS entries. This is something that will be obvious when you look at it, or will leave you as mystified as you were before, depending on your experience level with the Definity system. There are many steps needed to successfully accomplish the restriction, so if it isn't clear how it's done feel free to call me or one of our staff at the number listed below.
Q: Is there any information stored in the Definity-G3i to identify calls to our system in cases of security or maliciousness?
A: There are many things that can identify these types of calls. First, if they are ongoing, report them to your network provider and ask for assistance in identifying the callers. For decades, the phone company has been able to "trap" malicious calls if trunks were configured for it. Getting them configured may be the difficult part, but be persistent. You may not be provided the information, but if you report the problem to the local police they can obtain the information from the network provider and prosecute if you are willing to sign for it.
Other means internally is known as either SMDR (Station Message Detail Recording) on early versions of your system, or CDR (Call Detail Recording) on later ones. Both SMDR and CDR need to be set up in software before collection of calls begins. You will find these under "change system features" on the 2nd or 3rd page for SMDR or under "change system cdr" for CDR. You will also need to make sure your trunk groups all have SMDR or CDR optioned to "y".
To get all of this straight, you will need to consult your "Upgrade & Additions" manual (or give us a call at the number below). Then you will either need to connect your system to a serial printer, a PC using a communications application allowing a "capture" mode, or a PC utilizing a call-accounting application. Even with all or part of this, you will need ISDN-PRI trunks to identify where the call originates, and many of these perpetrators know how to block their identification. For short-term use, if you don't already use a call-accounting package, I would recommend using a PC and a communications program such as ProComm in the capture mode. It's quick, easy and least costly.
Q: We are opening small offices of 15 to 30 people in locations across the country. Many of them will be connected to our regional offices which all have Definity-G3's. For these smaller offices we are considering either Partner, Merlin Magic, or Norstar key systems. What will be needed to connect any of these systems to our Definity network?
A: Another big question with multiple answers probably too long for this column. For starters, virtually any modern phone system can connect to any other modern phone system, with differing ease. It might be that you need to dial a different code for each location, and then the extension number, or it could be as easy as dialing the extension number and letting the Uniform Dial Plan software route the call for you. Normally, staying within the same manufacturer’s product line will garner better support both during installation/implementation, and ongoing.
I would like to suggest a system you didn't mention, a System75 R1V3 or Definity-G3V4 (or lower) from the secondary market. With any of these systems you will be close to the same costs, but will also be able to take advantage of the unique networking capabilities, not to mention the important significance of like architecture, allowing for movement of your hardware from location to location as size, closures, relocations, etc. occur.
Q: We're an interconnect very familiar with the Definity system, but are having trouble making DCS/UDP work between two sites. We think we have programming in place for all that the manual shows, but still can't call from one to the other. Any quick suggestions?
A: There's nothing quick about implementing DCS/UDP, and to even attempt it shows you must have good familiarity with the Definity. And as you also know, there's no way I will be able to address every possible problem you could have, as there are dozens of entries required to implement them. However, the most common problem I have noticed is the need for the trunk groups and their trunk-access codes to be the same in both switches not being adhered to. I would bet this is your problem. Let me know, please.
Q: We have just upgraded our Definity-G3 from a V3 to a V8. Our vendor, an independent telephone company not associated with Lucent, says it can no longer support us because they cannot log in to the system except with our "cust" login, which has no maintenance permissions. Is there any way we can continue to have our vendor log in with maintenance permissions?
A: In an earlier edition of this column I hinted at this problem. Lucent has added security levels to the V6, V7 and V8 releases of software that are very good at keeping anyone but them out of the system at levels needed for system maintenance. The security method is a great thing if you want to make sure nobody but Lucent is able to get into your system. If, however, your interest is having others than Lucent maintain your system you have only one option - that is to pay Lucent to add maintenance permissions to your "cust" login. This can be quite costly, and, coincidentally or not, would probably make your maintenance costs rise beyond what Lucent would charge you if you just used them. Did your upgrade include anything you needed beyond the capabilities of your V3, or was it just too good a deal to pass up? If it were the latter, it's another reminder that there's no such thing as a free lunch.
Q: We are a small company that has no dedicated system administrator. Recently, due to some customer complaints, we found that two of our trunks had been disconnected for several months. We noticed the Definity identified these disconnected trunks, but didn't light our alarm lights on our attendant console. Is there any way of the system notifying us when this occurs?
A: Yes, there is a way to make trunking problems identify as "minor" alarms, thus lighting the minor alarm light on the attendant console. You would need to have your vendor enter into the "set options" screen so that trunking minor alarms are reported as such, instead of the default setting. This downgrades trunking off-board alarms to "warning" alarms, which alert nobody.
Q: We are about to add another Definity system, which we intend to network to our current one. It is a secondary market purchase, so we are attempting the software implementation ourselves. We are also going to have to change our dial-plan from 3-digit to 4-digit dialing in order to utilize DCS and UDP. Is there an easy way to accomplish this without hand-entering each extension number, and if not, are there any pitfalls we need to avoid?
A: You are attempting a courageous mission, but one that's easy to perform with some time, patience and with paying detailed attention to the following steps. First, there is no fast and easy way (other than letting us do it for you), but it's not difficult either. I would suggest to first print out all existing stations, data-modules, hunt-groups, terminating-extension-groups, and vector-directory-numbers (i.e., "display station 200 count 1000 print"), and for quick reference, a "list station print". You will lose the personal-abbreviated-dialing lists of the stations, and if you want to re-enter them for the 4-digit stations, you will need to print them out, too. Then, beginning with the first extension, give the command "list groups-of-extension (extension number)", and note what groups the extension belongs to, as you will have to put the new 4-digit extension back in that group (this is probably the most forgotten step, and causes more problems than all the rest combined). Then "duplicate station (station number)" and add the new 4-digit station number with a port assignment of "x" and the same name as the 3-digit station. Next do a "remove station (3-digit station number) and note the port number before pressing "enter" to remove the station. Then "change station (new 4-digit station number)" and add the port that was on the 3-digit station in place of the "x" you originally entered. It is at this point you must go to the groups the station belonged to and enter the 4-digit station, if any, and don't forget the personal-abbreviated-dialing lists if you are going to replace them. I would be happy to discuss this with you if it's too convoluted here; just call.
Q: We have a System75 that was removed from an office we closed some time ago, and now want to install it in another location. Our account rep has told us we will need to upgrade to a Definity-G3V8 because there is no North American Numbering Plan in our ARS. Is this the only thing we can do with our System75?
A: Not by my standards. I felt the NANP upgrades were about 50% useless at the time everyone was scrambling to get upgraded before the axe fell. First, remember ARS lost the usefulness for its intended purpose when long-distance network providers gained the PIC capabilities. It does have a good means of restricting calling, but there is another method that was intended for that very purpose, so ARS isn't really necessary for that. There was a time when it was normal procedure to dial a 9 for local and an 8 for long-distance. If you institute that criteria (only needed if you have long-distance provided over a different trunk group than the local trunk group), then you have no need of ARS whatsoever if you give the local trunk group a TAC of "9" and the long-distance trunk group a TAC of "8". Remove it. Get rid of it. Kill it. 'Nuf said? Now use your System75 for at least the next five or more years and spend your telecommunications budget on necessary things.
Q: We constantly hear about the TCP/IP capabilities of the Definity G3v8 and wonder what we are losing if we don't upgrade to it. What's your opinion?
A: I view it about the same as video teleconferencing. Did you install one? If you did, you're probably among the 70% who hardly ever use it, and maybe among the many who have disconnected it. If you didn't, have you missed it? Probably not. The attribute of TCP/IP is its functionality over your LAN. That means you can send your voice calls over the same wiring that your PC network uses. If you are a mega-corporation with a new installation of a campus somewhere, that's probably an option to consider. If you are a mid-to-large corporation, even if you are contemplating moving to a new location, do you know how much wiring you can install for the cost of an upgrade and the hardware necessary for your Definity to become LAN compatible? I think TCP/IP is going to be a thing for all end users to consider one day, but I don't think that day has arrived yet for the masses. It's another of those bleeding edge commodities at this time that's probably best left to those who have bulging budgets and a curiosity that dictates living on that edge. Someday, however, I bet even I will join 'em.
Hunt Group
Q: Our hunt group application is angering many of our field personnel who call into it, and I need a better solution. Currently we have each agent in our office assigned as the only member in his own hunt group so we can put the call in queue if he is active on a call. After several minutes we have the call transferred to another hunt group comprised of all ten of the agents. At times this may take a caller out of queue and place him at the tail end of another, which makes him unhappy. How can I keep the queue positioning and yet have the call escape to be answered by another agent if one is available?
A: This is a common problem with small call-centers. Because you are small doesn't mean you don't need sophisticated solutions. This is always a dilemma, as it usually equates to dollars. What you need is called Expert Agent Selection (EAS), utilizing Call Vectoring. This method of call-center processing is capable of look-ahead routing which determines who is available for the call and sends it based on criteria you can define, including the agent's ability to manage the call. It also allows the caller to escape to a voice-mail box or elsewhere if they don't want to remain in queue. EAS and Vectoring, along with an Announcement circuit pack is the answer to your problem, but application of it would take up this whole column, so I suggest you contact your vendor, or call us for further information about it. The downside of it is that Lucent's licensing can be very expensive.
Voice Mail
Q: How does one maintain two extension numbers but only one Audix MailBox? We are a police agency and have occasions to be called on more than one number, but would like the convenience of maintaining only one MailBox for multiple members of our force.
A: I hope I'm not going to show my ignorance here, but I really could only come up with one solution. That was to have one of the numbers forward to voice mail in normal fashion, and have the other cover to a remote coverage point (i.e., r1) that was sent outside the system by dialing 9 and then the DID number of the first extension. One can also try covering to a hunt group that's been remote-call-forwarded (by the attendant or user with a COS that has "operator permissions") to the first extension. Good question..... any readers have answers to this one?
Q: Is there any way we can forward voice mail messages between our Definity Audix system at our corporate office and our DuVoice system located on our Definity at another site?
A: Both of those systems support the AMIS protocol, which was designed specifically for what you are trying to do. AMIS (Audio Messaging Interchange Specification) basically places a call from one voice mail system to the other, accesses the proper mailbox, and the transferring system plays the message while the other system records it via a plain old telephone call. It seems to have a bad rap as an archaic method, but those systems we have implemented were very successful, especially in light of the nominal cost compared to digitally networking two Audix systems. AMIS is an underused and mostly unknown method of transferring messages between systems. Based on our luck with it, we highly recommend trying it.
Q: Our Audix Small 8.2 is serving us well, but we have been told by Lucent/Avaya that we will not be supported any longer. What is the secondary market doing for its support, if anything?
A: Happily, upon the notification you mention, the secondary market is awash with hardware and knowledge to support the Audix Small (or Large, for that matter) in any software release for years to come. I say "happily", for the hundreds of secondary market Distributors and Dealers are delighted it is being abandoned by Lucent/Avaya. That system is, as it has been for years, functioning quite well, and serving the needs of the end users needing only voice-mail and/or automated attendant. From here on out, the upgrading to more storage and ports will only get more economical, as many who were scared into upgrading to Intuity Audix have traded-in their old systems, and Lucent/Avaya has auctioned them to the secondary market distributors, and this has caused the pricing to drop significantly. It's a great place to stay for quite awhile if you don't need any other options than voice-mail or auto-attendant, or need to be on the "bleeding edge". All that is necessary is to scan the advertisements in Telecom Reseller to find a secondary market dealer who will, if not themselves, refer you to a dealer who can support you as well, if not better, than you currently enjoy (enjoy?......... whatever!).
Q: We have need for wireless telephones in our manufacturing plant. We have tried the Transtalk9000 series phones a few years back, and they had too great a failure rate and were noisy with interference on them. Are any of the cordless phones on the market compatible with the Definity?
A: Yes, just about all of the cordless phones, including the new 2.4Gigahertz models, are compatible with analog ports on the Definity. However, the new Lucent Transtalk 9031 model is less problematic than the older ones you tried and seems less susceptible to interference. In addition, there is the wireless module that emulates cellular capability in the Definity and allows wiring these antenna "sites" over just about any campus or industrial complex of any size.
Q: We have two Attendant (Operator) positions that share incoming call responsibilities. At times such as breaks there will only be one position in operation, and the other on "Position Busy". We have an 800-Number that comes in on a hunt group where both of the Attendants are members. However, when either position is on "Position Busy", it still receives incoming calls for the 800-Number. The other numbers in the trunk groups don't have this problem. What can we do to correct this?
A: There are two methods to correct this. The problem stems from the fact that the "Position Busy" button has no effect on anything but the console queue, and since you are using a hunt group, there is no way of knowing the position is indeed "Busy". You can either get rid of the hunt group and put its extension number in the "Listed-Directory-Number" screen or you can remove the attendants from being members of the hunt group, give the hunt group a coverage path that covers ALL calls in the coverage criteria, and then make point one of the coverage path "0" or "attd". This will then send these calls to the attendant queue which recognizes the "Position Busy" status of each attendant.
Q: We have calls going to a conference room phone for which we can't trace the origin. How do we find out what calls are directed to an extension?
A: The first thing to determine is if it is a DID number. If so, that's probably your problem. Make it a non-DID number and no outside calls should be able to reach it. The second place to look is to see if it's a member of some group that gets calls. You can do that by giving the command "list groups (extension no.)".
The next place is to see if it's listed as a destination of a trunk group as mentioned in the answer to the question above. It might also be a coverage point for some other station, which should show up in the "list groups" query. You can also "list coverage-path" to see if it shows up as a destination. And lastly, and the most difficult to find, is to see if it's a point of a "route-to" in a vector, which would require looking at all the vectors one-by-one. On the Far-Side, it's also possible it's the night-destination of a hunt-group, and your system is in the night mode when calls go to it. Some systems are left in the night mode all the time. If you can't find the problem with these hints, give me a call and tell me my answer in this column flunked, and I'll help you find it.
Q: Are there any alternatives in the secondary market for certified Definity training? We are looking for the possibility of on-site training for both system administration and end user training, or at least an alternative to traveling to Denver each time we want to train an administrator.
A: I'm not sure of all of the "certified" folks that are available, but there are many who have been trainers for Avaya/Lucent who are no longer employed there who on their own offer training of both types you are looking for. They are located in most major metropolitan areas of the United States, though are not usually of high profile and are therefore not easy to contact. I hereby extend an invitation for those folks who read this column that are trainers, or those who may know of a trainer or training company, to contact me and I will list their names in a future issue. There are also companies in the secondary market who offer training, as we do, but like us, only in our facility at this time. We hope to suitcase the class for major metropolitan seminars in the future, but that's no help to you now.
Q: As a technician for an Interconnect, I am frustrated with the endless finger pointing when either trying to repair or install a T-1. Arguing is nonexistent without test results from a T-Bird or equivalent test device, whose price leaves them out of the question for small interconnects. Is there any way of testing within the Definity to prove where the circuit problem lies?
A: Yes, the Definity has a terrific method for positive identification, but unfortunately, it's buried inside the Tier-II's login, and not below that. It allows for a set-up, watches the call and reports where and what happened to it. Most of us have no access to that login, so the next best method, and one which has never failed me in testing hundreds of T-1's, is setting the "Synchronization" screen's primary source to the T-1 you are testing. Then, in succession, give the commands "disable sync", "set sync <DS1 Board location>, and finally "enable sync". Once the last command is completed, do a "status sync". If the T-1 has connectivity, it will show connected to the DS1 board, otherwise it will show connected to the Tone-Board. If you show Tone-board synchronization, check all the physical connections onsite, i.e., the CSU connections, PBX cable connection, etc. If all of your connections look good, replace the DS1 or the cables or the CSU or all of them, and test again. If you have the same results, your problem will undoubtedly be with the network provider. This has never failed me. You may think having all of this spare equipment is extreme, but if you don't have it with you, you have no business attempting the job, and it's more cost efficient than purchasing a T-Bird. Remember, this is for the T-1 only, so if you are having trouble with the trunking using the T-1 facilities, that will be another problem entirely assuming you are synchronized with the DS1 board.
Q: We have three sites with Definitys, all connected with T-1's. These sites also have our WAN connected via their own T-1's. I have heard that we could combine these facilities to reduce the number of T-1's to accomplish this. How is this done?
A: In the United States, a T-1 is made up of 24-channels, each 64K in bandwidth. Your question didn't specify how much bandwidth your WAN required. If you can do with 64K bandwidth between each site, the easiest way to do this is to remove one of the voice channels from the voice T-1, and add a tie-trunk group with that channel as the only member, and its destination a data extension whose port is connected to a 7400C data module. The Definity PBX is a very, very good data switch. If, however, you need more bandwidth than 64K, you will instead need a multiplexer that will split out the channels you specify to an RJ48 interface for the Definity, and a V.35 interface for your WAN. The cost for these units is approximately $2000.00.
Q: We have just been informed by our long-distance provider that we are getting a large number of calls from a foreign country, some with very long call durations. They have told us to check to see if we are being used as a switching center for these calls, sending them out of our system to various points in this country. Where do we begin?
A: There are many ways these clever people have discovered to gain entry to PBX systems and back out again to make long-distance calls at your expense. The most obvious protections deny the transfer of calls that come into a system to go back out without at least human intervention. Convenient means have been programmed into your system to allow you to be able to transfer yourselves through, but it leaves you vulnerable to thieves, just like leaving valuables in your unlocked car. You need to make sure you lock your system, and don't leave any keys where hackers can find them easily.
First of all determine whether your system needs any automated capability to transfer incoming calls back out. If not, eliminate trunk-to-trunk transfer in the system parameters features screen. If you do need some activity in and back out, utilize COS and COR restricted trunk groups accessed via Remote Access that is controlled with restrictive COR's and long "barrier codes", and utilized "account codes". Also, much of the toll fraud is accomplished via Voice Mail systems. In the AUDIX make sure you have "enhanced call transfer" selected or set to "yes". Most of all, have your system reviewed by a consultant who specializes in toll fraud. Lucent offers a great toll-fraud protection program, though I'm not sure if it's only for their maintenance clients.
Q: What is the function key setup for administering a Definity remotely with a PC?
A: The easiest method is to answer "4410" at the "terminal type" line where it shows a default of "513". This is where you login to the system and it asks you for the terminal type. This method will assign your function keys for you upon each command. If you want to use the "513' terminal type, the function setup is as follows: (I forget some, but the essential ones are these, all case sensitive) F-1 = escape+Ow for "Cancel", F-3 = escape+SB for "Enter", F-5 = escape+Om for "Help", F-7 = escape+[U for "Next Page" and F-8 = escape+[V for "Previous Page" (the O's are capital o's, not zero, and the [ is the left bracket). The "escape" is different for different programs, but many, such as Procomm Plus, use ^[ (upper-case 6 and a left-bracket) for the escape sequence. For example, this would make "Next Page" program as ^[[U. This, too, can be confusing, so a call to us might help clear any cobwebs I may have spun.
The very best program we have found for administering either a Definity or an Audix system is "Definity Site Administration" available from Lucent/Avaya. DSA, as it is known, has the function keys assigned for you, whether you use "4410" or "513" emulation. Until this program came out, we advised against using anything but Crosstalk Mark-IV(out of production) or Procomm Plus. Terranova was a confusing mess, though I understand recent versions of it work OK.
Q: We have been proposed an upgrade that will enable us to use TCP/IP functionality with our Definity. This has been defined to us as the best option for distribution for our PBX, and touted as a necessity. What exactly is the advantage for us?
A: It's hard to define, not knowing your size and condition of flux. If you are a large (several hundred or a few thousand port system that is installing in a location where new wiring is necessary, there may be an advantage to it. If you are located where you have all the wiring in place for your system, it doesn't make much sense to me to upgrade just to have TCP/IP. My understanding is that you can use the TCP/IP functionality to send your voice communications over your data wiring in prioritized packets eliminating the need for double wiring; i.e., a run for voice and a run for data to each workstation. If my information is correct , TCP/IP only works well on a LAN. Assuming it will work well some day over a WAN, then there may be good reason then to upgrade to it to reduce the cost of networking both voice and data systems separately, assuming the prioritized packets don't slow your data network to unusable functionality. Based on the information above, my recommendation would be to wait for functionality over a WAN before considering it necessary, unless it will save cost-justifiable dollars in additional wiring to the LAN. If any of our readers can add information supporting or contrary to the above, I invite them to contact me, and I will run the information in the next issue.
Q: We received the following question from Dawn Paolino:
We have reached the capacity of bridged appearances on our 8434DX phones, or so we have been told. We understand there is a maximum of 24 lines (main number and rollover) including 1 CAM. We now have to add two phones to an assistant’s desk to handle coverage. Does Lucent have another product available that will allow the assistant to know who is being called so they can answer, "John Doe's office; how can I help you?"
A: Yes, there is another way of identifying covered calls, and I believe it works much better. However, it does require the use of display phones at the covering location. I am a strong anti-bridged-appearance implementer. If there is any way of doing things without the use of bridged-appearances, I believe it to be a better way. I never use them! I'm as passionate about that belief as a vote counter in Florida. If you use a voice terminal at the covering location that has a display, the covering reason is given on the display in easy-to-understand code. First, it will say something like "Local to John Doe" or "Dawn Paolino to John Doe", and then in the far-right side, there will be a code of either a "b", "d" or "s". The "b" means the called voice terminal is busy on another call, and the call is temporarily bridged at that called voice terminal. The "d" means the call was not answered at the callers terminal in the prescribed number of rings, and that the call is temporarily bridged at that called voice terminal. The "s" means all calls at the called voice terminal are covering because their "send-calls" button is pushed. In as much as you are using 8434DX voice terminals, you already have the capability, and the information should already be appearing on your displays if only you would use call-coverage instead of those !@#$%^&* bridged appearances.
Q: From Arron Meyer, some information about TCP/IP was received:
I work for an Avaya Business Partner and wish to comment on your TCP/IP information in the November 2000 Definity Demystified column.
The TCP/IP functionality in the G3 is used mainly for integration with adjuncts at this time. When R9 becomes available, Avaya's 4600 series of IP telephones also becomes available for use. The reason that a switch upgrade to the G3 is usually installed by us, in my experience, is simple: the customer is adding a new adjunct that requires IP integration, such as CenterVu CMS or Intuity Audix. Why? Mainly because X.25 integration is being phased out. The 7400D data module will no longer be manufactured after December, effectively ending a key product for X.25 Processor Interface. Avaya will no longer support new installations of adjuncts using X.25 switch integration. It will continue to support customers who already have adjuncts integrated using X.25. Also, if the customer is a G3csi (Prologix), the only option for digitally integrating an adjunct such as Intuity Audix is TCP/IP using a TN799/TN799B/TN799C Control-LAN board. The only other option is Mode-Code integration as the G3csi platform does not provide for a Processor Interface circuit pack. For IP phones or IP trunks, at MedPro (Media Processing) board will also be required.
A: Thanks, Arron for your response, but from it I see further reason that TCP/IP's time has not yet come. Definity R9 is now available, as are its 4600 series of voice terminals, but they still only allow TCP/IP connectivity over a client's LAN, so the only advantage I can see is the elimination of dual runs of station wiring, which if already in place (which was the case in the question in the column) makes it a moot point. And the view of most of us who appear in this forum is that the OEM isn't the only place a client can find the products they need, as the secondary market is awash with perfectly good equipment with the same or better warranties and guarantees. The idea that products such as the 7400A, B, C or D data modules are effectively ended because of the OEM's decision to quit manufacturing them just doesn't hold water. And I love the fact that Avaya will no longer support new installations of adjuncts using X.25 integration, as there are hundreds of us out here who can and will. Avaya's a good choice, but not the only one.
As for the Prologix, it's a product whose time never did come. For less than the cost of a new Prologix, a client can obtain a secondary market G3s with all the same or better coverage. And Mode-Code integration works flawlessly, as many third-party manufacturers of voice-mail products will agree. On the issue of IP trunks, or MedPro boards, you are ahead of me on those, and I will have to defer to you on them, though I think perhaps its relevance to most clients will be negligible as a reason to upgrade to a software version just to include TCP/IP. Thanks for the information Arron, and let's keep the dialogue going to better inform Definity Users and Administrators, as I found your information very educational.

Walt Medak is president of Medak & Associates, Inc. He can be contacted by phone at 800-452-6477 X5001 or by email at .

Reprinted with permission from Telecom Reseller magazine

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