PBX Specifications

Introduction

Avaya Messaging is a vendor neutral application which can integrate with many telephone systems. Use the information in this section as a general guideline to the kinds of integrations Messaging can support, as well as what is required from the phone system in order for specific features to work. While Messaging requires SIP for integration, media gateways provide compatibility for PBXs that use different protocols.

Multiple PBX/Node Support

Avaya Messaging is able to integrate with multiple PBXs or nodes at once, allowing you take advantage of all the capacity a site may have. This will also be a great option for legacy sites which are implementing additional PBX for higher capacity. A site can add new PBX to their legacy system and use them as one through Messaging without having worry about compatibility.

The only requirement for multi-PBX / multi-node support is that the PBX in question is compatible with Avaya Messaging. It is recommended that all PBXs/nodes utilize SIP trunks.

SIP

SIP (Session Initiation Protocol) is a set of rules that provide the basic signals used to initiate, manage, and terminate communications sessions. SIP is an open standard which allows carrier voice equipment to interoperate seamlessly with customer premise equipment. The logical voice channel established between them is a SIP Trunk. A SIP Trunk is a virtual phone line that utilizes the customer's Internet connection for access. Trunks come with unlimited local inbound minutes and long distance usage based at rates far below traditional phone service. SIP Trunking is a business-class telecommunications solution that delivers local, toll-free, domestic and international long distance service.

SIP integration is highly recommended due to reduced costs and universal compatibility. Even most legacy hardware will be able to utilize SIP through a compatible media gateway.

Tested & Verified PBX

SIP Direct

Following PBXs are integrated directly into the voice server through SIP.

* - Fax detection is not supported through Iwatsu ECS.

SIP Gateway

Following PBXs are integrated through a media gateway to the Messaging server.

For specific part numbers of SIP Gateways, please refer to the Dialogic Media Gateway portions of the following sections:

•T1 on page 76

•T1 on page 76

•E1 on page 78

CTI

CTI stands for Computer Telephone Integration. CTI combines data with voice systems in order to enhance telephone services. For example, automatic number identification (ANI) allows a caller’s records to be retrieved from the database while the call is forwarded to the appropriate party. An outbound example would be automatic telephone dialing from an address list.

Tested & Verified PBX

CTI Integration: Rich Presence and Call Control

Following PBXs are integrated directly into the voice server through CTI.

Inband

Inband integration is possible on supported switches through the use of Dual Tone Multiple Frequency (DTMF) signalling. Strings of DTMF tones are transmitted on the analog voice channel after the channel connects to answer the call but before the voice is cut through.

Typically the string contains the Calling Line Identification, the Called Party Identification, reason for the call (re-direction or direct call), and will allow the following functionality among others:

•Direct Log-In in which Messaging recognizes a direct station call, identifies the internal caller's extension number and prompts the caller to enter the security code of the subscriber mailbox associated with that number.

•Call Forwarding to a personal greeting if an extension is Busy, not answered, Do Not Disturb or Forward All. Both internal and external callers can be forwarded to the subscriber's personal greeting. Depending on how the subscriber's mailbox is configured and what information is provided by the telephone system an appropriate greeting can be played for both a Busy and a Ring No Answer condition. Callers can then leave messages in the subscriber's mailbox or be presented with another list of options through voice menus.

•Call Routing based on the trunk number, DNIS number or forwarded PBX extension number. These types of calls can be routed to a specific mailboxes, ACD agent or call center. The time frame (when the voicemail is waiting for signaling) is configurable so that it can be adjusted regardless of the PBX.

T1

SUPPORTED PROTOCOLS

Robbed Bit Signaling

Channel Associated Signaling (CAS), also referred to as Robbed Bit Signaling, is a method of signaling each traffic channel rather than having a dedicated signaling channel (like ISDN). In other words, the signaling for a particular traffic circuit is permanently associated with that circuit.

The most common forms of CAS signaling are loopstart, groundstart, Equal Access North American (EANA) and E&M. The biggest disadvantage of CAS signaling is its use of user bandwidth to perform signaling functions. In addition to receiving and placing calls, CAS signaling also processes the receipt of Dialed Number Identification Service (DNIS) and automatic number identification (ANI) information which is used to support authentication and other functions.

Each T1 channel carries a sequence of frames. These frames consist of 192 bits and an additional bit designated as the framing bit, for a total of 193 bits per frame. Super Frame (SF) groups 12 of these 193 bit frames together and designates the framing bits of the even numbered frames as signalling bits. CAS looks specifically at every sixth frame for the timeslot's or channel's associated signaling information. These bits are commonly referred to as A- and B-bits. Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively.

ISDN Signaling Concepts

The Integrated Services Digital Network (ISDN) is a digital communications network capable of carrying all forms (voice, computer and facsimile) of digitized data between switched end points. This network is a digital-switched system that makes a connection only when requested.

Control over switched connections is provided by a protocol of messages that pass between the two ends of the digital link. Any type of equipment can be connected to an ISDN provided the equipment is capable of generating a digital bit stream that conforms to ISDN standards.

ISDN technology offers the benefits inherent in digital connectivity such as fast connection (setup and tear down), fast Direct Dialing In service (DDI) and fast Automatic Number Identification (ANI) acquisition.

ISDN protocols use an out-of-band signaling method carrying signaling data on a channel or channels separate from user data channels. This means that one signaling channel (D channel) carries signaling data for more than one bearer channel (B channel). This signaling technique is referred to as common channel signaling (CCS). Signaling data carries information such as the current state of the channel (for example, whether the telephone is on-hook or off-hook). Common channel signaling allows the transmission of additional information, such as ANI and DNIS digits, over the signaling channel.

An ISDN Primary Rate Interface (PRI) trunk provides a digital link that carries some number of TDM (Time Division Multiplexed) channels:

•   a T-1 trunk carries 24 64 Kbit channels, 23 voice/data channels (B channels) and one signaling channel (D channel) on a single 1.544 MHz digital link

•   an E-1 trunk carries 32 64 Kbit channels, 30 voice/data channels and two additional channels: one signaling channel (D channel) and one framing channel to handle synchronization on a single 2.048 MHz digital link.

The ISDN digital data stream contains two kinds of information: user data and signaling data used to control the communication process. For example, in telephony applications user data is digitally encoded voice data. Voice data from each time slot is routed to a separate B channel. Signaling data carries information such as the current state of the channel (for example, whether the telephone is on-hook or off-hook). The signaling information for all B channel information is routed to the D channel of the device.

Dialogic Media Gateway

SIP

The Dialogic® 2000 Media Gateway Series is a turnkey appliance that seamlessly merges traditional PSTN technology with IP networks. This economical gateway helps consolidate typically separate voice and data networks and provides new and differentiated communications services. Without making radical, disruptive, and expensive upgrades to existing PBX equipment, service providers and enterprises can realize the benefits of a converged voice and data network.

E1

QSIG is the European association for Standardising Information And Communication Systems. QSIG has a long history of producing standards related to the interworking of communications equipment within Private Integrated Services Networks (PISNs). PISN standardisation is undertaken by the various task groups of its technical committee: TC32. Much of their effort is put towards the definition of the intraPISN signalling system commonly referred to as "QSIG".

QSIG is an ISDN based protocol for signalling between nodes of a Private Integrated Services Network (PISN)

Dialogic Media Gateway

SIP

The Dialogic® 2000 Media Gateway Series is a turnkey appliance that seamlessly merges traditional PSTN technology with IP networks. This economical gateway helps consolidate typically separate voice and data networks and provides new and differentiated communications services. Without making radical, disruptive, and expensive upgrades to existing PBX equipment, service providers and enterprises can realize the benefits of a converged voice and data network.