Customer Interaction Center PBX Feature List

CIC provides a full digital telephone system with features like ANI, DNIS, and T-1/E-1 Support.

PBX System Features

  • Fully programmable Dial Plans.
  • Customizable routing schemes.
  • Virtual PBX – Remote users with same functionality as in the office.
  • Support for TLS and SRTP (New in 3.0).
  • May configure use of SRTP for selected stations, by line, or region-to-region calling. (New in 3.0).
  • Multiple sites connected via TIE lines.
  • NT-like administrative interface.
  • Universal line ports – PBX trunk ports also serve as ACD ports, voice messaging ports, IVR ports, fax ports, and custom voice processing ports.
  • Graphical application design tool for completely open interface to all PBX processing.
  • Processor is the “Interaction Engine” processing thousands of transactions per second
  • Interaction Engine is event-driven – PBX interaction events treated same as Web, ACD, and data events.
  • Real-time status viewing of PBX events including line activity, station activity, and user activity.
  • Advanced, open reporting architecture – standard reports for line activity and user call detail.

Trunk Lines

  • Trunk types supported:
    • Analog CO
    • T-1
    • ISDN Primary
    • E-1
    • EuroISDN
    • ISDN BRI
  • Trunk Allocation:
    • Inbound only
    • Outbound only
    • Both Inbound and Outbound
    • Line Groups – lines may be in multiple line groups
  • T-1 Line Configuration:
    • Configured per channel (supports fractional T1’s)
    • Direction (Inbound/Outbound/Both)
    • Supports Wink or Immediate Start
    • Channel/Board/Line Mapping
  • Primary Rate ISDN Configuration
    • Configuration settings per service provider
    • AOC-E billing protocol has been added to the ISDN services. This allows European companies to obtain charge information at the conclusion of a call. National ISDN support has been added for North America

Dialing Plans

  • Dialing PlansFully Programmable (using graphical design tool) if desired.
  • Standard Dial Plan configured in Interaction Administrator.
  • Unlimited amount of Dial Classifications.
  • Classification assignment based on number patterns
  • Least cost routing – identify dialed number to allocate active call for least cost routing
  • Enable localized dial tone frequencies when configuring in various countries.
  • Local Exchanges identified in Dial Plan configuration.
  • Local Area Code identified in Dial Plan configuration.
  • North American Numbering Plan – CIC ships with a standard dial plan configuration in Interaction Administrator.
  • International dialing plan – customize dial plan configuration for international formats.
  • User security to block users from dialing international, long distance, 900 toll and other configured call classifications. (Dial Plan may be configured in Interaction Administrator to pick up other types of toll dialing)
  • Dial Plan handling when placing calls automatically blocks users/stations without dialing privileges.
  • Dial Plan supports required account codes for specific number types such as associating an account to a long distance call.
  • Dial Plan supports number lists for easier pattern matching.
  • Support for authorization codes to be entered from an individual station phone to track calls to users. Calls appear on the call detail report.
    • Dial the phone number, enter two asterisks (*), then dial user extension number and password. For example, 355 1553 +** + 8467 1234
  • SIP numbering plan – Configure a specific SIP line group for a particular gateway for a SIP station. This means that geographic routing can be handled.
  • SIP support for regional dial plans:
    • Map codec usage to particular locations.
    • Dial pattern selection based on region the devices reside in. This will allow you to configure remote locations to use a lower bandwidth codec.
    • Use Dial plan filters to configure area codes and least cost routing on the network out local gateways.
    • Set up locations based on LAN configuration.
    • Complete overview of each location with its devices assigned to it. This also allows you to view the different codecs used between each location configured in the system.
  • Simulate Call feature to perform complete dial plan testing without making actual call.
  • Allow for reservation of lines for specific users/dial classifications.

Caller Identification

  • ANI / Caller ID
  • ANI/Caller ID Custom Number/Name Pairs for display to users and voicemail text configured in Interaction Administrator
  • DNIS/DID table for routing directly to user or workgroup queue
  • Interaction Attendant automatically routes DNIS/DID to automated attendant menus, greetings, etc.
  • Analog Caller ID Configuration:
    • Enable/Disable CallerID on trunk
    • DNIS Enabled/Disabled on T1 channel (Analog DID requires additional Exacom interface box)
  • Caller ID Formats Supported:
    • CLASS (USA)
    • CLIP (UK)
    • ACLIP (Singapore)
  • T-1 Configuration:
    • ANI Enabled (Yes/No)
    • DNIS Enabled (Yes/No)
    • Max Wait for ANI/DNIS
    • Max Digits
    • Inter-Digit Delay Timeout

Extensions

  • Extensions – User Assigned Extension follows User Workstation Login.
  • Workstation Extension May be fixed to station.
  • Standalone telephones for break rooms, warehouse floors, conference rooms, and users with no Interaction Client interface.
  • ADSI phones (e.g. Aastra Powertouch 350) for message waiting indicators and caller ID displays. **See note on last page
  • Analog telephone sets (compatible with most analog telephone devices)

PBX Call Routing Features

  • Workgroup of users as a “hunt group” – configure users in a workgroup to perform sequential ringing. Will ring available users as opposed to physical stations.
  • Group Ring – configure user workgroup to simultaneous ring all users in the workgroup.
  • Station Group – sequential
  • Station group – simultaneous
  • Station group – must answer or timeout
  • Time of day routing on all of the above.
  • Configured Station Features:
    • Supports station restrictions such as international, long distance and 900 numbers
    • Dialing plan and speed dials are programmable through the graphical design interface on CIC.
    • Call forward for stations when busy/on phone
    • Must answer
    • Timeout
  • Automatic route selection
  • Night service – route calls to particular station groups after hours.
  • Music on hold (music shipped with system is royalty free).
  • Operator Target – user or administrator may set their operator target out of voicemail. This also may be an Interaction Attendant profile so that operator target may be a menu selection or based on time of day the user destination for the operator target may change.

Station Handset Features

  • Analog – connected to station board
    • Hook-flash menu
      • Conference calls
      • Transfer caller (blind transfer)
      • Hold caller
    • Message waiting indicator (has caveats based on unified messaging usage and type of phone device).
    • Call Pick-up from a telephone Pick up a call using a code sequence including any extension or workgroup queue that the user/station has rights to access.
    • Account code support by dialing *and the account code after the dialed number.
    • Authorization codes – Support for authorization codes to be entered from an individual station phone to track calls to users. Calls appear on the call detail report.
    • CallerID – Type II Bellcore phones.
    • Caller Name – with supported callerID interfaces on the telephony boards and display phone.
    • Call waiting tone.
    • Press the * key to reach the system’s main menu.
  • Analog – example menu phone Interactive Intelligence i390 additional menu features from buttons on phone and screen.
    • DTMF dialing from handset for outbound calls
    • Pickup
    • Access voicemail
    • Set status to DND
    • Call forward
    • Change status
    • Login to station
    • Call control – hold, redial, transfer, conference, pick up new call, flash to access flash features, disconnect, take a call off of hold, disconnect current call.
    • Message waiting indicator
    • Speaker
    • Speaker volume up/down
    • Mute
    • Recent calls list
    • Dial a recent call
    • Delete a call from recent calls list
  • Analog – connected to channel bank
    • Same as above – channel bank must support features

SIP

  • Use validated phone model which ensures phone’s compliance with SIP RFC’s.
  • Support for authorization codes to be entered from an individual station phone to track calls to users. Calls appear on the call detail report.
  • Call parking on Polycom phones – park calls to an orbit and the system holds the call on that orbit.
  • Call park pickup on Polycom phones – press the call park button again and then dial the orbit where the call is parked.
  • Zone paging – available on Polycom SIP-compliant telephones. Page another user or small group of users on the intercom of the phone.
  • Group pickup – Dial a short key sequence to pick up a call ringing within your stations group.
  • Shared line appearances on Polycom IP phones. Executive/Assistant and others can share a line. This means that a particular extension appears on two phones.
  • Example phone model: Features of the Polycom IP 550.
    • Hold
    • Transfer (consult and blind)
    • Conference (3-way)
    • Split conference into 2 calls on hold
    • Call pickup via *95 + extension
    • Call Park and pickup parked orbit call
    • List all parked calls
    • Group pickup
    • Zone paging
    • Shared line appearances
    • CallerID
    • Caller name
    • Distinctive Ring (New in IC 3.0)
    • Free seating/hoteling/hot desking – user may log in to the phone via *98 extension plus password
    • Access voicemail via messages button
    • Last number redial
    • Message waiting indicator
    • Multiple call appearances
    • Multiple station appearances (not bridged)
    • Mute
    • Display – 320×160 pixels grayscale
    • Date and time display
    • Who am I? – extension is displayed
    • Call log including missed calls
    • Call timer
    • Stored directory on phone by user of phone
    • Speaker – Full duplex
    • Multiple Ethernet ports
    • Power over Ethernet requires separate inline power cable
    • Supports DNS SRV records
    • Volume buttons
    • Hands free RJ9 headset jack
    • Support for SRTP (New in IC 3.0)

Auto-Provisioning of Polycom Phones (New in IC 3.0)

  • Users or Administrators may provision a managed SIP phone with a configured station, using a default system provisioning menu. (New in IC 3.0)

Configuration and Maintenance of Polycom Phones

  • Administrators may set phone model templates with feature sets directly in Interaction Administrator. (New in IC 3.0)
  • Administrators may alter feature sets on Polycom phones directly from Interaction Administrator, push the updated file down to the phone and set the phone to reboot automatically. (New in IC 3.0)
  • Administrators may schedule the reboot for one or multiple Polycom phones directly from Interaction Administrator and plan the reboot for after hours. (New in IC 3.0)
  • Administrators may push new firmware down to Polycom phones directly from Interaction Administrator. (New in IC 3.0)
  • Administrators may manage Polycom firmware levels on a per-model basis. (New in IC 3.0)

Toll Restrictions

  • Long Distance – station and/or user
  • International – station and/or user
  • 900 Service – station and/or user
  • Control at Multiple Levels – set up control of security by the following:
    • Specific User
    • Users in Workgroup
    • Default User Setup
    • Specific stations

Call Control Features From Interaction Client Windows version

  • Conference Calls:
    • User Conference – user may set up the conference.
    • Moderated Conference – an administrative assistant may set up a conference and disconnect themselves from the conference after it is created.
    • Conference outside parties – conference any combination of internal and/or external parties.
  • Conference Parties per Telephone Interface:
    • The following explains the maximum size of 1 conference call and is dependent upon the telephony architecture chosen. In all cases please see the telephony product notes for details on resource planning.
      • Intel Springware boards – maximum is 16 parties in one conference from the DISI boards.
      • Intel DM3 boards – maximum is 60 parties in one conference.
      • Aculab – maximum is 96 with echo cancel if all parties are on the same Prosody board.
      • Intel HMP software – Maximum is 64 conference parties – be sure that you understand how this fits with other resource limitations – it may not be feasible in most configurations to reach this maximum based on RTP audio resource availability.
    • Transfer conference calls
    • Simple, Graphical Interface
  • Call Transfers
  • Consult transfer
  • Blind
  • Send call to user voicemail
  • Internal
  • External
  • Camp-on
    • Camp-on another user and be notified when:
      • User changes status to an available state
      • User changes status to any other status
  • Park calls – calls may be parked on a user’s extension.
  • Alert on parked call – based on a timer set in the Interaction Client – a parked call may alert the user that it is still parked.
  • Timeout on parked calls – Parked calls may revert back to an operator or another extension after a timeout has been reached.
  • Call Coverage – such as executive/assistant applications.
    • Call forward all calls.
    • Call forward busy/DND – all call, external only or internal only.
    • Call forward ring no answer all calls, external only, or internal only.
    • Send calls to original user’s voicemail if second destination does not answer.
    • Send calls to second user’s voicemail if second destination does not answer.
  • Paging Support
    • From the Interaction Client users may page another IC user’s pager. Support for alphanumeric and numeric pagers.

SIP Support

  • Full support for SIP as defined in the SIP RFCs 3261 and 2543.
  • CIC server functions as a SIP gateway, a SIP call proxy/redirect server, and as a SIP softswitch.
  • Support for mixed traditional telephony trunks and analog stations with SIP stations in the same server.
  • Support for mixed traditional telephony trunks with analog stations and IP trunks connected via an optional SIP gateway
  • Support for all SIP-based IP telephony solution for all trunks and stations.
  • Supports dynamic audio processing in all- SIP configurations for greater scalability and reliability.
  • Support for Dialogic HMP—an all-software based switching and media processing matrix for IP telephony.

SIP Soft Phone (New in IC 3.0)

  • Configure as a station.
  • Use in conjunction with a USB headset and either the Interaction Client .NET Edition, Outlook Edition, or the basic soft phone interface.
  • User or administrator may auto-provision the SIP Soft Phone.
  • User may select the codec to use.

Interaction Media Server

  • Handles low level audio processing outside the IC server for increased scalability and system reliability
  • No lost audio in the case of switchover or loss of connection to the CIC server. Audio remains in progress
  • MxN architecture: multiple Interaction Media Servers can support activity for one IC server or multiple IC servers can point to a single Interaction Media Server
  • May be “regionalized” to reduce activity across the network to conserve bandwidth
  • Supports call recording, live monitoring, transcoding, and playing of queued audio (music or pre-recorded announcements while callers are in ACDWait state).
  • Supports QoS
  • Supports playing of on hold music, ringback, call recording compression using TrueSpeech, and transcrypting (RTP to SRTP or vice versa) (New in IC 3.0)
  • Support for out of box audio for all SIP calls – even recorded ones. Without SIP media server, IC server media resources are used for call recording in SIP implementations. 2 RTP audio resources are used in the CIC server without the media server.
  • Call recording with G.711 RTP Audio compression
  • Call recording with G.729 RTP Audio compression
  • Redundant media servers supported

Legacy PBX Integration

  • TIE lines from IC server to legacy PBX – connect to another PBX via TIE lines for conference and transfer.
  • Remote IC users – connect virtual users via IC’s Remote Interaction Client – requires a TIE line for each simultaneous call to a remote user. Remote ACD agents require a TIE line or channel for each remote agent connected on a persistent connection.
  • Simplified Message Desk Interface (SMDI)
  • Use CIC behind PBX for Unified Messaging
  • Release Link Transfer (RLT) Support
    • Support RLT (AKA take back and transfer) with select PBXs when calls are switched through the PBX to CIC.
  • Q.SIG name and number send/receive support between CIC and Q.SIGcompatible PBXs

Multiple telephony platform support

  • Support for Dialogic software-based Host Media Processing (HMP), Dialogic H.100 bus, Aculab H.100 bus, and Cisco AVVID (CallManager).
  • Aculab Support
    • Dynamic reconfiguration of Aculab lines and network interface settings
    • Aculab support to reinitialize a trunk without restarting the server
    • Support for TDD/TTY on Aculab-based systems

Cisco CallManager Support

  • Improved scalability in Cisco CallManager environments (up to 255 CTI ports per server) and support for multiprocessor CIC servers.
  • Remote stations – configure remote stations that are regularly used by users and may require licensing.

**Note: These features require handler development. Please check with your District Sales Manager or certified Interactive Intelligence Partner for quotation on implementation services.

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