Hold Call Waiting Hold Call Waiting is a compound feature combining hold and answer and provides a convenient way to hold an existing call and answer a waiting call through a single button press. Hold Music (Music on Hold) The IP Office system supports up to 4 sources of music on hold - one system source which may be external, internal (WAV) or tone, plus up to 3 additional internal sources. The internal sources are .WAV files saved either in volatile memory, or on the optional memory card in an IP500 and IP406 V2. The .WAV file must be 16bit PCM mono and sampled at 8Khz with a maximum duration of 30 seconds. Alternate sources for music on hold are selectable for use by Incoming Call Routes or Hunt Groups. On IP500 systems, each source can be up to 90 seconds long (30 seconds on IP406 V2 and IP412). External music on hold sources connect to the 3.5mm Audio socket on all IP Office control units. Park As an alternative to placing a call on hold, a call can be parked on the system to be picked by another user. The call park facility is available through the user's telephone, one-X Portal for IP Office, Phone Manager or SoftConsole. Calls are Parked against a ‘park slot number’ which can be announced over a paging system so the person the call is for can go to any phone and collect the call by dialling the park slot number. For convenience Phone Manager has 4 pre-defined park buttons. On digital phones with DSS/BLF keys it is possible to program Park keys that will indicate when there is a call in a particular park slot and allow calls to be parked or retrieved. There is a system configurable timeout that determines how long a call may remain parked before it is re- presented to the extension that originally parked the call. Automatic Callback Feature • When calling an extension that is busy, set the system to call you when the extension becomes free. This feature is also called "Ringback When Free". • When calling an extension that just rings, set the system to call you when the extension is next used. This feature is also called "Ringback When Next Used". Benefit • Carry on with other work and let the system initiate a call for you when the extension becomes available. Description Depending on the type of phone a user has, call back when free is accessed by dialing a short code while listening to internal busy tone, selecting an option from an interactive menu or pressing a programmed DSS/ BLF key. Callback when free can also be activated from Phone Manager. You can also set a callback when free or a callback when next used using a short code.
Tones IP Office generates the correct user tones for the geography. These tones are generated for all IP Office extension types, analog, digital and IP. Supported tones are: • Dial, both primary and secondary depending on geography • Busy • Unobtainable • Re-order • Conferencing tone depending on geography Caller ID Feature • Display of the caller’s number on incoming calls, where supplied by the service provider. • Sending of calling number on outgoing external calls. Benefit • Confirmation and recognition of who is calling. • Storage of Caller ID numbers for return calls. • Directory name matching to Caller ID numbers. • Screen-Popping customer records in compatible applications. Description Where supplied by the service provider, the IP Office can receive and use the callers Caller ID. The Caller ID is passed through to the answering phone or application and is included in any call log or history supported by the phone or application. If the Caller ID matches a number in the IP Office's Directory, the matching directory name is shown instead of the number. Where IP Office Phone Manager, or the TAPI service is used to link to database software on the users PC, it is possible to have an automatic query performed on the supplied Caller ID and have the caller’s record in front of the user before the call is answered. For outgoing calls the IP Office can insert a system wide Caller ID or set a flag to have Caller ID withheld. For users with a direct dial number routed to their extension, that direct dial number is also used as their Caller ID for outgoing calls. Alternatively short codes can be used to specify the Caller ID that should be sent with outgoing calls. Note that the sending and receiving of Caller ID is subject to the service provider supporting that service. The service provider may also restrict which numbers can be used for outgoing Caller ID. Hold A call may be placed on hold with optional Hold music. A held call cannot be forgotten as it is presented back to the extension after a timeout set by the system's administrator. See also Park . 128 Toggle Calls Toggle Calls cycles round each call that the user has On Hold to their extension locally within the system,
Trunk Programs Overview See Flowchart 7-2 for an overview of the ISDN trunk programming procedure. o If you are installing all standard trunk groups, skip Program *16 (the default is for all standard TGs). o If you are installing BRI ISDN, use Program *16 to assign BRI type and Program 16 to assign BRI CO lines into trunk groups. o If you are installing PRI, use Program *16 to assign PRI types. Do not use Program 16. Do not change settings in Program 16 for trunk groups marked for PRI. Primary Rate Interface (PRI) Programming 1.Run Program *16 to define the ISDN trunk group type as: non-ISDN, Primary Rate Interface (PRI), or Basic Rate Interface (BRI). 2.For BRI trunk groups, run Program 16 to assign the trunk groups. Then, skip to Step 7. 3.For PRI, run Program *65 to assign B-channels. 4.For PRI, run Programs *66-1, *66-3 to define trunk groups parameters and channels. (Program 16 will be assigned automatically.) 5.To assign Call-by-call features (PRI only), run Programs *66-2, *66-4 to assign Call-by- call trunk groups codes and network ID. 6.For PRI, run Program *67 to assign call direction for ISDN trunk groups and call type capabilities allowed for each trunk group. 7.For BRI and PRI, run Program *68 to define the parameters for Calling Number to be presented for a call depending upon whether the ISDN trunk group uses DID, private numbers or use default number programmed in *68-2. 8.For BRI and PRI, run Program *66-5 to assign Listed Directory Numbers (LDNs) to channel groups and index numbers. Then run Program *66-6 to assign LDN channel groups to ISDN trunk ports. Then run Program *66-7 to assign trunk groups to each LDN. 9. Run Program *68-2 to define whether or not the system presents a “user” Caller ID when calls are made using ISDN services. This program also enables users to dial a code to send Caller ID or to block it. 10.For BRI and PRI, run Program *69 to define the parameters for Calling Number to be presented. Each phone can present a unique Caller ID number, such as DID or a private number. If neither is chosen, the default is the Directory Number entered in Program *68-2 for outgoing calls. 11.Run Program *64 to assign each ISDN trunk group to use DID/DNIS programming, or to provide Direct in Line (DIL) ringing for incoming calls. 12.See “ISDN Related Programs” on Page 7-3, and make the appropriate programming assignments.
Toshiba ISDN. System Programs Overview System programming for ISDN has some commonalities with T1 programming. ISDN programming follows two different paths, depending on whether Basic Rate Interface (BRI) or Primary Rate Interface (PRI) is being installed. Refer to Flowchart 7-1 and the following steps for an overview of each program’s purpose. ISDN features can be programmed onto the following systems as described: o DK40i and DK424 – support BRI S/T o DK40i and DK424 – will support BRI U basic features in Release 4.2 o DK424 supports PRI. ISDN Related Programs o Program 10-1–System Assignments: LED 16 determines the initialization process for BRI lines connected to the ISDN network (see 3-27). o Program 10-4–ACD/ISDN Parameters: LEDs 11 and 12 set the T-Wait ISDN PRI and BRI timers, respectively. T-wait timers are used for reducting overloading ISDN terminals during concurrent initialization, such as following a wide-area power failure. LED 13 specifies whether or not 3.1 kHz audio calls can be received as speech calls. LED 14 enables the ISDN “Start” code to be sent when the Speed Dial button is pressed (see 6-6). o Program 39–Telephone Feature Buttons and Program 59–Attendant Console Feature Buttons: A new button is provided called the 6WDUW button in both Program 39 (see 3-135) and 59 (see 3-111) to defeat the timeout when dialing is complete. Also a 6XE button is provided for entering sub-address information. o Programs 55-1 and 55-2–Least Cost Routing Modified Digits: Enables a sub-address separator (see 5-21). o Program 30 – Station Class of Service: LED 12 defines whether the button is used for dialing a sub-address or as a dialing separator. When LED 12 is ON, only the standard stations are affected. The DKT and EKT ports are ignored (see 3-73). If the first dialed digit is , the will be treated as a feature access code. If is entered at any point after that (in other words, as long as it is not the first digit dialed), then the system will treat the following numbers as a subaddress. A second should then be entered to indicate the end of the subaddress.
DID Digit Length Selection (Memory Block 1-1-20) Use this Memory Block to define the number of DID digits. Default: 3 8.3.2 DID Digit Conversion Assignment (Memory Block 1-1-21) Use this Memory Block to enable the DID Digit Conversion table. Default: NO 8.3.3 DID Digit Conversion Table (Memory Block 1-1-22) Use this Memory Block to assign DID numbers to ring at station numbers, closed number (plus outgoing digits), or tenant number. Default: Not Specified 8.3.4DID Forward Station Number for Busy Station or Undefined Digit (Memory Block 1-1-23) Use this Memory Block when the DID conversion Table is enabled to define where digits are routed when undefined or the station is busy. Default: NON (Not Assigned)
Call by Call Service allows multiple services to share a PRI line. When a call is originated or terminated, an Information Element (IE) called the Network Specified Facility (NSF) is added to the SETUP message to identify the service associated with the call. The number of Simulated Facility Groups (SFGs) that can be simultaneously used for each service must be restricted. The total number of SFGs must be less than the number of PRI channels. The SFGs are determined when contracting with the network. The network determines whether calls that exceed the restriction are rejected or diverted. For outgoing calls, the KTS counts the number of calls in progress per group, and rejects excess calls. For Call by Call to operate, the number of B channels used for PRI must be specified using Memory Block 1-13-00 (PRT Channel Assignment), and Call by Call service must be assigned to each PRT using Memory Block 1-13-03 (Call by Call Service Selection).
Call Wait (Camp-on) ● After receiving station busy tone, dial＊ . ● Camp-on tone is heard in the called station. ● When called party answers, talk or hang up to transfer another call to the called party. Last Number Redial ● Lift handset, press OHD/Speaker or dial from active keypad. ● Dial 5 5 2 or press speed button + dial ‘＊’ + Hold/Save button. Storing Station Speed Dial Numbers ● Press the [TRANS/PGM] and [SPEED] Button. ● Dial speed bin number. Range (XXX-ZZZ) ● Dial speed dial number you wish to store. (iPECS-MG : Dial number with CO Access code) ● Press the [HOLD/SAVE] button. ● Enter the name associated to the number. ● Press the [HOLD/SAVE] button. ● You will hear confirmation tone. Using Station Speed Dial Numbers ● Press the [SPEED] button ● Dial the desired speed dial bin number. Group Call Pick-up When hearing an unattended phone ringing in your area, ● Lift handset. ● Dial 5 5 6. ● You will be connected automatically to the caller. ※ Note: you must be in the same pick-up group.
Placing an Outside Call ● Lift handset, press OHD/Speaker or dial from live keypad. ● Dial 9. ● Dial the desired number. Placing an Intercom Call ● Lift handset, press OHD/Speaker or dial from active keypad. ● Dial the station number. Placing an Outside Call on Hold ● While connected to an external call, press [HOLD] button. Retrieving a Outside Call on Hold ● Press the flashing flexible button. ※ Note: calls will automatically recall after pre-defined time Re-directing an Incoming Call (Call Pick-up) When you hear another phone ringing in your area, ● Lift handset, press OHD/Speaker or dial from live keypad. ● Dial 7. ● Dial the extension number of the ringing station. ● You will be connected automatically to the caller.
(E1 PRI) Overview E1 PRI trunks are provided by the installation of an E1 PRI trunk card into the IP Office control unit. E1 PRI trunk cards are not supported with the IP Office Small Office Edition control unit. Dual port E1 PRI trunk cards are only supported with the IP412 control unit and in Slot A of the IP406 V2 control unit. For full details of installation refer to the IP Office Installation manual. Each physical E1 PRI trunk port supports up to 30 channels for calls. E1 trunks can be set to either ETSI or QSIG operation modes. T1 trunks are provided by the installation of an T1 PRI trunk card into the IP Office control unit. The trunks on these cards can be configured for T1, PRI or QSIG operation. For full details of installation refer to the IP Office Installation manual. Dual port T1 PRI trunk cards are only supported with the IP412 control unit and in Slot A of the IP406 V2 control unit. Each physical trunk port supports up to 24 channels in T1 mode, 23 channels in PRI and QSIG modes. • Dialing Complete The majority of North-American telephony services use en-bloc dialing. Therefore the use of a ; is recommended at the end of all dialing short codes that use an N. This is also recommended for all dialing where secondary dial tone short codes are being used.
As we described in the previous issue, Network Address Translation or NAT is a means of expanding the number of available IP addresses that can be used under the IPv4 addressing scheme. With NAT certain publicly routable IP addresses are provided to a company that has an Internet connection. Other, specific numbers are used by the company for internal communications only, and are not publicly routable, nor are they visible to anyone outside the Local Area Network of that company. Firewalls do not apply NAT to the Application Layer. As SIP is an Application Layer protocol, the IPv4 addresses and domain resolution are not translated for Application Layer routing. SIP traffic cannot traverse these traditional enterprise firewalls and NAT devices, and as a result, the firewall/NAT device incorrectly routes all SIP traffic, which includes Voice over IP. When a SIP phone call attempts to traverse a typical firewall, although the TCP/IP addressing is correct, the IP addresses within the SIP protocol information header are not corrected. As a result, when a far-end WAN device receives a SIP request the SIP addresses are the private IP addresses of the SIP device behind the typical firewall. These private IP addresses are not routable back to the original source. Ingate fixes these issues. Ingate SIParators/Firewalls contain a SIP Proxy, SIP B2BUA, SIP Media Relay, and a SIP Registrar – features not found on traditional firewalls -- that allow the traversal of the IP addresses within the SIP protocol. The Ingate Firewall or SIParator uses these tools to replace the private addresses with publicly routable addresses so that the calls can be connected, and then assigns the correct internal IP address to the call so that it can be delivered to the proper recipient on the inside of the network. The Ingate will allow the network traversal of VoIP (and SIP trunking) calls to various carriers/service providers from the IP-PBX. It controls both incoming and outgoing SIP communications and routes it to the intended users and devices. The advantage of the Ingate Firewall is that it will allow all voice traffic as well as data traffic to traverse the enterprise firewall/NAT/AL
SIP trunking is a term applied to the services offered by LECs (Local Exchange Carriers), ILECs (Independent Local Exchange Carriers), CLECs (Competitive Local Exchange Carriers) and ITSPs (Internet Telephony Service Providers) to terminate Voice over IP (VoIP) calls to the Public Switched Telephone Network (PSTN). SIP Trunking allows enterprises and small businesses to eliminate a PSTN gateway at their site and outsource that function to a carrier. It is typically a lower-cost alternative to Primary Rate Interfaces (PRIs) because SIP trunks can be purchased in single-trunk increments (as compared to 23 channel increments for a PRI). Other ways in which SIP trunks decrease costs: q With SIP trunks, a single network can be maintained within the organization, rather than having both a voice and data network. q Internet bandwidth can be used more efficiently. q Moves, Adds and Changes can be completed without major wiring upgrades. SIP Trunks are delivered in several ways: Over the Public Internet – SIP Trunking Anywhere Allows any enterprise, anywhere, to adopt SIP Trunking and assign some, possibly unused, bandwidth to voice at no extra charge for the connection, and providing the highest ROI. Managed Services Carriers supply a dedicated, fully managed connection from their Point of Presence to the enterprise site. This service offers quality of service guarantees, but is somewhat more expensive. MPLS Delivery The carrier, usually an LEC, ILEC or CLEC, will delivery a managed service using Multi-Protocol Label Switching to insure the highest voice quality and reliability. The voice quality, even over an un-managed public Internet connection, is excellent. Typical savings over PRIs range from 40-60% with the payback period for the equipment required, which may include an upgrade to the IP-PBX and the installation of an Ingate SIParator or Firewall, has been shown to range from 4 – 12 months. With these facts in mind, there is no question that SIP Trunking offers compelling advantages for businesses large and small.
SIP invitations are used to create sessions that carry session descriptions, which allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features for users. SIP also offers a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols, such as UDP, TCP and TLS. The SIP requests and responses are written in plain text within the datagram of the IP Header. Contained in the SIP requests and responses are the addresses of the source and the destination of the participants. These addresses are SIP URI’s, which have a UserInfo and Host Address, and this host address can either be an IP address or a domain name. For example, a SIP URI can look like “sip:firstname.lastname@example.org.” Therefore, the routing of SIP is done using IPv4 addresses at the Application layer and does not route at the Transport or Network layer. As the addressing and routing of SIP are done at the Application layer, the biggest problem the SIP protocol now has is the disconnect between the IPv4 addressing and routing at the Application layer versus the IPv4 addressing and routing at the Transport and Network layers. Network Address Translation (NAT) occurs at the Transport and Network layers. AVAYA, NORSTAR, NEC, SAMSUNG, MITEL, PANASONIC, TOSHIBA t.646.872.2025
NEC ASPIRE MULTI BUTTON TELEPHONE. If you and your co-workers handle each other's calls, you might want to be in a Department Callinggroup (page 37). Someone calling your group's number goes through to any- one who's available. You can even have Department Step Calling (page 39) send your personal calls to your group when you're not avail- able. To answer a call already ring- ing a co-worker's phone, use Group Call Pickup (page 54). When you're on a call and you want the others in your area to listen in on the conversation, activate Group Listen (page 55). Your co-work- ers hear the call through your telephone's speaker. If you frequently call the same co-worker, you can have Ringdown (page 105) automatically call them for you.
AVAYA Partner Backup and Restore You should back up the system programming periodically especially if you are changing the processor module or upgrading the system, or before and after any major programming changes. You can back up your system programming to either the internal memory of the PARTNER ACS R7.0 processor or a Backup/Restore PC Card. You can back up the programming automatically or manually. Occasionally, you may have to restore programming from the backed-up file. The system may automatically restore a backup translation image if the system detects that the battery backed up translation image is corrupted.
AVAYA Partner ■ If you want to be able to intercept calls routed to an auxiliary device—such as an answering machine, a voice messaging system, or an auto attendant—make sure Automatic Extension Privacy is Not Assigned for the auxiliary equipment extension. ■ Single-line telephones and system telephones without a programmed Privacy button cannot override this feature once it is assigned to an extension. ■ If Automatic Extension Privacy is Assigned at an extension, the green light is lit automatically after programming the Privacy button to indicate that Privacy is currently active. ■ Automatic Extension Privacy applies only to active calls. Any user can retrieve a held call unless Exclusive Hold is used.