Voice Processing Features

Transcription

1 Voice Processing Features CONTENTS PAGE 1. Introduction A. Voice Processing Applications B. Time Slot Groups C. Music-On-Hold Parameters D. Communication Between the System and the Voice Processing Unit Voice Processing Unit Networking A. Network Operation Overview B. AMIS Networking C. Dial-Up Networking D. Networking E. TCP/IP Networking Automated Attendant A. Automated Attendant Applications B. Automated Attendant Recall Destination C. Automated Attendant Custom Audiotex Recordings D. Call Screening E. Extension ID Call Routing Announcement A. Call Routing Custom Audiotex Recordings B. Call Screening C. Digit Translation D. Digit Translation Nodes Multilingual Capability Voice Processing Fax-On-Demand A. How Fax-On-Demand Works B. Fax-On-Demand Setup Record-A-Call Scheduled Time-Based Application Router (STAR) Voice Mail A. Voice Mail And Message Notification/Retrieval Applications B. Mailboxes C. Group Lists D. Message Notification to Stations E. Cascading Remote Message Notification F. Subscriber Use of Voice Mail G. Non-Subscriber Use of Voice Mail H. Voice Mail System Administrator Features VOICE PROCESSING Page 4-1

2 CONTENTS PAGE 10. Directories A. Locating a Name B. Accepting a Name C. Requesting Additional Information D. Using the Directories SMDR Information Storage and Retrieval Unified Messaging A. Level B. Levels 2 and Automatic Fax Detection A. Unified Messaging Required for Fax B. Fax Card Required C. Programming Call Routing Announcements and Mailboxes D. Fax Message Format Page 4-2

3 1. INTRODUCTION NOTE: As of version 8.0, the OS/2 Voice Processing Unit is no longer supported. A. VOICE PROCESSING APPLICATIONS 1.1 Voice Processing can be used for any of the following applications. Each application is described in detail in this chapter. Automated Attendant: The automated attendant is a programmable feature that can be used to provide automated call answering service. Calls can transfer, forward, or directly ring in to an automated attendant. When an automated attendant answers a call, it plays a recording that gives dialing instructions. After hearing the recording (or at any time while it is playing), the caller may then dial an extension or mailbox number. Automated Attendant Recall Destination: If a call that is transferred by the automated attendant is not answered before the Transfer Voice Processor timer expires, the call recalls the Automated Attendant Recall Destination. The Recall Destination announces that the station is unavailable and allows the caller to leave a message (if the station has an associated mailbox) or dial another extension. Call Routing Announcement: Call Routing Announcements can be used two ways: A Call Routing Announcement application can be used in place of a playback device. The playback device function is especially useful for programming hunt group announcement and overflow stations. When called, the Call Routing Announcement application will play a recording and then hang up. The Call Routing Announcement application can use Digit Translation which allows the caller to press a single digit for access to a mailbox, a Fax-On-Demand function, or a station or hunt group that has an associated mailbox or extension ID. Digit translation can be programmed for each digit 0-9, #, and *, plus a Timeout that is used when the caller does not enter a digit. Each digit can lead to a digit translation node that has its own digit translation values. This layered Call Routing Announcement digit translation creates a tree of programmable digit translation nodes. 4 VOICE PROCESSING NOTE: The phone system now supports a Primary and Secondary Language which can each be programmed as American English, British English, Japanese, or Spanish. Digit Translations, therefore, can be programmed to use either language. Message Notification/Retrieval: Only one Message Notification/Retrieval application may be created and it must be created to allow Voice Mail message notification and quick mailbox access. Introduction Page 4-3

4 Record-A-Call: This feature allows a station user to record an ongoing call in a mailbox message. When a station user enters the Record-A-Call feature code, the system places a call to the station s assigned Record-A-Call application. When the application answers, the system sets up a conference call with the station s Record-A-Call mailbox. At default, the mailbox plays a greeting to indicate that the recording is in progress. STAR: The Scheduled Time-Based Application Router (STAR) enhances the programmability of the Voice Mail application greetings. With STAR, applications can be programmed to play alternative greetings for holidays and weekends. A STAR application is a table of up to 20 entries that serves as a routing table, which tells Voice Processing which application will be used, based on day and time information in the table. (The applications are programmed to play the greetings, not the STAR application. The STAR routes the call to the right application.) Voice Mail: This application handles all calls that are directed to Voice Mail (other than through the Message Notification/Retrieval application) by subscribers and non-subscribers. Callers will hear the main company greeting, followed by a menu of available options. Stations can forward or transfer calls directly to their mailbox using this application. Directory Services: Directory services provide callers with a list of mailboxes and extension IDs. Station Message Detail Recording (SMDR) Information Storage: SMDR information can be stored on the Voice Processing Unit s hard disk and then processed using call record sorting software, such as Inter-Tel Inside Track. 1.2 The applications are given extension numbers when they are created. Voice Processing application extension numbers are at default, but may be reprogrammed. B. TIME SLOT GROUPS 1.3 Each application is assigned to a Time Slot Group which determines the maximum number of voice channels that will be used by the applications in that group. Voice channels are used for processing calls between the phone system and the Voice Processing applications. 1.4 The programmed voice channel limit for the time slot group may exceed the number of voice channels actually provided by the hardware, due to hardware limitations and/or heavy system traffic. For example, if an Automated Attendant application is assigned to a time slot group that has a programmed limit of five voice channels, it can normally support five simultaneous transfers of outside calls to extensions. However, if only four voice channels are available (because all other channels are in use by other applications or the hardware only supports four channels), a fifth call cannot be completed to that (or any other) application. When all voice channels are busy, intercom callers will hear reorder tone and see a CALL CANNOT BE COMPLETED display; outside callers will hear ringing but their calls will not be answered. Page 4-4 Time Slot Groups

5 1.5 Because multiple voice channels can be made available to the applications, callers are unlikely to encounter busy resources. In addition, the use of applications as recall destinations and attendants make it even less likely that a caller will not reach an available resource. For example, an Automated Attendant application can have one Automated Attendant Recall Destination with several voice channels as its recall destination and another Automated Attendant Recall Destination application (with several voice channels) as its attendant. Recalls from that Automated Attendant application would be very likely to reach an available voice channel at one of the Recall Destination applications. AUTOMATED ATTENDANT Recalls Attendant Recalls C. MUSIC-ON-HOLD PARAMETERS 3.1 In previous system versions, the system determined the music source a caller heard when the call was waiting based on the trunk group on which the call resided. As of version 7.0, the system can be programmed to determine the music source a caller hears based on the device for which the caller is waiting. By default, the system determines the music source based on the trunk group on which the call resides. 3.2 The following fields are programmable for each voice processor application: First Automated Attendant Recall Destination application Second Automated Attendant Recall Destination application Audio for Calls Camped onto this Device: Defines the audio that a caller who is camped-on to the device hears. Audio for Calls Holding for this Device: Defines the audio that a caller who is holding for the device hears. Audio for Calls Ringing this Device: Defines the audio that a caller who is ringing the device hears. The default is Ringback. 4 VOICE PROCESSING NOTE: If the trunk group s audio field, including Music-On-Hold, is set to Use Next Device s Audio Source, the system uses the station's programming. If the field is set to any other option, such as Board, the system uses the trunk group s audio source, overriding station programming. 1.6 When a caller on a node with the new V7.0 or later functionality waits for a device on a node without this functionality, the caller hears the music source programmed for the node trunk group on the destination node. This is identical to the previous functionality. 1.7 When a caller on a node without the new V7.0 or later functionality waits for a device on a node with this functionality, the caller hears the music source programmed for the device for which the caller waits. This is the new functionality. Time Slot Groups Page 4-5

6 D. COMMUNICATION BETWEEN THE SYSTEM AND THE VOICE PROCESSING UNIT 1.8 The software and hardware that control the Voice Processing applications is located in the Voice Processing PC. The phone system must be able to communicate with the Voice Processing PC for the applications to function. If the phone system is unable to communicate with the Voice Processing PC, the applications cannot be used and the following will occur: Direct ring-in calls to an application will be sent to the primary attendant (if there is one). Intercom calls to applications will receive reorder tone. Display phones show <Application name> IS UNPLUGGED. Administrator phone users will see an Alarm 203 (Voice Processing: Communications Link Down) if the system detects a loss of communications between the phone system and the Voice Processing PC RS232 port. Calls will not recall to an application. Instead a recalling call will remain at the station or hunt group and ring until it is answered or the Abandoned Call timer expires. If an application is used as a hunt group overflow or announcement station, calls will not be sent to the application, but will remain camped on to the hunt group. A transfer to an application will camp on until the Recall timer expires. Then it will follow the usual recall path. Calls cannot be forwarded or system forwarded to an application. SMDR information will not be recorded in the buffer on the Voice Processing PC hard drive. Database programming for the Voice Processing features is not allowed. 1.9 Communication between the system and the Voice Processing Unit can be via RS232 connection to the equipment cabinet or TCP/IP socket connection to the system CPC or CP Server. For an ATM-based system, the Voice Processing Unit should be placed close to the first cabinet that provides the first eight voice mail ports. The first or second cabinet must also host the Options Card (OPC) which is used for the remaining 32 voice mail ports. Page 4-6 Communication Between the System and the Voice Processing Unit

7 2. VOICE PROCESSING UNIT NETWORKING NOTE: AMIS and other networking methods require specific premium features. Make sure the software license supports the required premium features. 2.1 The Voice Processing Unit network is programmed separately from the phone system network. There can be up to 100 nodes in the Voice Processing Unit network. There are new programming fields that provide networking support: Node: The Network Node table contains information about each Voice Mail node in the network. This table contains information that allows the system to identify nodes, contact nodes, and deliver messages to nodes. Each node has its own internal message queue (similar to a mailbox s message queues) and will store messages destined for other nodes until they are delivered. Network Mailbox: A network mailbox is a mailbox on the local Voice Processing Unit that accesses a mailbox on a remote node or the node itself. A Network Mailbox is very similar to an Extension ID. If you log on to a Network Mailbox, the only options available are the directory name, the password, and the transfer method. There are two types of network mailboxes, a direct network mailbox and a node network mailbox. A direct network mailbox is a network mailbox that accesses a mailbox on a remote node. To do this, the network mailbox is programmed to use a specific mailbox number. A node network mailbox does not refer to a specific mailbox on a remote node, but instead, refers to a remote node in general. To do this, the mailbox is not programmed with a specific mailbox number. This tells Voice Processing to send the call to the remote node programmed. The directory name for this type of mailbox should be something that indicates the location of the node rather than an individual s name. For example, Denver or Main Office. 4 VOICE PROCESSING 2.2 Voice Processing has several different methods for networking Voice Mail systems together. Each node is programmed with a Network Type field that identifies how the local node communicates with the remote node. This field can be programmed to the following: IC or CO AMIS: Voice Processing sends messages via public network telephone lines or a private network using AMIS protocol. Voice Mail messages are sent by placing a call through the phone system to the destination Voice Processing Unit s phone system. An off-node Call Routing Announcement station, CO trunk group, or hunt group at the destination is programmed to receive the AMIS signals. Dial-Up: The Voice Processing Unit sends messages via modems using TCP/IP protocol over telephone lines. The originating and destination Voice Processing PCs must have the Windows NT Remote Access Server (RAS) service and a TCP/IP connection. Voice Mail messages are sent during a temporary TCP/IP session with RAS between the originating Voice Processing Unit s modem and the destination Voice Processing Unit s modem. The telephone lines may be public network trunks or tie lines. The lines should be programmed to ring in directly to the modems so that an extension number is not required. The Voice Processing Unit sends messages using an protocol. The originating and destination Voice Processing Units must have an client that provides a connection to the server. Voice Mail messages are converted into messages, sent over the IP network, and then converted back to Voice Mail messages at the destination Voice Processing Unit. Voice Processing Unit Networking Page 4-7

8 TCP/IP: Voice Processing Unit sends messages using TCP/IP protocol. The originating and destination Voice Processing Units must have a TCP/IP service that provides a gateway to the IP network. Voice Mail messages are sent during a temporary TCP/IP session between the originating Voice Processing Unit and the destination Voice Processing Unit. A. NETWORK OPERATION OVERVIEW 2.3 The Network Node table is used to define information about each node in the Voice Mail network. Before the local node can send messages to a remote node, there must be an entry in the Network Node table for the remote node. 2.4 To leave messages for mailboxes on remote nodes, you must have network mailboxes on the local node. If the local node only has network mailboxes associated with specific mailbox numbers, the local node can only leave messages for those mailboxes. If the local node has a network mailbox for a remote node (no mailbox number specified), the local node can leave messages for any mailbox on the remote node (as long as the Validate Network Mailboxes flag is disabled). Leaving Messages 2.5 Users can leave messages for any network mailbox as either a subscriber or a non-subscriber. Leaving messages for a direct network mailbox is just like leaving messages for a regular mailbox. After entering the network mailbox number, Voice Mail will play the mailbox directory name (if it is recorded, otherwise, the mailbox number), followed by the recording instructions, and the caller can leave a message as usual. 2.6 Users can also leave messages using a node network mailbox. Users who call the node network mailbox will hear a prompt asking them to enter the desired mailbox on the remote Voice Processing Unit. After that step, it is the same as leaving a message for a regular mailbox. Node network mailboxes are useful when you do not know the direct network mailbox number and/or when the mailbox on the remote node does not have a direct network mailbox on the local node. (If the Voice Processing Unit attempts to deliver the message from a local mailbox user to the remote node and the mailbox number is invalid, the message will be returned to the sender.) 2.7 If the Validate Network Mailboxes flag is enabled, users can only leave messages for mailboxes on remote nodes that have a direct network mailbox on the local node. If there are no direct network mailboxes that refer to mailboxes on the remote node, the caller will hear a prompt indicating that the message cannot be delivered to the remote node. 2.8 Unlike regular mailboxes, network mailboxes (both direct and node mailboxes) do not have their own message queues. When a message is delivered to a network mailbox, the Voice Processing Unit looks up the mailbox s node information, and stores the message locally. The messages are stored until they are transmitted to the remote node or returned to the sender. Page 4-8 Network Operation Overview

9 Undeliverable Messages 2.9 The Voice Processing Unit delivers messages to another node by calling the System Number of the remote node. If the remote number is busy or there is no answer, the Voice Processing Unit will continue attempting to contact the remote node until the remote node answers, or until it has reached the Maximum Network Call Attempts limit (see page 6-144). If the Voice Processing Unit has made consecutive, unsuccessful Maximum Network Call Attempts (due to busy or no answer), it will stop attempting to contact the remote node and return any messages pending for the node. The messages are then considered undeliverable. Undeliverable messages pending for a node are handled according to the Undeliverable Network Messages Destination field. The field can have a value of Delete, Sender, or System Administrator: When the field is set to Delete, all undeliverable messages are deleted. When the field is set to Sender, all undeliverable messages are returned to the sender s mailbox, if possible. If the sender is unknown, the messages are returned to the System Administrator s mailbox. If the System Administrator s mailbox does not exist, then the messages are deleted. When a subscriber listens to a returned message, he will hear a prompt indicating that the message was undeliverable. When the field is set to System Administrator, all undeliverable messages are returned to the System Administrator s mailbox, if it exists. Otherwise the messages are deleted. 4 VOICE PROCESSING Examples 2.10 To help understand networking configurations, refer to the diagram on the next page. In the diagram: There are three nodes in the network, Chandler, Denver, and Houston. Chandler and Houston both have Voice Processing Unit, and Denver has a non-inter-tel Voice Mail that supports AMIS. The Voice Mail network depicted in the diagram does not have a universal numbering plan. Delivery schedules are not shown Look at mailbox 1721 in Chandler. This is a direct network mailbox that refers to mailbox 1421 in Houston. When someone delivers a message to mailbox 1721 in Chandler, the message will be transmitted to Houston (via direct Dial-Up) and delivered to mailbox Note that if someone in Chandler delivers a message to mailbox 1421, it will be delivered to Chandler s local mailbox 1421 and not to Houston s mailbox The Chandler location also has two node network mailboxes, 2001 (for Denver) and 2002 (for Houston). These two mailboxes allow users in Chandler to deliver messages to any mailbox in Denver and Houston, respectively, if the Validate Network Mailboxes flag is disabled. (If the flag is enabled, users in Chandler can only deliver messages to direct mailboxes 200 and 212 in Denver, and 1421 in Houston.) 2.13 Houston can only send messages to two mailboxes in Denver (200 and 212) and one mailbox in Chandler (1200). This is because Houston only has three direct network mailboxes and no node network mailboxes. If Houston had node network mailboxes for Denver and Chandler, users in Houston could send messages to any mailbox in Denver or Chandler (if the Validate Network Mailboxes flag is disabled, as explained above) Details about the configuration in Denver are not listed because this node is assumed to have a non-inter-tel Voice Mail that supports AMIS networking. The only information Chandler and Houston need to know about Denver are the mailbox numbers (if they want to set up direct network mailboxes) and Denver s system number. Undeliverable Messages Page 4-9

10 FIGURE 4-1. Example of Voice Processing Unit Network CHANDLER : Local Mailbox Node # Remote Mailbox None None Network Node Table: Node Description System # Type 1 Denver AMIS 2 Houston Dial-Up DENVER System Number: (303) Local Mailboxes: Non-Inter-Tel Voice Mail System Using AMIS HOUSTON Local Mailbox Node # Remote Mailbox Network Node Table: Node Description System # Type 1 Denver AMIS 2 Chandler Dial-Up Page 4-10 Example of Voice Processing Unit Network

11 B. AMIS NETWORKING 2.15 Audio Messaging Interchange Specification (AMIS) is a vendor-independent protocol for networking Voice Mail systems. The Voice Processing Unit supports AMIS-Analog which uses DTMF control signaling and analog message transmission Voice Processing uses Call Routing Announcement applications (local or off-node) to detect and process incoming AMIS calls. The Call Routing Application s greeting must consist of only a 15-second pause (one long and one intermediate pause), because AMIS will disconnect if it detects voice AMIS-Analog uses DTMF signaling to exchange control information between the originating system and destination system. Voice Mail messages are transmitted in analog form over the public telephone network. This means that the originating system uses its standard playback function to transmit, and the destination system uses its standard record function to receive. Because messages are transmitted in this manner, message transmission time is exactly the length of the message plus synchronization and control signaling overhead (which is typically less than one minute) AMIS-Analog specifies that the destination system terminates the call if it detects a period of silence of ten seconds or more during message transmission. Therefore, messages being transmitted should not contain periods of silence that are longer than ten seconds. They should also not contain any DTMF tones or call progress tones that may cause the destination system to disconnect or stop recording prematurely. 4 VOICE PROCESSING AMIS Call Processing 2.19 An AMIS call is initiated by the originating system calling the destination system over the public telephone network. When the call is connected, the originating system and destination system perform a handshake to establish the AMIS session The originating system then identifies itself with its system number and AMIS protocol version. After acknowledgment by the destination system, the originating system specifies the type of message to be delivered, the length of the message (if available), the originator s mailbox number, and the recipient s mailbox number. The originating system then begins transmitting the previously recorded message by using its standard playback function. The destination system records the message and delivers it the recipient s mailbox Outgoing AMIS calls are made by the Message Notification/Retrieval application. When a network mailbox that is set up for AMIS networking receives a message, it triggers the Voice Processing Unit to make an outgoing AMIS call to deliver the message The Voice Processing Unit begins by calling the System Number of the remote node. If the remote number is busy or there is no answer, the Voice Processing Unit will continue attempting to contact the remote node until the remote node answers, or until it has made the Maximum Network Call Attempts. If the Voice Processing Unit has made consecutive, unsuccessful Maximum Network Call Attempts (due to busy or no answer), it will stop attempting to contact the remote node and return any messages pending for the node If the remote node answers the call, the Voice Processing Unit will establish the AMIS session with the remote node. After a successful handshake, the Voice Processing Unit will start the AMIS session. (If this is unsuccessful, it will disconnect and try again.) If the destination system successfully acknowledges the AMIS session, the message transmission process starts. AMIS Networking Page 4-11

12 2.24 The Voice Processing Unit begins the message transfer process by transmitting the message type (new or reply), message length (if available), the originating mailbox, and the destination mailbox. If the remote system responds with an unsuccessful acknowledgment, the Voice Processing Unit will return the message. If the Voice Processing Unit has not reached the per-call message limit (nine messages per standard AMIS-Analog), and there are more messages, it will continue with the next message. Otherwise, it will end the AMIS session and disconnect. Incoming Calls 2.25 Call Routing Announcement applications are the only applications that can receive and process incoming AMIS calls. If any other application answers an incoming AMIS call, the application will eventually time out. If a Call Routing Announcement application answers an incoming AMIS call, the application s Accept Network Calls flag must be enabled for it to process the AMIS call. If it is not enabled, the call will time out After the Voice Processing Unit sets up the AMIS call, it performs the following tests to see if it can deliver the message: Checks to see if there is enough disk space to receive the message. Checks to see if the destination mailbox exists. Checks to see if the destination mailbox is full If any of the tests fail, the Voice Processing Unit will send the appropriate response to the originating system and then wait for the next message. If all the tests pass, it will begin recording the message. The Voice Processing Unit will continue recording the message until it detects DTMF tones or at least ten seconds of silence, or until the recording exceeds the maximum message length. If it detects ten seconds of silence or if the recording exceeds the time limit, it will disconnect. If it detects DTMF tones during the recording, and the DTMF tones are not the End Message signal, then it will ignore the tones and will continue recording When the Voice Processing Unit receives the End Message frame, it delivers the message to the destination mailbox and waits for the next frame. If the next frame is the End Session frame, it will disconnect. If the next frame is another Message Information frame, it will verify that it has not received the maximum number of messages during this call. If it has, it will send an error response and disconnect. If it has not, it will begin receiving the next message. AMIS-Analog Protocol Limitations 2.29 These limits will be enforced when the Voice Processing Unit places calls to or receives calls from another Voice Processing Unit or a non-inter-tel Voice Mail system that supports AMIS-Analog. Maximum of nine messages per call. Maximum of one recipient per call. If the same message is to be delivered to multiple mailboxes, the message must actually be transmitted multiple times. Maximum message length is eight minutes. Certified delivery of messages is not supported. Priority message indication is not supported. Private message indication is not supported. Full duplex message flow is not supported. After the originating system has transferred its messages to the destination system, if the destination system has pending messages for the originating system, it cannot transmit them during the same call. The destination system must make a separate call to transfer its pending messages. In other words, only the system that originates the call can transfer messages during that call. Page 4-12 Incoming Calls

13 The retransmission of data frames in the event of an interdigit time out is not supported. Software Configuration 2.30 The following list describes the programming requirements necessary for AMIS networking. System\Premium Features: The Voice Processing Unit Networking AMIS premium feature is required. System\Devices and Feature Codes\Stations: There must be at least one off-node device for a station on the remote node programmed for AMIS networking. This offnode device is required before creating the network mailbox below. Voice Processor\Devices\Applications: There must be a Message Notification/Retrieval Application programmed with outgoing access. There must be a Call Routing Announcement application that has the Accept AMIS Calls flag enabled. The Call Routing Application s greeting must consist of only a 15-second pause, because AMIS will disconnect if it detects voice. Voice Processor\Devices\Nodes: The node table must contain an entry that is programmed for AMIS networking. Voice Processor\Devices\Mailboxes: There must be at least one network mailbox on the local Voice Processing Unit that refers to a network node programmed for AMIS networking. For example, if entry 5 in the Network Node Table is programmed for AMIS networking, then at least one network mailbox must have node 5 in its Network Node Number field. Voice Processor\Networking\AMIS: The IC or CO System Number field must be programmed to allow the Voice Processing Unit to identify itself when establishing an AMIS session and to allow the remote AMIS system to be able to return calls to the local Voice Processing Unit. The Accept Unknown Network Calls flag must be enabled (or there must be a programmed node for each remote node from which the local Voice Processing Unit expects to receive AMIS calls). If the Accept Unknown Network Calls flag is enabled, the local Voice Processing Unit can accept calls from any Voice Mail system that supports AMIS networking. If this flag is disabled, the local Voice Processing Unit can only accept network calls from nodes that are programmed in the network node table. 4 VOICE PROCESSING Software Configuration Page 4-13

14 C. DIAL-UP NETWORKING 2.31 Dial-Up Networking uses TCP/IP protocol via modem for sending messages. With a dial-up connection, one PC calls another using a modem, and the two PCs establish a network connection Messages are forwarded, one or more at a time, over telephone lines, using modems and the TCP/IP protocol. The Remote Access Service (RAS) and TCP/IP services are installed on the originating and destination Voice Processing PCs. Messages are forwarded by opening a temporary TCP/IP session with RAS. RAS uses the originating Voice Processing Unit s modem to connect to the destination Voice Processing Unit s modem and RAS. The destination system s RAS opens a temporary TCP/IP connection to the destination Voice Processing Unit. Once the connection is established, it behaves just like a direct physical network connection The telephone lines used for this type of Voice Mail networking may be direct connections from telco to the modems, or they can be lines from the Inter-Tel telephone system. It is best that the telephone line connected to the destination PC s modem be a direct connection or a DID. This will reduce the possibility of timing problems. Refer to the illustration below. Voice Processing Unit Voice Processing Unit TCP/IP Service RAS Modem Modem Telephone Lines RAS TCP/IP Service 2.34 To set up a Dial-Up network: 1. Set up the TCP/IP Service and RAS Service as described in INSTALLATION (page 3-185). 2. Make sure the Voice Processor Dialup/ Networking premium feature is enabled. 3. In the Voice Processor\Devices\Nodes directory, right-click to create your nodes. A warning will appear that states you must restart database programming. When you have created all of the nodes, select Yes and then OK to end the database programming session. 4. In the Session Manager, restart the database programming session. 5. In the Voice Processor\Devices\Nodes directory, you should see the newly created Voice Processing Node. Program the description. 6. Select Network Type and scroll to Dial-Up. 7. Select System Number next to the description you just programmed. This system number, for this method, will be the telephone number for the Remote Node Voice Processing Unit s modem. 8. Create off-node associated and/or non-associated mailboxes in Devices - Mailboxes - Remote Node. Page 4-14 Dial-Up Networking

15 D. NETWORKING 2.35 With Networking, messages are forwarded one at a time via an IP network using an protocol. An client (Microsoft Mail, Lotus Notes, Internet Mail, etc.) is installed on the Voice Processing Unit. Communication is performed by sending messages to an server; this server forwards the message to the destination server and from there to the destination Voice Processing Unit. See the illustration below. Voice Processing Unit Voice Processing Unit Client Server IP Network Connection Server Client To set up an Network: 1. Set up the Service as described in the INSTALLATION section (page 3-185). 2. Make sure the Voice Processor Dialup/ Networking premium feature is enabled. 3. In the Voice Processor\Devices\Nodes directory, right-click to create your nodes. A warning will appear that states you must restart database programming. When you have created all of the nodes, select Yes and then OK to end the programming session. 4. In the Session Manager, restart the database programming session. 5. In the Voice Processor\Devices\Nodes directory, you should see the newly created Voice Processing Node. Program the description. 6. Select Network Type and scroll to Select Voice Processor and then Gateway and set the appropriate System Value (MAPI, VIM, or SMTP/POP3). 8. Depending on what option is used, you need to program the following fields: MAPI: Needs the Username, Administrator Address, and the Gateway Password. VIM: Needs the Username, Administrator Address, and the Gateway Password. SMTP/POP3: Needs the Address, Real Name, Username, SMTP Server, POP Server, Administrator Address, and the Gateway Password. 9. Go to Voice Processor\Devices\Mailboxes, and program the Mailboxes that will use the Gateway feature with the following fields: FAX/ Forwarding: Select one of the options (Disabled, Forward Only, Forward and Copy, or Unified Messaging). Address for Voice Messages: Enter the appropriate address for the mailbox. FAX Delivery Address: If the mailbox owner desires a FAX Delivery address, enter this information. In most cases, this will be the same as their address. VOICE PROCESSING Networking Page 4-15

16 E. TCP/IP NETWORKING 2.37 Messages are forwarded, one or more at a time, using the TCP/IP protocol. The TCP/IP service is added to the originating and destination Voice Processing PCs. A temporary TCP/IP session from the originating system to the destination system is established, and messages are forwarded. See the illustration below. Voice Processing Unit Voice Processing Unit TCP/IP Service Internet Connection TCP/IP Service 2.38 To set up a TCP/IP Network: 1. Set up the TCP/IP Service as described in the INSTALLATION section (page 3-185). 2. Make sure the Voice Processing Dialup/ Networking premium feature is enabled. 3. In the Voice Processor\Devices\Nodes directory, right-click to create your nodes. A warning will appear that states you must restart database programming. When you have created all of the nodes, select Yes and then OK to end the database programming session. 4. In the Session Manager, restart the database programming session. 5. In the Voice Processor\Devices\Nodes directory, you should see the newly created Voice Processing Node. Program the description. 6. Select Network Type and scroll to TCPIP. 7. Select System Number next to the description you just programmed. This system number, for this method (TCP/IP), will be the IP number for the Remote Node Voice Processing Unit. Remember, this is TCP/IP Voice Mail networking, and both Voice Processing PCs are on a TCP/IP Network. If you don t know the IP number, you must get it. 8. Create off-node associated and/or non-associated mailboxes in Devices\Mailboxes\Remote Node. Page 4-16 TCP/IP Networking

17 3. AUTOMATED ATTENDANT 3.1 The automated attendant is a programmable feature that can be used to provide an automated call answering service. Calls can transfer, forward, or directly ring in to an automated attendant. A Voice Processing Unit is required for the Automated Attendant feature on the Inter-Tel system. Calls to the automated attendant application are processed as follows. CALL TO AUTOMATED ATTENDANT Automated Attendant answers and plays a greeting followed by a menu of options. Caller selects option Caller does not select an option 4 Call is sent to station, hunt group, voice mail, or operator destination. Caller uses the directory and can select the name of desired party. Call is sent to the Automated Attendant s designated dial-0 operator. VOICE PROCESSING Call is sent to station, hunt group, voice mail, or operator destination. 3.2 When an automated attendant answers a call, it plays a recording that gives dialing instructions. During or after the recording, the caller may then directly dial a station extension number, Voice Mail access number (if there is no associated mailbox), or hunt group pilot number. Or, the caller may use the directory to look up the desired extension. 3.3 In a network setting, a trunk on another node can ring in to a Voice Processing application. (See page 5-61 in FEATURES and page in PROGRAMMING for trunk group information.) 3.4 When the automated attendant answers a call, the caller will hear the company greeting, followed by instructions and the list of available options. The caller then has the following options: Dial a station extension number: If an extension number is dialed, the call is transferred to the selected station. If ringback tones are enabled, the caller hears ringing while the call is being transferred. If ringback is not enabled, the caller hears music (refer to PROGRAMMING, page 6-248). If the called station is forwarded, the call follows the programmed forward. Dial a hunt group number: When a hunt group number is dialed, the call is transferred to the selected hunt group. The call rings or circulates according to how the hunt group is programmed (linear or distributed). If ringback tones are enabled, the caller hears ringing while the call is being transferred. Dial the Voice Mail application s extension number: The caller can reach the Voice Mail main greeting by dialing the application s extension number (access number) assigned to the Voice Mail feature. The caller can then leave a message as a non-subscriber or access any of the Voice Mail subscriber features. (See page 4-34 for Voice Mail information.) Automated Attendant Page 4-17

18 Use the directory: If the caller does not know the extension or mailbox number of the desired party, he or she can spell the name using the keypad buttons and look up the number in the directory. (This option can be disabled in the database. Or, if there are no names recorded for the individual mailboxes or for the system s extension IDs, this option will not be provided.) Directory names can be sorted by first or last name. (Refer to page 4-72 for information about using the directory.) Dial the operator access destination: If the caller needs further assistance, dialing 0 will access the Voice Processing Unit s programmed operator destination. Or, if the caller is on a rotary telephone and cannot enter a digit, the call will be automatically transferred to the operator destination. (The operator access destination is programmed in the database. There can be separate destinations for day and night modes.) NOTE: Due to the natural characteristics of the trunk, the volume level of DTMF tones transmitted over the trunk may be substantially reduced before reaching the phone system and Voice Processing Unit. This natural degradation in tone volume may adversely affect the reliability of the Automated Attendant feature. Other factors which can affect automated attendant performance are trunk noise and the quality and strength of the DTMF tones generated by the off-premises phone itself. A. AUTOMATED ATTENDANT APPLICATIONS 3.5 There are a number of different uses for this feature. Below are two examples: Direct ring-in calls to a busy attendant could be forwarded to an automated attendant (using the Call Forward If Busy feature or system forwarding). Calls could ring in directly to an automated attendant application s extension number when the system is in day and/or night mode. B. AUTOMATED ATTENDANT RECALL DESTINATION 3.6 When a station receives a call that has been routed through the automated attendant, the call is handled as a transferred call, and the display shows TFR FROM (description). If the call is not answered before the Transfer Voice Processor timer expires, the call recalls the automated attendant s recall destination. The recall destination is usually the Automated Attendant Recall Destination application which announces that the called station is unavailable and allows the caller to choose to leave a message (if the station has an associated mailbox) or dial another extension number. CALL TO AUTOMATED ATTENDANT Automated Attendant answers and transfers the call to the destination selected by the caller. The party receiving the transfer does not answer before the Transfer Voice Processor timers expires. Call goes to the Automated Attendant Recall Destination where caller selects option. Call is sent to station s associated mailbox Call is sent to station, hunt group, voice mail, or operator destination. Page 4-18 Automated Attendant Applications

19 3.7 If the Recall Destination fails to answer a call, it is automatically sent to the recall destination s programmed attendant. If the call is not answered there, it is disconnected after the Abandoned Call timer expires. 3.8 If an invalid number is dialed, the caller is prompted to enter another number. If a caller fails to make an entry before the Inactivity Alarm timer expires, the caller is prompted again to make an entry. 3.9 The caller cannot access trunks or enter feature codes through the Automated Attendant application. Trunk access codes and feature codes are considered invalid numbers. C. AUTOMATED ATTENDANT CUSTOM AUDIOTEX RECORDINGS D. CALL SCREENING 3.10 Custom audiotex recordings are made using the Voice Mail System Administrator s mailbox as described on page Each recording is associated with a recording number and assigned to the application(s) in database programming or using the System Administrator s mailbox. (Refer to page 4-57 in this section and page 6-89 in PROGRAMMING.) 3.11 Calls transferred from the Automated Attendant or a Call Routing Announcement application can be screened, announced, or unannounced When a call is received by an automated attendant or Call Routing Announcement application, and the caller enters an extension number, the station s programmed Transfer Method determines how the call will be transferred. The Transfer Method flags can be programmed in the database. If allowed in mailbox programming, they can also be programmed by the mailbox user. The available Transfer Methods are as follows: Announce Only: The caller is asked to state his or her name. Then the call is transferred to the associated extension number. When the station user answers the transfer, the application plays the caller s name and completes the transfer. Screened: The caller is asked to state his or her name. Then the call is transferred to the associated extension number. When the station user answers the transfer, the application plays the caller s name. The station user has the options of replaying the name, sending the call to Voice Mail (if the extension has a mailbox), transferring the call to another extension, accepting the call, or rejecting the call. Unannounced: The call is transferred to the associated extension number after the Voice Processing Unit checks the station to determine its status (busy, available, ringing, etc.). This is the default method In a network setting, the Voice Processing Unit can provide call screening for a destination extension on another node. However, the node where the Voice Processing Unit is connected must have an off-node device programmed for the destination extension and access to the remote node Refer to page for instructions on receiving announced or screened calls. See page 4-51 for subscriber mailbox programming. 4 VOICE PROCESSING Automated Attendant Custom Audiotex Recordings Page 4-19

20 E. EXTENSION ID 3.15 Extension IDs are used in conjunction with transferring calls through the Automated Attendant or using the Extension Directory. The extension ID allows callers to be transferred to stations and applications that do not have mailboxes. It also allows the station or application to have a recorded name in the directory. This feature is set up and initialized using database programming and Voice Mail features If an Extension ID has been created in database programming for a station extension number or a Call Processing Card (CPC) modem, either the principal owner of the extension or the Voice Mail System Administrator must set up (initialize) the ID with a new password and record a name for use in the Extension Directory. At default, the extension ID password is the same as the extension number If the Extension ID has not been initialized, calls can still be transferred to the associated extension number. However, they cannot be accessed from the directory. If the Extension ID s user name has not been recorded, it cannot be heard when callers access the Extension Directory. The name must be recorded in order to fully initialize the Extension ID. See page 4-42 for more information on initializing Extension IDs. NOTE: To provide system security, all extension IDs should have a password. To make the passwords difficult to guess, they should not match the station s extension number or consist of one digit repeated several times. The default password should be changed the first time the user logs in Once a password has been set up and the name recorded, the extension owner may access Extension ID Options which allows the associated directory name and password to be modified If an Extension ID has not been created for a station, callers using the automated attendant cannot be transferred to that destination. Instead, these callers will receive a system recording notifying them of an invalid entry and are routed back to the automated attendant s main menu Individual extension IDs can be programmed as Private in database programming. A Private number is included in the directory and can be dialed if the caller knows the number, but the extension number information is not available to the caller; only the name is played Extension IDs can also be programmed as Unlisted. That means that the number will not be included in the directory but can be dialed if the caller knows the extension number Calls transferred from the Automated Attendant or a Call Routing Announcement application to stations with extension IDs can be screened, announced, or unannounced. Programming flags determine the methods used for transferring calls to stations with extensions IDs. See page 4-19 for details In a network setting, the Voice Processing Unit cannot create extension IDs for off-node stations included in wildcard ranges. Each associated station must have its own off-node device entry. (See Appendix A Networking for more information.) Page 4-20 Extension ID