Alcatel OmniPCX Bedienungsanleitung PDF herunterladen (Seite 6)

Alcatel-Lucent OmniPCX Enterprise Communication Server6

ABC-R: Routing Mechanism

• Adaptive routing

• Automatic Route Selection (ARS):

¬ Access to alternative routes

based on caller rights

¬ Multi-tenant

¬ ARS time dependent: day of the

week, hour, and minute of the day

¬ ARS per IP domain

• Cost limit barring per user/

installation status

• Dialing command tables with

add/delete digits for number

• Direct or indirect multi-carrier access

• Five daily tables per weekly table

• Information (voice prompt)

to callers if cheapest route

is not available

• Information (voice prompt) to

caller if he/she needs permission

to use a more expensive route

• Break in

• Break out

• Break in via secured DISA

• Forced on net homogeneous

numbering plan

ABC Network Management

• Management from attendant

• CLI local management

• Centralized or distributed man-

agement from external applications

• Topology map

• Configuration

• Alarm management

• Call accounting

• Directory

• Traffic analysis

• Security

• Graphical user interface

• Remote management

• Alarm domains and alarm

centralization

• Audit mechanism

• Broadcast of configuration data

• Remote maintenance access

• Remote software downloading

Voice-Based Virtual

Private Networks (VPN)

• ABC VPN via overflow

to public ISDN, QSIG

• Dissociating signaling over IP

network (lower bandwidth)

• Full ABC services

QSIG – Basic Call

• Calling line identification

• Connecting line identification

• Data calls bearer

• Malicious call identification

• Sub-address

QSIG – Generic

Function Protocol

• QSIG – Supplementary services

• Advice of charge

• Call completion on no reply

• Call completion to busy

subscriber

• Call forwarding – busy

• Call forwarding – no reply

• Call forwarding – unconditional

• Calling/connected line

identification restriction

• Calling/connected name

identification restriction

• Calling line identification

presentation

• Calling name identification

presentation

• Call transfer

• Call offer

• Connected line identification

presentation

• Connected name identification

presentation

• Generic functional procedures

• Message waiting indication

• Path replacement

DPNSS

• Call back when free (call

completion on busy phone) (9)

• Call forwarding (11)

• Call offer (14)

• Call waiting (17)

• Centralized night service (25)

• Conference (13)

• Do not disturb (32)

• Hold (12)

• Inquiry call (13)

• Intrusion (10)

• Loop avoidance (37)

• Route optimization (19)

• Service independent

string (16) name

• Transfer (13)

• Voice and data call (6 and 7)

Session Initiation

Protocol (SIP)

• SIP proxy/registrar/redirect

server and SIP gateway

• Server redundancy (active/passive)

• Authentication (http digest)

Supported SIP standards

The Alcatel-Lucent OmniPCX Enter-

prise Communication Server

is based on the latest SIP standards.

The following RFCs are supported.

They can be found on the IETF site

at http://www.ietf.org

SIP messages are used in strict

conformance with RFCs.

Standards driven SIP evolutions

for 3rd party endpoints and SIP

trunking:

• SIP Telephony services described

in draft ietf sipping service

example-15

• RFC 4504 conformance SIP

Telephony Device Requirements

and Configuration (hardware,

IP, security characteristics, SIP

conformance, …)

RFC SIP

• 2543 (obsolete by RFC 3261,

3262, 3263,3264, 3265): SIP:

Session Initiation Protocol

• 2782: A DNS RR for specifying the

location of services (DNS SRV)

• 2822: Internet Message Format

• 3261: SIP: Session Initiation Protocol

• 3262: Reliability of Provisional

Responses in SIP (PRACK)

• 3263: SIP: Locating SIP Servers

• 3264: An Offer / Answer model

with SDP

• 3265: SIP-Specific Event Notification

• 3311: The SIP UPDATE Method

(session timer only)

• 3323: Privacy Mechanism for the

Session Initiation Protocol (SIP)

• 3324: Short term requirements for

network asserted identity

• 3325: Private Extensions to the

Session Initiation Protocol (SIP)

for Asserted Identity within

Trusted Networks

• 3265: SIP-specific Event Notification

• 3515: The Session Initiation Proto-

col (SIP) refer method

• 3891/3892: The Session Initiation

Protocol (SIP) ‘Replaces’ Header/

Referred-By Mechanism

• 3398: Integrated Services Digital

Network (ISDN) User Part (ISUP)

to SIP Mapping

• 3966: The telephone URI for

telephone numbers (url tel not

supported)

• 4497: Inter-working between SIP

and QSIG

draft-ietf-sip-answermode-07:

Requesting Answering Modes for

the Session Initiation Protocol

MMUSIC (Multiparty Multimedia

Session Control)

• 2327: SDP: Session Description Protocol

IP Telephony

• 2617: HTTP Authentication : Basic

and Digest Access Authentication

• 1321: Authentication for Outgoing

calls

• 2833/4733: DTMF Transparency.

RFC 2833 replaced by RFC 4733

• 3842: A message Summary and

Message Waiting Indication Event

Package

• 4028: The session timers in the

Session Initiation Protocol

• 3725: Best current practices for

Third party Call Control (3 pcc) in

SIP (scenario 1). Invite without SDP

• 3960: Early Media (partial)

AFT (Audio fax Transmission)

• 1889/1890: RTP: A transport pro-

tocol for Real-Time applications

• 2198: RTP Payload for Redundant

Audio data

• 3550: RTP: A Transport Protocol for

Real-Time application (audio only)

• 3551: RTP Profile for Audio and

Video Conferences with Minimal

Control (audio only)

• 3711: The Secure Real Time.

Supported on A-LU IP Phone and

Softphone

• 3362: T38 ITU-T Procedures for

real time Group3 Fax Relay /

communications over IP

• 2617: HTTP Authentication: Basic

and Digest Access Authentication

SIP end-point features

• Call hold

• Consultation hold/toggle

• Transfer early attended

• Transfer attended

• Call forward, unconditionally

• Call forward on busy

• Call forward on no answer

• Third-party conference

• Voice mailbox access

• Message waiting Indication

• Do not disturb

• Call admission control

• Accounting

• CoS/barring

• Speed dial

• DTMF

• Fax T.38

• Twinset

• Hunting group

• CSTA monitoring and Call Control

Hospitality and Infocenter

compatibility

SIP remote extension features

• RFC 2976 –DTMF info

• Call hold

• Consultation hold/toggle

• Transfer early attended

• Transfer attended

• Call forward, unconditionally

• Call forward on busy

• Call forward on no answer

• Third-party conference

• Call park/unpark

• Call pickup/supervision

• Automatic call back on busy

or no answer

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