The information contained in this document is believed to be accurate in all respects
but is not warranted by Mitel Networks Corporation.
The information is subject to change without notice and should not be construed in
any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its
affiliates and subsidiaries assume no responsibility for any errors or omissions in this
document. Revisions of this document or new editions of it may be issued to incorporate such changes.
No part of this document can be reproduced or transmitted in any form or by any
means - electronic or mechanical - for any purpose without written permission from
Mitel Networks Corporation.
TRADEMARKS
The trademarks, service marks, logos and graphics (collectively “Trademarks”)
appearing on Mitel's Internet sites or in its publications are registered and unregistered
trademarks of Mitel Networks Corporation (MNC) or its subsidiaries (collectively
"Mitel") or others. Use of the Trademarks is prohibited without the express consent
from Mitel. Please contact our legal department for additional information:
legal@mitel.com
For a list of the worldwide Mitel Networks Corporation registered trademarks, please
refer to the website: http://www.mitel.com/trademarks.
Mitel’s Power Over Ethernet (PoE) Powered Device (PD) products are covered by one
or more of the U.S. patents (and any foreign patent counterparts thereto) identified at
Mitel’s website: www.mitel.com/patents
For more information on the PD patents that are licensed, please refer to.
Here you will find information relating to safety, data protection and legal matters
besides product and documentation information.
Please read through the product and safety information carefully.
1. 1About Mitel
Mitel (Nasdaq:MITL) (TSX:MNW) is a global leader in business communications that
easily connect employees, partners and customers - anywhere, anytime and over any
device, for the smallest business to the largest enterprise. Mitel offers customers maximum choice with one of the industry’s broadest portfolios and the best path to the
cloud. With more than US$1 billion in combined annual revenue, 60 million customers
worldwide, and #1 market share in Western Europe, Mitel is a clear market leader in
business communications. For more information, go to www.mitel.com.
1. 2Product information
Purpose and function
MiVoice Office 400 is an open, modular and comprehensive communication solution
for the business sector with several communication servers of different performance
and expansion capacity, an extensive telephone portfolio and a multitude of expansions. They include an application server for unified communications and multimedia
services, an FMC controller for mobile phone integration, an open interface for application developers, and a multitude of expansion cards and modules.
The business communication solution with all its elements was designed to cover the
full spectrum of communication requirements of businesses and organizations in a
user and maintenance-friendly way. The individual products and parts are co-ordinated
and cannot be used for other purposes or replaced by outside products or parts (except to connect up other authorized networks, applications and phones to the interfaces certified for that purpose).
User groups
The phones, soft phones and PC applications of the MiVoice Office 400 communication solution are particularly user friendly in design and can be used by all end users
without any specific product training.
The phones and PC applications for professional applications, such as the PC operator
consoles or call centre applications, do require training of the personnel.
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Product and Safety Information
Specialist knowledge of IT and telephony is assumed for the planning, installation, configuration, commissioning and maintenance. Regular attendance at product training
courses is strongly recommended.
User information
MiVoice Office 400 products are supplied with safety and product information, quick
user guides and user guides.
These and all other user documents such as system manuals are available for download from the MiVoice Office 400 DocFinder as individual documents or as a documentation set. Some user documents are accessible only via a partner login.
It is your responsibility as a specialist retailer to keep up to date with the scope of functions, the proper use and the operation of the MiVoice Office 400 communication solution and to inform and instruct your customers about all the user-related aspects of the
installed system:
• Please make sure you have all the user documents required to install, configure and
commission an MiVoice Office 400 communication system and to operate it efficiently and correctly.
• Make sure that the versions of the user documents comply with the software level of
the MiVoice Office 400 products used and that you have the latest editions.
• Always read the user documents first before you install, configure and put an
MiVoice Office 400 communication system into operation.
• Ensure that all end users have access to the User Guides.
Download the MiVoice Office 400 documents from the internet: www.mitel.com/DocFinder
Mitel Schweiz AG hereby declares, that the MiVoice Office 400 products
• comply with the basic requirements and other stipulations of Directives EMC
(2014/30/EU) and LVD (2014/35/EU).
• are manufactured in conformity with RoHS according to Directive 2011/65/EU.
You can find the product specific declarations of conformities under www.mi-
tel.com/regulatory-declarations.
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Product and Safety Information
Use of third party software
MiVoice Office 400 products comprise, or are partially based on, third-party software
products. The licence information for these third-party products is listed in the user's
guide of the MiVoice Office 400 product in question.
Exclusion of Liability
(Not valid for Australia. See the chapter "Limited Warranty (Australia only)", page 20
for the limited guarantee in Australia.)
All parts and components of the MiVoice Office 400 communication solution are manufactured in accordance with ISO 9001 quality guidelines. The relevant user information
has been compiled with the utmost care. The functions of the MiVoice Office 400 products have been tested and approved after comprehensive conformity tests. Nonetheless errors cannot be entirely excluded. The manufacturers shall not be liable for any
direct or indirect damage that may be caused by incorrect handling, improper use, or
any other faulty behaviour. Potential areas of particular risk are signalled in the appropriate sections of the user information. Liability for loss of profit shall be excluded in
any case.
Environment
MiVoice Office 400 products are delivered in recycled, chlorine-free corrugated cardboard packaging. The parts are also wrapped inside a protective fleece made of polyethylene foam fleece or polyethylene film for added protection during shipping. The
packaging is to be disposed of in accordance with the guidelines stipulated under current legislation.
MiVoice Office 400 products contain plastics based on a pure ABS, sheet
steel with an aluminium-zinc or zinc finish, and epoxy resin-based PCBs.
These materials are to be disposed of in accordance with the guidelines stipulated under current legislation.
MiVoice Office 400 products are disassembled exclusively using detachable screwed
connections.
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Product and Safety Information
1. 3Safety Information
Reference to hazards
Hazard warnings are affixed whenever there is a risk that improper handling may put
people at risk or cause damage to the MiVoice Office 400 product. Please take note of
these warnings and follow them at all times. Please also take note in particular of hazard warnings contained in the user information.
Operating safety
MiVoice Office 400 communication servers are operated on 115 or 230 VAC mains
power. Communication servers and all their components (e.g. telephones) will not operate when mains power fails. Interruptions in the power supply will cause the entire
system to restart. A UPS system has to be connected up-circuit to ensure an uninterruptible power supply. Up to a specific performance limit a Mitel 470 communication
server can also be powered redundantly using an auxiliary power supply. For more information please refer to your communication server's system manual.
When the communication server is started for the first time, all the configuration data is
reset. You are advised to backup your configuration data on a regular basis as well as
before and after any changes.
Installation and operating instructions
Before you begin with the installation of the MiVoice Office 400 communication server:
• Check that the delivery is complete and undamaged. Notify your supplier immedi-
ately of any defects; do not install or put into operation any components that may be
faulty.
• Check that you have all the relevant user documents at your disposal.
• During the installation follow the installation instructions for your MiVoice Office 400
product and observe to the letter the safety warnings they contain.
Any servicing, expansion or repair work is to be carried out only by technical personnel
with the appropriate qualifications.
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Product and Safety Information
1. 4Data protection
Protection of user data
During operation the communication system records and stores user data (e.g. call
data, contacts, voice messages, etc.). Protect this data from unauthorised access by
using restrictive access control:
• For remote management use SRM (Secure IP Remote Management) or set up the
IP network in such a way that from the outside only authorised persons have access
to the IP addresses of the MiVoice Office 400 products.
• Restrict the number of user accounts to the minimum necessary and assign to the
user accounts only those authorisation profiles that are actually required.
• Instruct system assistants to open the remote maintenance access to the communication server only for the amount of time needed for access.
• Instruct users with access rights to change their passwords on a regular basis and
keep them under lock and key.
Protection against listening in and recording
The MiVoice Office 400 communication solution comprises features which allow calls
to be monitored or recorded without the call parties noticing. Inform your customers
that these features can only be used in compliance with national data protection provisions.
Unencrypted phone calls made on the IP network can be recorded and played back by
anyone with the right resources:
• Use encrypted voice transmission whenever possible.
• For WAN links used for transmitting calls from IP or SIP phones, use as a matter of
preference either the customer's own dedicated leased lines or VPN encrypted connection paths.
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Product and Safety Information
1. 5About this document
This document describes the system functions and features of communication servers
of the MiVoice Office 400 series. The expansion stages, system capacity, installation,
configuration, the operation and maintenance, the technical data, the DECT planning,
and the possibilities for networking several systems into a private network (PISN) or an
Mitel Advanced Intelligent Network (AIN) are not part of this Manual. They are described in separate documents.
The document is intended for planners, installers and system managers of phone systems. Basic knowledge of telephony, especially of ISDN and IP technology, is needed
to understand the content.
The system manual is available in Acrobat Reader format and can be printed out if
necessary. Navigation in PDF format is based on the bookmarks, table of contents,
cross references and index. All these navigation aids are linked, i.e. a mouse click
takes you directly to the corresponding places in the Manual. We have also ensured
that the page numbering in the PDF navigation corresponds to the page numbering of
the Manual, making it much easier to jump to a particular page.
Referenced menu entries and parameters appearing on terminal displays or on the
user interfaces of the configuration tools are highlighted in italics and in colour for a
clearer orientation.
Special hazard alert messages with pictograms are used to signal areas of particular
risk to people or equipment.
Hazard:
Failure to observe information identified in this way can put people and hardware at risk through
electrical shock or short-circuits respectively.
Warning:
Failure to observe information identified in this way can cause a defect of the product or to a
module.
Note:
Failure to observe information identified in this way can lead to equipment faults or malfunctions
or affect the performance of the system.
General Considerations
Special symbols for additional information and document references.
Note
Failure to observe information identified in this way can lead to equipment faults or malfunctions
or affect the performance of the system.
Tip
Additional information on the handling or alternative operation of equipment.
See also
Reference to other chapters within the document or to other documents.
Mitel Advanced Intelligent Network
Particularities that have to be observed in an AIN.
References to the MiVoice Office 400 configuration tool WebAdmin
If an equals sign is entered in the WebAdmin search window , followed
by a two-digit navigation code, the view assigned to the code is directly displayed.
Example: Licence overview (=q9) view
The corresponding navigation code is available on the help page of a view.
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Product and Safety Information
1. 6Limited Warranty (Australia only)
The benefits under the Mitel Limited Warranty below are in addition to other rights
and remedies to which you may be entitled under a law in relation to the products.
In addition to all rights and remedies to which you may be entitled under the Competition and Consumer Act 2010 (Commonwealth) and any other relevant legislation, Mitel
warrants this product against defects and malfunctions in accordance with Mitel's authorized, written functional specification relating to such products during a one (1) year
period from the date of original purchase ("Warranty Period"). If there is a defect or
malfunction, Mitel shall, at its option, and as the exclusive remedy under this limited
warranty, either repair or replace the product at no charge, if returned within the warranty period.
Exclusions
Mitel does not warrant its products to be compatible with the equipment of any particular telephone company. This warranty does not extend to damage to products resulting
from improper installation or operation, alteration, accident, neglect, abuse, misuse,
fire or natural causes such as storms or floods, after the product is in your possession.
Mitel will not accept liability for any damages and/or long distance charges, which result from unauthorized and/or unlawful use.
To the extent permitted by law, Mitel shall not be liable for any incidental damages, including, but not limited to, loss, damage or expense directly or indirectly arising from
your use of or inability to use this product, either separately or in combination with
other equipment. This paragraph, however, is not intended to have the effect of excluding, restricting or modifying the application of all or any of the provisions of Part 5-4 of
Schedule 2 to the Competition and Consumer Act 2010 (the ACL), the exercise of a
right conferred by such a provision or any liability of Mitel in relation to a failure to comply with a guarantee that applies under Division 1 of Part 3-2 of the ACL to a supply of
goods or services.
This express warranty sets forth the entire liability and obligations of Mitel with respect
to breach of this express warranty and is in lieu of all other express or implied warranties other than those conferred by a law whose application cannot be excluded, restricted or modified. Our goods come with guarantees that cannot be excluded under
the Australian Consumer Law. You are entitled to a replacement or refund for a major
failure and for compensation for any other reasonably foreseeable loss or damage.
You are also entitled to have the goods repaired or replaced if the goods fail to be of
acceptable quality and the failure does not amount to a major failure.
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Product and Safety Information
Repair Notice
To the extent that the product contains user-generated data, you should be aware that
repair of the goods may result in loss of the data. Goods presented for repair may be
replaced by refurbished goods of the same type rather than being repaired. Refurbished parts may be used to repair the goods. If it is necessary to replace the product
under this limited warranty, it may be replaced with a refurbished product of the same
design and colour.
If it should become necessary to repair or replace a defective or malfunctioning product
under this warranty, the provisions of this warranty shall apply to the repaired or replaced product until the expiration of ninety (90) days from the date of pick up, or the
date of shipment to you, of the repaired or replacement product, or until the end of the
original warranty period, whichever is later. Proof of the original purchase date is to be
provided with all products returned for warranty repairs.
Warranty Repair Services
Procedure: Should the product fail during the warranty period and you wish to make a
claim under this express warranty, please contact the Mitel authorized reseller who
sold you this product (details as per the invoice) and present proof of purchase. You
will be responsible for shipping charges, if any.
Limitation of liability for products not of a kind ordinarily acquired for personal, domestic
or household use or consumption (e.g. goods/services ordinarily supplied for businessuse).
Limitation of liability
1.1 To the extent permitted by law and subject to clause 1.2 below, the liability of Mitel to you for any non-compliance with a statutory guarantee or loss or damage arising out of or in connection with the supply of goods or
services (whether for tort (including negligence), statute, custom, law or on any other basis) is limited to:
a) in the case of services:
i)the resupply of the services; or
ii)the payment of the cost of resupply; and
b) in the case of goods:
i)the replacement of the goods or the supply of equivalent goods; or
ii)the repair of the goods; or
iii) the payment of the cost of replacing the goods or of acquiring equivalent goods; or
iv) the payment of the cost of having the goods repaired.
1.2 Clause 1.1 is not intended to have the effect of excluding, restricting or modifying:
a) the application of all or any of the provisions of Part 5-4 of Schedule 2 to the Competition and Consumer
Act 2010 (the ACL); or
b) the exercise of a right conferred by such a provision; or
c) any liability of Mitel in relation to a failure to comply with a guarantee that applies under Division 1 of Part
3-2 of the ACL to a supply of goods or services.
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Product and Safety Information
After Warranty Service
Mitel offers ongoing repair and support for this product. If you are not otherwise entitled
to a remedy for a failure to comply with a guarantee that cannot be excluded under the
Australian Consumer Law, this service provides repair or replacement of your Mitel
product, at Mitel's option, for a fixed charge. You are responsible for all shipping
charges. For further information and shipping instructions contact:
Manufacturer:
Mitel South Pacific Pty Ltd ("Mitel")
Level 1, 219 Castlereagh Street
Sydney, NSW2000, Australia
Phone: +61 2 9023 9500
Note:
Repairs to this product may be made only by the manufacturer and its
authorized agents, or by others who are legally authorized. Unauthorized repair will void this express warranty.
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System interfaces
2System interfaces
This chapter features the different types of digital and analogue network and terminal interfaces and points out a number of configuration particularities. The chapter
ends with special interfaces for door intercoms and general bells.
Tab. 1System interfaces and channels
Ter mExplanation
B channel User information channel: Each connection occupies one user information
D channelControl and signalling channel: Channel for control and signalling as well as
2B+D / 30B+D2 B channels and 1D channel / 30B channels and 1D channel
PortsPhysical connection points on the communication server for network inter-
Network interfacesNetwork-side connection possibilities for the communication server
Special interfacesOther connection possibilities for the communication server
• Ethernet interface on the basic system
• Door Intercom SystemsSpecial interface for connecting door intercom systems
• General BellSpecial interface for general bell
1)
CAS (channel-associated signalling) is also used in some countries (e.g. Brazil).
external
channel, e.g. 2 user information channels (connections) can be occupied
simultaneously using one basic access.
for packet data transfer.
faces and terminal interfaces
Digital network interface 2B+D:
A terminal interface S configured as BRI-S external.
1)
For connection to one or more SIP providers. An SIP access contains a maximum of 30 channels.
An analogue network connection has 1 user information channel.
Digital terminal interface 2B+D:
Connection for Euro ISDN terminals, Terminal Adapters and ISDN PC cards.
A maximum of 2 digital system phones or one DECT radio unit can be operated on a proprietary DSI bus.
Digital terminal interface for linking up IP system phones and SIP phones
(softphones and desk phones).
An analogue terminal connection has 1 user information channel.
Central interface for connecting WebAdmin, a CTI server, IP system phones,
SIP terminals, for the network-side connection to an SIP service provider, or
to implement a private network, etc.
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System interfaces
2. 1Network Interfaces
The system supports the following types of network interfaces:
• Basic rate interface BRI-T for connection to
– the public ISDN network
– the private leased-line network
• Basic access BRI-S
– the private leased-line network
– a terminal with its own direct dialling plan (DDO)
• Primary rate access PRI for connection to
– the public ISDN network
– the private leased-line network
• SIP access via the Ethernet interface on the basic system for connection to SIP provider.
• Analogue network interface FXO for connection to the public analogue network
external for connection to
2. 1. 1Basic Access Variants
A basic access is a digital network interface for connection to the public network or to
the private leased-line network. It can be set for the protocols DSS1 (public ISDN network) and QSIG / PSS1 (private leased-line network).
A basic access has two 64 kbit/s user information channels and one 16 kbit/s control
and signalling channel (2B+D).
One call or data connection can be set up on each bearer channel (B channel) simultaneously and independently of the other B channel.
A basic access can be barred for outgoing calls (Outgoing barred).
Basic accesses for connecting the communication server to the public network can be
operated as point-to-point and, with some network providers, also as point-to-multipoint (multiple subscriber number) access.
There are two types of basic access:
• Basic rate interface BRI-T
• Basic rate interface BRI-S
external
2. 1. 1. 1Basic rate interface BRI-T
Basic access T is suitable for connection to both the public ISDN network and the private-leased-line network.
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System interfaces
PINX 1PINX 2
BRI-T
BRI-T
S
extern
451
...
455
222 224 51
...
222 224 55
Communication
server
Device with DDI
evaluation
BRI-TBRI-T
S
external
2. 1. 1. 2Basic rate interfaces BRI-S external
The basic access S external is a BRI-S interface configured as external (setting Pro-
tocol BRI-S) in the interface configuration).
The basic access BRI-S
• For connection to the private leased-line network or
• For connecting DSS1 terminal equipment, which evaluates the DDI number sent by
the communication server and routes the call accordingly (e.g.an external fax
server, see also "Direct Dialling Out (DDO)", page 33
Fig. 1BRI-S external in a private leased-line network: PINX-PINX connection
external is designed for the following purposes:
).
Fig. 2BRI-S
external in a DDI configuration
Note:
An BRI-S interface configured as external is a fully-fledged network interface and is no longer
available as a user-network interface. A basic access BRI-S
nection to the public ISDN network.
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external cannot be used as a con-
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System interfaces
2. 1. 1. 3Point-to-Point and Point-to-Multipoint Connections
Basic accesses can be configured as point-to-point or as point-to-multipoint (TEI
Management setting in the configuration of the network interfaces).
Point-to-Multipoint Connection without a communication server
The basic access in point-to-multipoint configuration allows a selective dial-up of the
terminals connected in parallel using MSN, the Multiple Subscriber Number. Here the
network itself provides a kind of direct dialling, so to speak.
Point-to-Multipoint Connection with communication server
If a communication server is connected using point-to-multipoint, a direct dial number
must be created for each MSN number, with all the digits of the MSN number.
NT1: Network Terminal
MSN: Multiple Subscriber Number
U/T:ISDN reference point
DSI:Digital user-network interfaces
FXS: Analogue terminal interface
Fig. 4Basic rate interface in point-to-multipoint configuration, with single-digit direct dial and par-
Combinations are also possible in the case of several lines, e.g. one line in point-tomultipoint configuration and the remaining in point-to-point configuration.
allel terminal
Note:
If terminals (e.g. MSN1) are connected in parallel on the BRI-T interface, “Collision detec-
tion” has to be activated as the communication server and the terminal influence each other.
This also applies in cases where analogue connections are used on NT1.
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System interfaces
NT1
NT1
NT1
FXS
BRI-S
DSI
U
U
U
403 03 10
"
"
Communication
server
Line group
BRI-T
BRI-T
BRI-T
Point-to-Point Connection without Direct Dial
Without direct dialling in, only one call number is available. The individual internal users can only be reached indirectly via the number.
This variant is suitable above all for systems with primarily outgoing traffic.
Fig. 5Several basic accesses with line group in point-to-point configuration, without direct dial
number
Note:
Do not connect any terminals between the NT1 and the communication server.
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System interfaces
FXS
BRI-S
DSI
U
U
U
403 20 99
403 20 00
NT1
NT1
NT1
Communication
server
BRI-T
BRI-T
BRI-T
Line group
with DDI
DDI No.
Point-to-Point Connection with Direct Dial
With direct dial the individual communication server users can be reached directly via
their direct dial number.
NT1: Network Terminal
DDI: Direct dialling
U/T:ISDN reference point
DSI:Digital user-network interfaces
BRI-S: ISDN terminal interface
FXS: Analogue terminal interface
Fig. 6Several basic accesses with line group in point-to-point configuration, with direct dial number
Note:
Do not connect any terminals between the NT1 and the communication server.
Periodic Reactivation of Layer 2 on the BRI-T Interface
1)
Layer 2 of the BRI-T network interface can be reactivated periodically every three minutes so that incoming calls are not rejected already at the local exchange after potential temporary interruptions in the U interface. To do so, set the parameter Layer2
activation of the BRI-T network interface to Special.
Note:
In some countries BRI-T network interfaces are deactivated once a certain amount of time has
elapsed without traffic, and are only reactivated when the communication server once again
requests a connection.
1)Only in Germany and Austria.
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System interfaces
FXS
BRI-S
DSI
U
U
U2
2B+D
2B+D
30B+D
Communication
server
Channels
PRI
BRI-T
BRI-T
NT1
NT1
NT1
PINX 1PINX 2
PRI
PRI
PRI
2. 1. 2Primary rate interface PRI
A primary rate access is a digital network interface for connection to the public network
or the private leased-line network. It can be set for the protocols DSS1 (public ISDN
network) and QSIG / PSS1 (private leased-line network).
A primary rate access has thirty 64 kbit/s user information channels and one 64 kbit/s
control and signalling channel (30B+D). One call or data connection can be set up on
each bearer channel (B channel) simultaneously and independently of the other B
channel.
Notes:
Primary rate accesses can only be used as point-to-point connections.
Fig. 8Primary rate access in a private leased-line network: PINX-PINX connection
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System interfaces
CAS on the primary rate interface
CAS (channel-associated signalling) is also used in some countries (e.g. Brazil). With
this method the signalling data is transmitted over the voice channel. The type of signalling can be selected for each PRI interface (=dg).
Note:
DSP resources (CAS senders/receivers) are required for CAS. With Mitel 470, DSP resources
for 1 PRI interface are already available on each basic system (on the DSP chip on the call
manager card with fixed function assignment). If more CAS senders/receivers are needed, they
can be assigned on a configurable DSP chip. On an Mitel 430, the DSP resources for CAS
senders/receivers must always be assigned on a configurable DSP chip (=ym).
2. 1. 2. 1Clock synchronization
The clock frequency of a communication server is provided (synchronized) by the public network via the basic accesses BRI-T and the primary rate accesses PRI.
Should synchronization by the public network fail (due, for example, to exchange line
interruptions), the communication server will use its own clock. This frequency deviates
at most by 5 ppm from the nominal value, which ensures that the Mitel DECT system
also remains available.
In a private leased-line network, PINXs that are synchronized by the public network
pass on the clock reference to PINXs that are not connected directly to the public network.
Synchronisation on a private fixed network must be carefully planned to ensure there
are no synchronisation loops.
All the private leased-line network connections and public exchange line circuits are
automatically in a shared clock reference table when the communication server is configured for the first time.
If a communication server is not networked in a PISN, the clock reference table can be
left as it is; only the initial reference may have to be assigned differently.
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System interfaces
BRI-S
ext.
Cells
Communication server
Up-circuit
system
PRI
BRI-T
2. 1. 2. 2Digital down-circuit connection with QSIG
If a down-circuit communication server is connected with an up-circuit communication
server via digital lines (BRI-T, PRI), all the features as per QSIG are available providing
the up-circuit communication server supports the QSIG protocol.
The down-circuit communication server is configured in accordance with the rules for
networked systems.
The up-circuit communication server has a connection to the public network. It can also
be an MiVoice Office 400 system or a third-party product, provided it supports the
QSIG protocol.
As a rule the down-circuit communication server is connected with the up-circuit communication server via its own fixed lines. The interfaces can be basic rate interfaces
(BRI-T) or primary rate interfaces (PRI). Connections on BRI-S
also possible instead of connections on a BRI-T interface, provided at least one BRI-T
interface is available for synchronization via the ISDN network.
Example: Down-circuit connection with cordless system
external interface are
Fig. 9Digital down-circuit connection with QSIG
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77
76
75
DDIBRI-S
Fax receivers with
allocated a DDI
numbers
Fax servers
ISDN
Communica-
tion server
S
external
2. 1. 2. 3Direct Dialling Out (DDO)
If an external fax server is connected to an S bus, individual fax recipients allocated a
DDI number can be specifically addressed. In terms of routing technology, this corresponding to a DDO (Direct Dialling Out) function.
The external fax server forwards the incoming faxes via e-mail to the relevant PC stations that are set up as fax recipients.
Fig. 10Direct Dialling Out (DDO) to a fax server
Due to the configuration of the BRI-S interface as BRI-S external and the use of the
DSS1 protocol, the fax calls can be routed via routes and trunk groups. This means
that all fax receivers that have been allocated a DDI number can be reached via a single BRI-S interface.
See also:
"Call to a DSS1 terminal equipment on the S Bus (DDO)", page 206
Tip:
The CPU2 applications card of an Mitel 470 communication server already contains a fax
server and its use is subject to the acquisition of the relevant licences.
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2. 1. 3SIP
2. 1. 3. 1What is SIP?
The Session Initiation Protocol (SIP) is a network protocol used for setting up, controlling and clearing down a communication session between two or more subscribers
(source: Wikipedia). SIP is an open standard and was developed by an IETF (Internet
Engineering Task Force) working group. While the text-based protocol has a great deal
in common with HTTP (Hypertext Transfer Protocol) in terms of both structure and sequence, it is not compatible with it.
SIP is now widely used in IP telephony. However SIP alone cannot enable VoIP connections. With the aid of the Session Description Protocol (SDP), SIP merely negotiates the communication modalities between the SIP subscribers. The actual audio
data stream is exchanged via other, more suitable protocols, such as the Real-Time
Transport Protocol (RTP) or the encrypted Secure Real-Time Transport Protocol
(SRTP). For this, the coded and compressed data is packed into packets and sent via
the User Datagram Protocol (UDP) or the Transmission Control Protocol (TCP).
The SIP connection is used to transmit not just voice but other multimedia data, too
(video, fax, text, etc.).
SIP subscribers have an address whose structure is similar to that of an e-mail address (e.g. URL: "sip:12345@sip-server.com"). SIP subscribers can be reached via
this address, regardless of their location. However, this is only possible if they register
with an SIP provider and regularly update their IP address.
Gateways at the SIP providers enable the transition into the public telephone network,
for example the leased-line network or the mobile phone network.
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SIP provider
Location Service
Registrar server
Proxy server
Redirect server
User Agent
User Agent
System interfaces
System components
SIP is based on a client-server architecture. Components may include a User Agent,
Registrar Server, Proxy Server and Redirect Server. The three servers are located at
the SIP provider and may be installed on the same system.
Fig. 11SIP components
• User Agent
User Agents are applications at SIP endpoints, i.e. software or hardware-based
components. The caller is referred to as the User Agent Client; the called party, as
the User Agent Server.
• Registrar Server
An SIP subscriber regularly sends his registration data and his IP address to the
Registrar Server. This information is stored in a database (Location Service).
• Proxy Server
The Proxy Server is responsible for contacts between the subscribers. Following a
request from a User Agent Client, the Proxy Server contacts the Registrar Server to
determine the current IP address of the User Agent Server. It then tries to make
contact with the User Agent Server.
• Redirect Server
The Redirect Server works in a way similar to the Proxy Server. However it hands
over the IP address of the User Agent Server directly to the User Agent Client, who
then takes charge of the connection setup.
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INVITE
OK
ACKNOWLEDGE
User Agent ServerUser Agent Client
Types of connection setup
Requests and responses are defined in SIP in order to set up a connection between
two subscribers. The User Agent Client generates a request, to which the User Agent
Server responds with a response.
There are essentially three methods for setting up an SIP connection. The descriptions
below are greatly simplified and explain only the principle and the different methods.
Method 1: Direct connection setup between the User Agents
The User Agent Client sends the "INVITE" request for a connection setup to the User
Agent Server. If the User Agent Server takes the call, he sends back the response
"OK" together with the connection parameters. The User Agent Client confirms this
with an "ACKNOWLEDGE" and the call connection is set up.
Fig. 12Direct connection setup
As the IP address changes depending on the User Agent’s location, this method does
not guarantee that the connection is set up.
Method 2: Connection setup using a Proxy Server
The User Agent Client sends the "INVITE" request for a connection setup with the User
Agent Server to the Proxy Server. The Proxy Server retrieves the current IP address of
the User Agent Server from the database of the location service and forwards the connection request to the User Agent Server. If the User Agent Server takes the call, it
sends the response "OK" back to the Proxy Server, which in turn forwards it to the User
Agent Client. The response contains all the connection parameters. From this point onwards the two User Agents communicate directly with each other. The User Agent Client confirms the connection parameters with an "ACKNOWLEDGE" and the call connection is set up.
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INVITE
INVITE
OK
OK
ACKNOWLEDGE
Location Service
User Agent ServerProxy ServerUser Agent Client
Database query
<IP address>
<IP address>
ACKNOWLEDGE
OK
INVITE
INVITE
ACKNOWLEDGE
User Agent Server
Location Service
Database query
<IP address>
Redirect Server
<IP address>
User Agent Client
Fig. 13Connection setup using a Proxy Server
This type of connection requires that the User Agents register with the Registrar Server
and regularly update their data.
Method 3: Connection setup using a Redirect Server
The User Agent Client sends the "INVITE" request for a connection setup to the Redirect Server. The Proxy Server retrieves the current IP address of the User Agent
Server from the database of the location service and sends it back to the User Agent
Client, who confirms the action with an "ACKNOWLEDGE". The User Agent Client now
sets up a direct connection with the User Agent Server, as described in "Method 1: Direct connection setup between the User Agents", page 36.
Fig. 14Connection setup using a Redirect Server
This type of connection also requires that the User Agents register with the Registrar
Server and regularly update their data.
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Signalling dataSignalling data
Communication server
as SIP provider
SIP phone
Voice data
SIP phone
2. 1. 3. 2Security aspects with VoIP
Security is an important aspect of VoIP telephony. The table below shows the three security objectives of data protection, authentication and integrity as well as ways of
achieving those objectives:
Tab. 2Security objectives
Security objective MeaningRemedy
Data ProtectionThird parties must not be able to read the
exchanged data
AuthenticationVerifying the identity of the remote stationUsing shared passwords and certificates
IntegrityThird parties must not be able to modify the
transmitted data
With these considerations it is important to note that the voice data and the signalling
data do not always run in parallel and may well take separate paths, as the following
example shows:
Data encryption
Using checksums
Fig. 15SIP data streams
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Situation without encryption (SIP/RTP)
If the signalling data and voice data are considered separately, the situation is as follows if encryption is not used:
Tab. 3Situation without encryption
Security objectiveSignalling dataVoi ce d a ta
Data ProtectionNot guaranteed.Not guaranteed
AuthenticationPartially guaranteed through password protectionNot guaranteed
IntegrityNot guaranteedNot guaranteed
Solutions
• Encryption of the SIP and RTP data at IP level using IPSec (Internet Protocol Security) and VPN (Virtual Private Network). The signalling data and the voice data are
protected if all the SIP components involved are within the VPN.
• Encryption of the SIP signalling data at the transport level using TLS (Transport
Layer Security) and of the voice data at application level using SRTP (Secure RealTime Transport Protocol).
For WAN links via the internet it makes sense to combine both methods.
Securiing the signalling data with TLS:
TLS works by exchanging certificates and requires the TCP transport protocol. The
communication server generates a trusted certificate and automatically uploads it to
the Mitel SIP phones, which then restart. A call connection between communication
server and terminal is established only if the two certificates match.
For standard SIP terminals the trusted certificate must be exported as a file and manually uploaded to the terminal. Certificates remain valid for long periods; however for security reasons they should be replaced at regular intervals. New certificates must also
be generated manually whenever the IP address of the communication system
changes. The settings can be found in the Certificate (=u9) view.
Securing voice data with SRTP:
The SRTP protocol is used to secure the voice data. Please note the following points:
• VoIP encryption must be activated (=3n).
• VoIP mode (=ym) must be set to Secure G.711 or Secure G.711/G.729.
•The NTP service (=ty) must be activated.
•A Secure VoIP licence is required.
Note:
Securing signal data with TLS, and voice data with SRTP, is also important for the connection
between the communication server and an SIP provider, as well as between the SIP nodes of a
private SIP network.
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See also
For more details on this subject please refer to the “Mitel Advanced Intelligent Network (AIN)
and IP Terminals” System Manual and to the online help.
2. 1. 3. 3SIP in MiVoice Office 400
If SIP is supported in MiVoice Office 400, Method 2: Connection setup using a Proxy
Serveris used exclusively.
A distinction is made between the following three application cases:
• Connection of SIP terminals as internal subscribers to MiVoice Office 400:
In this case, the MiVoice Office 400 communication server assumes the role of an
SIP provider for the SIP terminal and provides the Registrar and the Proxy Servers
internally. The terminal can be connected either internally on the same IP network
as the MiVoice Office 400 communication server or externally via a VPN connection
or using SRTP and TLS. This particular application is described in the chapter "IP
terminal interface", page 50.
• Connection of MiVoice Office 400 to one or more SIP providers:
Here the MiVoice Office 400 communication server itself is the User Agent. Access
to an SIP Provider is provided via an SIP network interface (SIP access). One SIP
access supports up to 30 SIP channels, i.e. up to 30 connections to one SIP provider are possible simultaneously. Access to the public telephone network is via a
gateway at the SIP provider. The connection to an SIP provider is described in the
chapter "SIP access", page 43
• Networking MiVoice Office 400 communication servers via SIP:
It is possible to network two or more MiVoice Office 400 communication servers via
SIP. The principle is comparable to QSIG networking on an ISDN basis. In the same
way as with QSIG networking, star-shaped centralised networking configurations as
well as meshed networking configurations are possible. For more information, refer
to the system manual "Private Networking with MiVoice Office 400".
.
Data encryption is designed to ensure that security is taken into account in all three application cases, especially when the VoIP data leaves the LAN. They can be external
home workstations, a connection of the communication server to the public network via
an SIP provider or the SIP networking of several systems at different locations.
SIP support in MiVoice Office 400 is continually being expanded and therefore depends on the software version of the communication server. A general overview of the
protocols and methods currently supported can be found in Tab. 4, Seite 41.
See also
You can find more useful information on SIP in MiVoice Office 400 such as FAQs, compatibility
lists, restrictions and support tips in the Knowledge Base on the Extranet site: https://pbx-
web.aastra.com.
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2. 1. 3. 4SIP RFCs supported by MiVoice Office 400
RFCs (Request for Comments) are numbered, freely accessible technical and organisational documents on the internet. They are drawn up by the IETF (Internet Engineering Task Force) and go through various stages until in the ideal scenario they establish
themselves as a standard. There is a whole series of RFCs dealing directly or indirectly
with SIP.
The RFCs are published on the following web site. Specific RFCs can be displayed directly using a search engine; you can also search for RFCs using keywords:
http://www.rfc-editor.org
The following RFCs are supported for connecting MiVoice Office 400 to SIP service
providers, terminals to MiVoice Office 400, and for SIP networking:
Tab. 4SIP RFCs supported by MiVoice Office 400
RFCTitle
2617HTTP Authentication: Basic and Digest Access
Authentication
2833
RTP Payload for DTMF Digits, Telephony Tones and
4733
Telephony Signals
3261SIP: Session Initiation Protocol including support for
TCP
3262Reliability of Provisional Responses in Session Initia-
4855Media Type Registration of RTP Payload Formats
4975The Message Session Relay Protocol
5246The Transport Layer Security (TLS) Protocol Version
1.2
5806Diversion Indication in SIP
Supported on
the SIP network interface
(SIP access)
✓✓✓
✓✓✓
✓✓✓
✓✓✓
✓
✓✓✓
✓✓✓
✓✓✓
✓✓✓
✓✓✓
Supported on
the SIP terminal interface
✓
✓
✓
✓
✓
✓
✓
✓
Supported
for SIP networking
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2. 1. 3. 5SIP access
The communication server can be connected to one or more SIP providers via the
Ethernet interface on the basic system. The communication server supports 10 SIP access with up to 30 channels per SIP access. One SIP Access Channels licence is required for each channel.
The communication server handles the SIP access in the same way as analogue or
digital network interfaces, i.e. they are grouped in one or more separate trunk groups.
The allocation to an SIP provider is defined for each trunk group. This means for example that international calls can be routed via SIP providers in different countries.
The communication server must register with a registrar of the SIP provider so that the
SIP messages can be forwarded to the proxy server and from there to the public network, for example via a gateway. At least one SIP account has to be set up for each
SIP provider. Each account contains a user name and password for identification with
the Registrar and an SIP identification number (SIP-ID). The SIP-ID is linked with a direct dial number so that outgoing and incoming connections can be made. A total of
500 SIP accounts can be set up and linked with the corresponding direct dialling numbers.
One SIP account per SIP provider can be set up as a default account. It can then be
used by users without an SIP account for outgoing calls via a corresponding route or
for incoming calls via a special call routing.
Besides the connection of communication servers to one or more SIP providers, several communication servers can also be networked via SIP.
System configuration
Set the SIP provider parameters using the configuration tool WebAdmin in the Call
routing (=df) view:
• Under Network interfaces click the Add button to create a new SIP provider.
• Double-click the SIP provider you have just created to specify the settings for this
SIP provider. In this overlay view, create and also configure the SIP accounts.
Tips:
– Alternatively you can also configure the SIP provider by importing an SIP provider profile.
For this you must first export the settings of an already configured SIP provider into an XML
configuration file.
– In the Call routing (=df)view you can access a context menu using your secondary
mouse key (right-clicking).
See also:
You can find more information about the procedure and individual parameters on line.
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2. 1. 4Analogue Network Interfaces
The analogue network interfaces support DTMF and pulse dialling1). A range of parameters in the System Configurations allows country-specific adaptations to the public
network as well as other settings.
The analogue network interface settings are only visible in the WebAdminconfiguration
network tool if analogue network interfaces are also actually available. Configuration is
carried out either with the analogue interfaces (=7g) or in call routing (=df).
See also:
You can find more information about the procedure and individual parameters on line.
2. 1. 4. 1Analogue down-circuit connection
With an analogue down-circuit connection the features of the up-circuit communication
server can also be utilized.
This results in the following special applications for the user:
• Depending on the system configuration the user makes phone calls in a complex
environment. The subscriber’s disposal is a large number of features at two levels
(subscriber’s own system and the up-circuit system). A short induction course helps
users to familiarize themselves quickly with the new environment.
• Practically all the systems used as up-circuit systems also feature the DTMF dialling
method on the analogue terminal line, in addition to pulse dialling. It is advisable to
give preference to the MFV / DTMF dialling method over pulse dialling.
• If the up-circuit communication server requires that subscribers wait for the exchange-free tone, all the entered abbreviated dialling numbers must be provided
with a hyphen "-" (interdigit pause) after the digits for exchange access. At this point
the communication server will again pause for the tone when dialling.
1) Pulse dialling in New Zealand is not supported.
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00–– 032 6000 00 00
Exchange seizure
When a number is via Line key dialled, all characters prior to
the followed hyphen are suppressed
Exchange seizure on up-circuit communication server
Waiting for exchange-free tone or dialling after e.g. 6 sec.
(The duration of the interdigit pause can be set.)
DSI
FXS
Up-circuit
communi-
cation
server
S bus
Communi-
cation
server
Example: Exchange access via exchange access prefix
Fig. 16Example of the exchange access prefix via up-circuit communication server
The following configuration steps are necessary:
1. The exchange access prefix of the up-circuit communication server must be entered
in the exchange digit barring (Exchange digit barring).
2. The corresponding analogue trunk lines are set to Behind communication server.
Consequence:
– The external digit barring is deactivated and the exchange digit barring is acti-
vated. The external digit barring of the up-circuit communication server has to be
used.
– Incoming calls are forwarded transparently to the user.
3. The corresponding analogue trunk lines are to be configured to the correct Dial
sort. If the up-circuit communication server provides DTMF and pulse dialling for in-
ternal users, it is advisable to configure DTMF.
Example: Enquiry call behind communication server
This feature can be used from both analogue terminals and system phones.
Fig. 17Enquir y call behind communication server
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Situation: The existing call connection of an MiVoice Office 400 user already seizes a
trunk line to the up-circuit communication server. The procedure for setting up an inquiry call depends on the type of terminal:
• Analogue terminal
– Flash: Dialling tone of the MiVoice Office 400 communication server
– Flash *42: Dialling tone of an up-circuit communication server
• System phones
– Enquiry call menu: Dialling tone of the MiVoice Office 400 communication server
– Key with macro "I*42": Dialling tone of an up-circuit communication server
Using the exchange's features
To activate features on the public network such as the exchange feature "Call Forwarding" from the system itself, you need to seize a trunk line. The feature can then be entered in accordance with the service provider's operating instructions.
2. 1. 4. 2Attenuation on analogue network interfaces
With analogue network connections you have a choice of four different attenuation settings:
• Long or
• Long D for long lines
• Short or
• Short D for short lines
On lines with a loop resistance < 280 Ω Short or Short D should be selected to avoid
problems with echo or instability (feedback).
The "… D" settings are used to increase the volume in an "analogue exchange - digital
terminal" connection type by 3 dB in both directions as this type of connection is generally perceived as too quiet. The reference level is modified accordingly on the expansion card. Due to the restriction to the aforementioned call type the "… D" setting does
not result in an increase if an analogue terminal interface is involved in a connection.
Note:
"… D" setting should not be used (or only once the stability conditions have been thoroughly
clarified) if the equipment (Terminal Adapter) operated on digital interfaces also features a fourwire to two-wire conversion, i. e. an analogue two-wire interface.
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4
ETSI
4
PC
4
ETSI
PINX
PINX
BRI-S
BRI-S
BRI-T
ISDN terminals
BRI-S
ext.
2. 2Terminal interfaces
The communication server supports digital and analogue user-network interfaces.
2. 2. 1Digital user-network interfaces
On each of these digital user-network interfaces several appropriate terminals can be
hooked up and operated simultaneously.
2. 2. 1. 1Terminal interface BRI-S
The S user-network interface is a digital 4-wire interface used for connecting ISDN terminals, Terminal Adapters and ISDN PC cards. Each of these interfaces has two 64
kbit/s user information channels and one 16 kbit/s control and signalling channel
(2B+D). This makes it possible to establish two independent call or data connections
simultaneously.
Fig. 18Terminal interface BRI-S
Up to 8 terminals can be operated on an S user-network interface. They are addressed
with the single-digit terminal selection digit (TSD).
Different modes are available for operating the BRI-S interface (BRI-S protocol in
the interface configuration):
•The ETSI mode is used to operate ISDN terminals, Terminal Adapters and ISDN PC
cards.
• With the BRI-S external mode a BRI-S interface can be used as a basic access for
private networking with QSIG / PSS1 or DSS1. It is no longer available as terminal
interface (see "Basic rate interfaces BRI-S external", page 25
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Format of the ETSI S-bus
Depending on application, the format can be configured on the ETSI S-Bus for each
BRI-S interface in the interface configuration (setting MSN format for BRI-S). More
information can be found in the online help.
Exchange Access Prefix for Terminals on the ETSI S Bus
For terminals on the ETSI S-bus the interface configuration can be used to select
whether or not the exchange access prefix of the CLIP should be truncated for incoming calls (setting Remove exchange access prefix). This setting is effective only in
the S-bus mode (BRI-S protocol = ETSI).
Mitel Advanced Intelligent Network:
In an AIN the call charge format of ISDN terminals depends on the country and is based on the
country configured with the region of the node concerned or user. User allocation takes priority
over a node-specific allocation.
Voice and data terminals on the BRI-S interface
Both voice and data terminals can be connected to the same BRI-S interface. When
designing the system, bear in mind that data terminals can also take up user information channels. ISDN routers and ISDN PC cards that support channel bundling can
take up both user information channels.
In mixed operation the availability of the terminals has to be taken into account.
One call or data connection can be set up on each bearer channel (B channel) simultaneously and independently of the other B channel.
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DSI
DSI
2
2
DECT cordless
phone
Communication server
System phones
DECT radio
unit
2. 2. 1. 2DSI terminal interfaces
The digital terminal interface DSI is a proprietary, system-specific two-wire interface on
which either the AD2 protocol or DASL protocol (Mitel 470 only) is running. The DSI interface is used to connect the following terminals:
•DSI-AD2:
– MiVoice 5300 series system phones (or older AD2 phones)
– Mitel DECT radio units(SB-4+, SB-8, SB-8ANT)
• DSI-DASL: Dialog 4200 series system phones (Mitel 470 only)
Fig. 19DSI terminal interfaces
Two system phones can be connected in parallel to a terminal interface DSI-AD2. Address allocation is done by means of a switch on the phone.
Notes:
– Only one DECT radio unit can be connected for each DSI-AD2 interface. An SB-8/SB-
8ANT radio unit with 8 call channels occupies two DSI interfaces.
– Only one system phone of the Dialog 4200 series can be operated on each DSI-DASL
interface.
The type of protocol on the DSI interface can be chosen in the cards and modules view
(=4g). The setting is always valid for the entire interface card (Mitel 470 only).
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System interfaces
Communication server
Ethernet
IP system
phone
SIP
terminal
Access
point
IP
softphone
WiFi mobile
phone
2. 2. 1. 3IP terminal interface
The IP terminal interfaces are implemented via an Ethernet interface on the communication server. MiVoice Office 400 supports IP system phones, Mitel SIP terminals and
standard SIP terminals from third-party manufacturers.
Fig. 20IP terminal interface
The following IP system phones are supported:
• Terminals of the series MiVoice 5300 IP
• IP softphone MiVoice 2380 IP
• MiVoice 1560 PC Operator
Like the digital system phones the IP system phones (softphones and desk phones)
communicate with the communication server via the AD2 protocol. Unlike digital system phones, however, call and signalling data is transmitted in the IP network. The devices are connected to the IP network. The number of IP terminals on the communication server is determined by the system limits.
The following SIP terminals are supported:
• Terminals of the series Mitel 6700 SIP / 6800 SIP
• Mitel SIP-DECT cordless phones:
• Mitel BluStar 8000i Desktop Media Phone
• Mitel BluStar for PC
• Mitel BluStar for iPhone/iPad
• Mitel BluStar for Android Phone/Tablet
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• Mitel BluStar for Conference Room
• MiVoice Conference Phone
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System interfaces
• Other Mitel or third-party SIP phones and SIP softphones
• WLAN and DECT terminals from Mitel and third-party manufacturers, connected
with the IP network via an access point.
The media data from SIP terminals is processed into packets using the SIP protocol
and transmitted using the RTP protocol. The number of SIP terminals on the communication server is determined by the system limits on the one hand and by the number of
licences on the other. For each SIP terminal operated on MiVoice Office 400, either
one Mitel SIP Terminals licence or one SIP Terminals licence is required.
2. 2. 2Analogue terminal interfaces
This 2-wire interface supports the following off-the-shelf analogue terminals:
• Analogue phones with DTMF or pulse dialling (earth key is not supported)
• Analogue radio units for cordless phones
• Two-wire door intercoms with DTMF control functions
• Group 3 fax machines
• Answering machines
• Modem
No call charges are transmitted to the connected terminals via analogue terminal interfaces.
CLIP display is possible (see Technical Data in the Mitel 415/430 System Manual or
the Mitel 470 System Manual).
One analogue terminal interface per communication server can be configured for connecting a general bell.
1)
1) Transmission with the T.38 protocol is recommended for Fax over IP.
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System interfaces
FXS
FXS
FXS
FXS
FXS
Communication server
Analogue phone
DTMF/Pulse dialling
Group 3 fax machines
Answering machines
Modem
General bell
Door intercom with DTMF
control functions
Fig. 21FXS terminal interface
2. 3Special Interfaces
The system supports a range of special interfaces.
2. 3. 1Ethernet interfaces
The Ethernet interface on the basic system is available for the following purposes:
• data exchange with WebAdmin
• signalling and transmitting voice data (VoIP) in an Mitel Advanced Intelligent Network (AIN)
• linking up the Mitel Open Interfaces Platform (OIP)
• the connection of a CTI, alarm, ATAS or messaging server, etc.
• the connection of IP system phones
• the connection of SIP terminals (softphones or desk phones)
• The connection to one or more SIP providers.
• Networking MiVoice Office 400 communication servers via SIP, etc.
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System interfaces
2. 3. 2Interface for door intercom system
There are different ways of connecting door intercom systems:
• Using an options card ODAB (only Mitel 415/430)
• Using an ordinary analogue terminal port
In a connection using an options card, the equipment or installation is controlled via relays and a control input on the options card.
In a connection using an analogue terminal port the TFE must be capable of sending
and receiving DTMF signals as the control is effected acoustically via a speech path.
A bell key is backed by an internal destination. The door intercom system can be addressed via an internal number.
A loudspeaker system can also be operated via the interface for door intercom systems.
See also:
"Special interfaces" Chapter in the Mitel 415/430 or Mitel 470. System Manual
2. 3. 3Interface for General Bell
Calls can also be routed to the general bell. Bells or lamps connected to the general
bell interface signal calls which can be answered by anyone from any user’s phone.
Coded ringing can be used to assign different ringing patterns to different destina-
tion persons or groups and, in this way, create a simple type of paging system.
Tip:
One analogue terminal interface per communication server can be reconfigured in such a way
that it is also used for connecting a general bell. This eliminates the need for an external ringing
voltage source.
See also:
"Special interfaces" Chapter in the Mitel 415/430 or Mitel 470. System Manual
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Numbering plan
3,1;3,1;
(
313
313
313
3,1;
KD]DD[[D
3,1;
313
3Numbering plan
This Chapter features the different types of internal and external numbering plans
available in the various systems. It explains the differences between internal numbering plans for the private network and external numbering plans for the public
network. It tells you what you need to know when creating numbering plans for
each particular network.
3. 1Numbering Plan Identifiers
The numbering plan is used to analyse numbers and allocate them to an addressable
destination. Two types of numbering plans (Numbering Plan Identification, NPI) are relevant to the system:
• The public network uses numbering plan identifier E.164, which is defined and
standardized by the ITU-T.
• Private networks use numbering plan identifier PNP (Private Numbering Plan). The
internal numbering plan of a communication server or PINX is also of the PNP type,
as is the private numbering plan supplied by the public network provider.
PINX 3 is a virtual PINX (Centrex)
Fig. 22Numbering plan identifiers in the public network and in the PISN (in PINXs)
1) depends on the network provider
1)
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Numbering plan
Numbers in a numbering plan are analysed with the aid of the Type Of Number (or
TON).
Numbering Plan Identifier E.164
Numbering plan E.164 comprises the following types of number:
Tab. 5E.164 types of number
Typ e Of N u mbe rStructureExample
Subscriber[SN]624 11 11
National[NDC][SN]32 624 11 11
International[CC][NDC][SN]41 32 624 11 11
Unknown[NP][NDC][SN]032 624 11 11
[IP][CC][NDC][SN]0041 32 624 11 11
[SN]Subscriber Number (user number)
[NDC]National Destination Code (national destination code or toll area code)
[CC]Country Code (country code)
[NP]National Prefix (national prefix)
[IP]International Prefix (international prefix)
The national and international prefixes (in Switzerland 0 for national and 00 for international long-distance traffic) are not part of the type of number. Prefix digits are sometimes also referred to as trunk prefixes.
PNP Numbering Plan Identifier
The PNP numbering plan comprises the following types of number:
Tab . 6PNP typ es o f n umbe r
Typ e Of N u mbe rStructureExample
Level 0[RIN] 1313
Level 1[RP1] [RIN] 60 1313
1)
Level 2
1)
The system supports private networks up to and with Level 1
[RIN]Regional Intern Number: all destination numbers within a Level 0 region
[RP1]Regional Prefix 1: Prefix for a Level 1 region
[RP2]Regional Prefix 2: Prefix for a Level 2 region
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Numbering plan
-FWFM
-FWFM
-FWFM
-FWFM
Fig. 23Levels as per PNP definition
-FWFM
3. 2The System's Numbering Plan
The system's internal numbering plan is the numbering plan used for a stand-alone
communication server or a PINX in a private network. The numbers entered in the
numbering plan are used both to dial up call destinations in the communication server
and to execute control functions. Call destinations and functions are grouped into categories.
The internal numbering plan:
• Assigns number ranges to the categories.
• Allocates their numbers to call destinations and control functions, making them obtainable and executable respectively.
As far as the call destination numbers are concerned, the system's numbering plan is a
PNP-type numbering plan.
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Numbering plan
3. 2. 1Categories in the Numbering Plan
The allocation of categories to numbers and number ranges can be freely configured,
provided a number of rules are observed. The default settings depend on the country.
Configuration
Open the numbering plan configuration directly with (=g4).
Alternatively, in the header of the call routing view (=df) click the numbering plan
link.
Rules for the internal numbering plan
• Numbers are always interpreted starting from the left.
• The various categories must be unequivocally separated through number allocation.
If, for example, the operator console has been allocated number 11, the numbers
11n cannot be allocated to any other categories. If, however, the operator console
has been allocated the number 111, the numbers 112 to 119 can be allocated other
categories.
• Numbers within a category do not necessarily have to constitute a coherent range;
instead, they can be spread over the entire number range (e.g.: user 200, 404, 550,
551, ...). However, for the purposes of clarity, we recommend that you define coherent ranges.
• The number length is variable and may consist of 1 to 12 digits. Numbers with more
than 12 digits will be truncated from the right.
Tab. 7Categories in the system's numbering plan with allocated numbers
CategoryNumber / Number Range
NameExplanationNumber1) Number rangeExplanation
Exchange access,
business
Exchange access, private
Operator phoneThe PC operator console can
Emergency numberThis call number is assigned
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Call charges are added up on
the Business Telephony or
Business Data Service cumulative counter.
Call charges are added up on
the Private cumulative counter.
be reached under this call
number.
global emergency destination
1 under which three destination numbers are stored
(depending on the switch
group and switch position).
0<ext. call No.>Prefix, truncated before
dialling out into the network
10<ext. call No.>Prefix, truncated before
dialling out into the network
11–
12–Up to 10 emergency
numbers can be
defined, all of them
assigned global emergency number destination 1.
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Numbering plan
CategoryNumber / Number Range
NameExplanationNumber1) Number rangeExplanation
Cost centre selectionThe call charges are explicitly
allocated to the selected cost
centre.
UsersCall numbers of the internal
users The users are assigned
one or more terminals.
Route selectionRoutes the outgoing call via
the selected route
User groupThese call numbers can be
used to dial user groups internally
MMCCInternal number of the MMC
Controller.
Remote maintenance
access PPP
Selects the configuration interface via PPP
Voice mailInternal call number of the
standard voice mail system. To
activate a mailbox a call is
rerouted to this call number.
Abbreviated diallingOther, user-definable call
numbers are stored under
these numbers
Door intercom system
4)
Control output
4)
Selects the door intercom851–
Freely configurable control
outputs for switching external
equipment
Call distribution element
Call distribution elements link
direct dialling numbers with
internal call numbers. A call
distribution element can be
assigned its own internal call
number.
PISN usersa) Internal users in a net-
worked system.
b) External users which are
displayed as internal users
and can be dialled as internal
users.
Own region prefixLevel 1 prefix for the region
allocation of a PINX in the
PISN
* - substituteSubstitute digit for pulse dial-
ling phones without * key
13<CC No.>
<ext. call No.>
Prefix, truncated
together with the CC
No. before dialling out
into the network
20 to n or 200
2)
to n
170 to n
–
3)
<ext. call No.>Prefix, truncated before
dialling out into the network
3)
860 to n
–
897–
898–
899–Only one voice mail no.
can be created.
7000 to 7999 –
853 to 856–
Not allocated –
Not allocated –
Not allocated –Prefix, truncated on
detection
Not allocated <Function code>
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1)
Default settings for Switzerland
2)
Depends on the number of terminal interfaces installed.
3)
Depends on the type of communication server
4)
Only with Mitel 415/430 and if the corresponding number of ODAB card(s) is fitted
3. 2. 2Exchange Access Categories
A call can be transmitted to the public network by selecting a prefix from one of the exchange access categories.
The cost type (Business, Private), cost centre (cost centre selection) or route (route selection) is determined according to the prefix selected.
Route selection prefixes are the internal call numbers of the routes.
Route selection can also be used for routing in the private leased-line network.
3. 2. 3Category for abbreviated dialling
Abbreviated dialling numbers facilitate the exchange traffic for numbers that are frequently used. They can also be used to activate functions via */# function codes more
quickly.
An internal or external call number or a function code and a name can be stored under
any abbreviated dialling number
Stored Numbers
If an external number is stored, the exchange access prefix must also be entered at the
same time. Prefix and number must be separated with a hyphen. The hyphen ensures
that when the number is dialled via a line key, the exchange access prefix is truncated.
Only the front portion of a number can be entered at any time. The rear portion must
then be suffix-dialled manually. Example:
The number 0-001212 and the name "NY" (for New York) are stored under the abbreviated dialling number 7500. Any user who wants to call Manhattan, New York, simply dials "NY" by name, then adds the local number.
Mitel Advanced Intelligent Network:
In an AIN with nodes in different countries the abbreviated dialling numbers must always
include the international prefix (e.g. 00) and the country code (e.g. 41). (Example:
0-0041326553333). This is necessary as the national portion of the number may well be identical in different countries. This prevents conflicts in the call routing and call number display
(CLIP).
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Name
The name is used:
• To dial by entering the name rather than the call number (dialling by name).
• To display the name on the user’s own system phone when the CLIP number of an
incoming call matches the number stored under abbreviated dialling (see "Replicating the Name Display in the Communication Server", page 77).
Digit Barrings and Exchange Access Rights
When an external destination is dialled via an abbreviated dialling number the number
stored bypasses the digit barring and the exchange access authorization.
When an external destination is dialled using dialling by name via abbreviated dialling,
only the exchange access rights are bypassed (more on digit barrings and exchange
access rights see "Digit barring", page 192
and "Exchange access authorization",
page 203).
3. 2. 4Category for emergency number
A total of 10 emergency numbers can be created in the numbering plan. The emergency numbers are used to quickly dial an emergency number destination.
All emergency call numbers dial global emergency destination 1. In the Emergency
destination view (=9r) configure 1 switch group and 3 destination numbers for emer-
gency call destination 1. When an emergency call number is dialled one of the three
destination numbers is dialled based on the switch position of the assigned switch
group.
Notes:
– When an external destination is dialled via the emergency number the digit barring and the
exchange access authorisations are bypassed.
– 49 emergency destinations can be defined in addition to the global emergency destination.
They are meant for individual assignment to terminals. If a terminal is assigned such an
emergency destination, one of the destination numbers of this emergency destination is
dialled while dialling an emergency call number, depending on the switch position of the
assigned switch group. Therefore, the emergency destination of the terminal has priority
over the global emergency destination.
Mitel Advanced Intelligent Network:
In an AIN the nodes can be located in different countries, which means it makes sense to enter
in the numbering plan the emergency number normally used in each country. Depending on the
assigned emergency destination and the switch position of the configured switch group the corresponding destination number is then dialled whenever the emergency number is dialled. The
assignment of the emergency number destination is configured for each node.
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3. 2. 5Category for users
3. 2. 5. 1Internal users
The call numbers within this category are assigned one or more terminals. The following terminal types are supported:
• IP system phones (hardphones and softphones)
• Digital system phones (DSI / DASL)
• Cordless DECT phones (system phones or GAP phones)
• Analogue terminals
• SIP system phones (Mitel SIP)
• SIP phones or SIP terminals by other manufacturers (standard SIP)
• BluStar phones (hardphones and softphones)
• Integrated mobile/external phones (with assigned internal call numbers for incoming
and outgoing calls; see "Integrating mobile and external phones", page 61
• Integrated mobile phones with user-friendly Mitel Mobile Client application (linked to
the communication server via an MMC Controller).
• Virtual phones (behave like analogue internal phones, except that they do not occupy any port since no hardware is available; see "Virtual terminals", page 65
• ISDN terminals on the BRI-S bus (phones, PC cards)
.)
.)
If an internal user is assigned a name, the user in question can be dialled internally by
entering the name instead of the call number (dialling by name), and the name is displayed on the destination user’s terminal on the system’s own communication server,
or another PINX in the PISN (CNIP).
3. 2. 5. 2Integrating mobile and external phones
Although the communication server does not have a GSM receiver, mobile phones can
be connected to MiVoice Office 400. The mobile phone is assigned to a user and can
be reached internally using his user number. If the mobile phone user dials a call number specially set up in the communication server, he can execute certain functions via
*/# function codes or make internal/external calls. This feature is not limited to mobile
phones, but can also be used for other external phones.
An in-depth integration is achieved through the Mitel Mobile Client application installed
on a mobile phone, as well as with the help of an MMC Controller connected to the
communication server. The Mitel Mobile Client allows the most important telephony
functions to be used conveniently, whereas the MMC Controller enables mobile users,
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for instance, to move back and forth between the internal WLAN coverage and the mobile radio network without the call being interrupted.
Integration step 1
• The mobile or external phone is assigned to a user and can be reached internally
using his user number.
• If the integrated phone user is assigned a direct dialling number, he can also be
reached from the outside.
• The status of assigned user is monitored and displayed internally (e.g. on team
keys). This is of course possible only for actively logged in phones or for calls to integrated phones set up via the internal user number.
• If the user of the integrated phone calls an internal user on his direct dialling number, the called party is shown the CLIP of the integrated mobile phone’s internal call
number.
• The external user of the integrated phone can dial in using specially set-up direct dialling numbers for which Mobile/external phone integration is configured as the
CDE destination; once the external user has been authenticated he obtains the internal dialling tone. He can then carry out specific functions via */# function codes in
prefix dialling or make internal/external calls. Several such direct dialling numbers
can be set up for each communication server or AIN. This can help to save considerable roaming charges in an AIN that covers several countries.
• One Mobile or External Phone Extension licence is required for each integrated
phone.
Integration step 2
The integration level 2 has also all functions of the level 1. Suffix dialling functions such
as enquiry calls or setting up a conference are additionally possible. This requires special DTMF receivers which must be activated throughout the connection. This in turn
requires DSP resources. This means that the following prerequisites are needed so
that the functions of integration step 2 can be used:
• The number of DTMF receivers required must be covered with GSM channels in the
DSP configuration (DSP (=ym) view). The number of assignable GSM channels
differs depending on the configuration server and DSP (see Mitel 415/430 and
Mitel 470 System Manual).
• If all GSM channels are busy, the functions of integration level 2 for the current call
connection are not available.
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• The enhanced functionality must be assigned to each integrated mobile or external
phone in the terminal configuration (Enhanced functionality parameter).
• Provider-specific case: If the integration of the mobile phone is obtained with separate lines to the provider, the parameter Allow enhanced functionality for direct in-
coming calls must be activated in the corresponding trunk group.
Mitel Advanced Intelligent Network:
The DSP resources must be made available at the node through whose network interface there
is a communication server − mobile/external phone connection.
See also:
An overview of the function codes supported at integration levels 1 and 2 can be found in the
"Mobile Phones on MiVoice Office 400" User’s Guide.
Automatic mobile/external phone authentication
The integrated phone can be automatically authenticated using the CLIP, and the user
hears the internal dialling tone after a ring back tone (Parameter Use CLIP for au-
thentication).
Note:
– For security reasons automatic authentication is not used with "Break-in" or "Special
Arrangement" situations as the incoming CLIP is not PSTN-verified in such cases. There
may be cases however (especially with SIP providers) where the CLIP is received as "verified" when in fact it is not. An unauthorized person can then dial into the communication
server and make calls or carry out certain */# procedures. After a first start automatic
authentication is switched off.
– In the case of a connection via analogue or SIP network interfaces the CLIP is normally
received "unchecked". To allow automatic authentication of the integrated phones nonetheless, the parameter Allow CLIP authentication even if CLIP is not screened must be activated in the corresponding trunk group (default setting = deactivated).
Manual mobile/external phone authentication
If automatic authentication is disabled, the integrated phone is manually authenticated
as follows:
1. The user of the integrated phone dials a direct dialling number specially set up.
A ring-back tone is obtained followed by a special authentication tone.
2. The user makes the following entries: <Internal user number> * <user PIN> #
The internal dial tone is obtained.
3. The user of the integrated phone can now make an internal/external call or execute
certain functions via */# function codes.
Note:
For both automatic and manual authentication the user PIN must be changed first. The default
value "0000" is not permitted.
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System configuration
The mobile/external phones are configured with the mobile/external terminal interface
settings. The navigation code =32 opens the first terminal of this type in the terminal
list. Refer to the online help for information on the individual settings.
Note:
In the case of an external call to an integrated mobile/external phone the caller’s CLIP is always
transmitted to the phone as redirecting information. This also applies to external calls to a user
who has redirected to an integrated phone. In this case the parameter Send redirecting informa-
tion must be activated in the trunk group settings and "Special Arrangement" must be activated
by the network provider.
See also:
A separate User’s Guide is available for the mobile phones on MiVoice Office 400. It includes
an overview of the functions that can be carried out using mobile phones.
3. 2. 5. 3Mitel Mobile Client / Mitel Mobile Client Controller
Mitel Mobile Client 4 is an application for the most common Smartphones with Android, iOS and BlackBerry operating systems. The most important telephony functions
are thus available via the menu, and the mobile phone user can move about freely between the WLAN and mobile network without the call being interrupted (seamless
handover).
The server-based Mitel Mobile Client Controller, connected to the communication system via SIP, is required for integration into the communication system. As a configuration and administration interface, it offers web-based management software. However,
Mitel Mobile Client Controller and Mitel Mobile Client 4 are normally configured automatically via WebAdmin.
Dual mode (GSM/ WLAN), address book search, quick call setup, Voice over IP and
"trueCLIP" are possible using the 2G/3G data channel between Mitel Mobile Client 4
and Mitel Mobile Client Controller. Mitel Mobile Client Controller is connected to the internet; therefore, a direct internet connection is not required for the communication
system.
Each Mitel Mobile Client 4 requires one MMC Extension licence.
See also:
With Mitel Mobile Client 4, a user's guide and and an administration manual are available for
each operating system.
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3. 2. 5. 4Virtual terminals
Virtual terminals respond in the same way as analogue internal terminals except that
they
• do not physically occupy a port as there is no hardware involved,
• do not require a B channel.
Other properties
• Virtual terminals are capable of sending and receiving messages via the third-party
CTI interface.
• A user who has been assigned only one virtual terminal is referred to as a virtual
user.
• When the caller dials a virtual user he obtains the ring-back tone or the busy tone (if
the user is already in a call).
• Virtual users belong to the group of users with their own DDI number, the maximum
number of which is restricted by the system limits per system.
• Virtual users have their own recall time, which can be set throughout the system. It
is used if no recall time is defined in the user setting (see also "Recall", page 371
Application examples:
• During an explicit call transfer without prior notice to a virtual user a call can be
parked for up to 900 seconds and then transferred using *86 User No.
• To integrate a PISN user into a user group, it is possible to accept a virtual user in
the user group using a CFNR to the PISN user.
• In third-party CTI applications virtual users can be used to send and receive messages.
).
3. 2. 6PISN users
This category comprises users who belong to the same private integrated services network (PISN) but are connected to a different PINX. They can also be users of a virtual
PINX.
The numbers of user groups, call distribution elements, abbreviated dialling destinations, routes or door intercoms can also be entered as PISN users, besides the numbers of internal users.
The configuration takes place in the PISN user(=gv) view.
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Numbering plan
201
202
203
A
B
C
216
217
218
D
E
F
201
202
203
21X
218
216
217
218
201
202
203
Internal user D
Internal user E
Internal user F
PISN user A
PISN user B
PISN user C
Internal user A
Internal user B
Internal user C
PISN users D, E
PISN user F
Numbering plan PINX 2
Numbering plan PINX 1
PINX 2
PINX 1
Entering PISN Users
There are two ways of entering PISN users:
• A PISN user’s call number is entered in full and unequivocally (Fig. 24, PINX 2).
• One number with wildcards is entered for several PISN users (group of PISN users,
Fig. 24
, PINX 1, PISN users D and E).
These variants can also be combined (Fig. 24
Fig. 24PISN users entered with and without wildcards
Entering the Number of a PISN User in Full
A complete PISN user number unequivocally identifies a user at another PINX or a virtual user.
Each unequivocal number of a PISN user can be allocated a name in the user configu-
, PINX 1).
ration. This enables:
• these users to be dialled by entering the name rather than the call number (dialling
by name)
• the name of a virtual PISN user to be displayed (CNIP)
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Entering Wildcards for a Group of PISN Users
A number with wildcards identifies a group of PISN users (Fig. 24
be:
• the internal users of one or more PINXs
• the PISN users of another region
The wildcard is entered as an upper case (e.g. 21X).
This method of entering PISN users helps to reduce the number of entries made.
Moreover, not all the changes made to the internal users of a PINX need to be updated
in the other PINXs. However, neither the call numbers nor the names of the individual
users in the group are stored in a phone book (it is not possible to retrieve the number
from a phone book nor is dialling by name possible, except if the number and name are
also stored locally in a private phone book).
Tip:
It is advisable to enter PISN users first with wildcards in an initial stage so that the numbering
plan is quickly and transparently available throughout the PISN, and is also already operational.
All the PISN users to be available using dialling by name can then be entered individually at a
later stage
Entering a Regional Prefix
If an individual or group entry belongs to another PISN region, the entry for the PISN
user must be preceded by the regional prefix.
, PINX 1). They can
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200
200
299
399
299
399
300
300
PISN
Region 60
Region 50
PINX 4
PINX 3
PINX 2
PINX 1
Example of Entering PISN Users
...
...
...
...
Fig. 25PISN with two regions
Tab. 8Entering PISN users in PINX 2
Varia nt
Number in full300200,201...29960200, 60201...6029960300, 60301...60399
Numbers partly with wildcards
Numbers with maximum
possible wildcards
Combination:
number in full and number
with wildcards
Number
of entries
1220X, 21X...29X602XX603XX
22XX60XXX
52XX, 21160XXX, 60211, 60311
PINX 1PINX 3PINX 4
PINX3 and PINX 4
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50
200 ...299
400 ...499
50
400 ...499
200 ...299
Internal users
PISN users
Internal users
PISN users
Own prex
Own prex
Shared Numbering Plan
Numbering plan PINX 2
Numbering plan PINX 1
PINX 2PINX 1
Region 50
3. 2. 7Separate Regional Prefix Category
This regional prefix allocates a PINX to a PISN region.
The PINX compares its own regional prefix entry with the first few digits of the call numbers of the following calls:
• All outgoing calls
• All incoming calls routed via a trunk group with the setting Network type = Private
If the first few digits match up with the PINX's own regional prefix, they will be truncated. The remaining number is then analysed and forwarded
3. 2. 8Shared Numbering Plan
PISN users are structured in the internal numbering plans of the PINX.
From the PINX’s viewpoint its own users are internal users and the users of the other
PINXs are PISN users.
If two or more PINXs are structured in such a way that they split the users’ number
range among themselves, we talk of a shared numbering plan. Together they form a
region, within which all users can be reached under the internal call number.
Fig. 26Shared numbering plan: two PINXs share the numbers of a numbering plan.
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Numbering plan
50
200
2300
60200
...299
...399
...60299
...399
...299
...60299
50
300
200
60200
60
200
50200
...299
...50399
Internal users
PISN users
Own prex
Internal users
PISN users
Internal users
PISN users
Own prex
Own prex
PINX 3PINX 2PINX 1
Shared Numbering Plan
Numbering plan PINX 3
Numbering plan PINX 2
Numbering plan PINX 1
Region 60Region 50
3. 2. 9PISN with different Regions
A PISN can be divided into several regions. Each region is identified by its regional
prefix.
Users who call a user in a different region first dial the prefix of the destination region,
then the internal number of the user they want.
Their specific regional prefix is specified in the internal numbering plan (=g4) of each
PINX.
The organization of the numbering plan does not depend on the PISN topology.
Fig. 27PISN with 2 regions and shared numbering plan for Region 50
Entering a Regional Prefix
In the example above the PISN users of a different region are entered with the regional
prefix (for example 60200 to 60299)
Another possibility is to define a route with call number 60 and to enter the PISN users
without regional prefix (route method).
The user dials exactly the same number, for example 60250, but this time the call is
routed as a route selection. It uses the route with call number 60 and not the one allocated to the PISN user in the user configuration. (In the example above the numbers
would have to be distributed differently since number ranges cannot be assigned
twice.)
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Single ringing tone
Time
Time
Double ringing tone
4Identification elements
Correctly identifying and displaying a call is the essential requirement for adequately
implementing the system's networking philosophy. This Chapter looks at how the
origin of a call is identified using different ringing tone patterns and how the caller's
number (CLIP) or name (CNIP) is displayed. It describes how CLIP and CNIP displays
are created under different system conditions, how they can be influenced, and how
to suppress the CLIP display.
A call is identified firstly by the type of acoustic ringing (i.e. ringing pattern) and, secondly, by the display on the terminal.
The default values are selected in such a way that the ringing patterns and displays
appear correctly in most cases. Changes to the settings are necessary only in exceptional cases.
4. 1Internal and External Ringing Patterns
The ringing pattern provides a means of identifying whether the call originates from
within the PBX (internal call) or from the outside (external call). The rhythm of the ringing pattern differs in each case
Fig. 28Single ringing tone
Fig. 29Double ringing tone
1) The way in which ringing patterns are assigned to internal and external calls varies from one country
to the next.
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1)
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Calls with the Internal ringing pattern:
• Calls from internal users
• Calls from the public network to analogue terminals if the interface configuration
(=7g) of the parameter Ring pattern is set to Single ringing tone.
• Calls from users from the private network (PISN users):
– Calls from the private leased-line network
– Calls from virtual network PISN users
An enquiry call from a user with an exchange call on hold if in the signalling settings (=nr)
•
the parameter Ringing pattern at enquiry destination is set to Internal ring melody.
Calls with the external ringing pattern:
• Call from the public network
•
An enquiry call from a user with an exchange call on hold if in the signalling settings (=nr)
the parameter Ringing pattern at enquiry destination is set to Internal ring melody.
The Ringing pattern at enquiry destination setting is valid throughout the system.
Note:
Certain terminals which automatically answer calls (e.g. fax machines) are not able to interpret the
double ringing tone correctly. With these terminals the configuration Ring pattern = Single ring-
ing tone can be used to force a situation where the single ringing tone is always used for all calls.
Alternative for the MiVoice 5300, MiVoice 5300 IP series of system phones and the
MiVoice 2380 IP IP softphone
Different ringing melodies can be configured separately in the terminal configuration for
each system phone to help differentiate between internal and external calls. If the parameter External ring melody is deactivated, the single and double ringing tone is
used to make the distinction; otherwise, the configured ringing melodies. If no distinction is required, the identical melody can be entered in both places.
Identifying the Origin of a Call
If an incoming call's CLIP number corresponds to numbering plan identifier E.164, the
system assumes that the call comes from the public network.
If an incoming call's CLIP number corresponds to numbering plan identifier PNP, the
system assumes that the call comes from the PISN.
If the CLIP number’s numbering plan identifier is (Unknown), the trunk group configuration is used to decide whether the call is signalled internally or externally (NPI call
unknown setting).
See also:
"Numbering Plan Identifiers", page 54
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%$
&/,3
&1,3 7(GLVRQ
,1HZWRQ
KD]DD[[D
%$
7(GLVRQ
&/,3
&1,3 7(GLVRQ
&2/3
&213 ,1HZWRQ
%$
7(GLVRQ
,1HZWRQ
KD]DD[[D
4. 2Displaying Numbers (CLIP) and Names (CNIP)
During both the ringing phase and the call itself the caller’s call number or name (or
both) are shown on the terminal’s display.
• The indication of the caller’s phone number is referred to as CLIP (Calling Line Identification Presentation).
• The indication of the caller’s name is referred to as CNIP (Calling Name Identification Presentation).
Fig. 30CLIP and CNIP
When the destination user answers the call, the number or name of the destination
user is transmitted and displayed to the caller:
• The indication of the number is referred to as COLP (Connected Line Presentation).
• The indication of the name is referred to as CONP (Connected Name Presentation).
Fig. 31COLP and CONP
These identification elements allow the use of other features such as logging unanswered calls on the destination user’s call log; the destination user can then return the
call by dialling the CLIP number.
These identification elements are available in digital networks and in some analogue
networks. As CNIP and CONP are not supported by the public network, the system
tries to replicate them by searching through the internal phone books for a number that
matches the CLIP or COLP number. If there is a match, the name entered there is displayed (see "Replicating the Name Display in the Communication Server", page 77
).
CNIP and CONP are supported in the private network under QSIG. They are both accepted and do not need to be recreated in the communication server.
The CLIP and COLP numbers also contain the information of the NPI numbering plan
type and the TON Type of Number (see "Numbering Plan Identifiers", page 54
).
The system needs this additional information for a correct number analysis, particularly
as a PINX in a PISN. It is not displayed on the user’s terminal.
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Identification elements
D
C
B
A
Communication
server
C’s CLIP
A’s CLIP
Note:
For CLIP display on analogue terminals the following conditions have to be met:
– In the interface configuration (=7g) the parameter Terminal supports call identification
must be activated.
– The terminal must support CLIP display.
– Restriction for Mitel 415/430: Different CLIPs can only be sent to 2 analogue terminals
simultaneously.
CLIP Numbers Outside the Registered Number Range
Sometimes the CLIP number transmitted to the public network is not within the registered number range Network providers have different ways of responding to this situation:
• The network provider uses the PINX master number as the CLIP number and sends
it on to the destination user.
• The network provider sends the CLIP number received, on to the destination user.
Usually this requires an agreement with the network provider (special arrangement).
In the following cases a PINX sends the CLIP outside the registered number range:
• If a freephone number (0800...) is to be displayed as the CLIP
• In the case of overflow routing via a different gateway PINX (see page 246
ample in Tab. 16).
• In the case of break-out routing (see page 251
)
• If a break-in situation is to be forced
and ex-
4. 2. 1Displaying the CLIP
CLIP functions process incoming and outgoing calls.
Fig. 32CLIP of an incoming and an outgoing call
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CLIP of an Incoming Call
User A calls user B:
User A sends his CLIP, which is received in the communication server by the trunk
group, processed and displayed to user B.
For more details see as of page 80
.
CLIP of an Outgoing Call
User C calls user D:
User C sends his CLIP number, which is processed in the communication server. If
there already is a direct dialling and a corresponding allocation, the CLIP number is
adapted and sent to user D.
For more details see as of page 80
.
The default configuration has been selected so that the CLIP display is correct. The
relevant settings do not normally have to be adjusted.
4. 3CLIP with Incoming Calls
The CLIP number of an incoming call is processed and presented in two stages:
• Analysis and processing of the CLIP number
• Presentation of the CLIP number on the destination user’s terminal
4. 3. 1Analysing and Editing the CLIP
The following information is necessary for specifying the CLIP properties in a PISN correctly. This sub-chapter can be skipped in the case of the configuration of a standalone communication server.
The system analyses and adapts the CLIP number of an incoming call as accurately as
possible so that the CLIP number is always displayed correctly, even in a PISN.
For this purpose CLIP number prefixes such as regional prefix, prefix and code are
evaluated, and the type of number adapted.
The tables below show how the system handles the type of number and the CLIP number of an incoming call.
Tab. 9Handling a CLIP number with NPI-type PNP or Unknown
Type of number (TON)Own region prefix
Unknown, Level 1, Level 2 yesRegional prefix is truncated,
noCLIP number and TON remain unchanged
Level 0noCLIP number and TON remain unchanged
1)
CLIP number has a regional prefix that matches the separate PINX.
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Conversion
TON is set to Level 0.
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Tab. 10Handling a CLIP number with NPI-type E.164
Type of number
(TON)
UnknownInternational prefixPrefix is truncated,
InternationalCountry code that matches the sepa-
NationalLong-distance code that matches the
SubscriberCLIP number and TON remain unchanged
See also the examples in "Examples of CLIP Displays in the PISN", page 93.
PrefixConversion
TON is set to International,
Further processing, see TON = International
National prefixPrefix is truncated,
No prefixCLIP number and TON remain unchanged
rate PINX
No matching country codeCLIP number and TON remain unchanged
separate PINX
No matching country codeCLIP number and TON remain unchanged
TON is set to National
Further processing, see TON = National
Code is truncated,
TON is set to National
Further processing, see TON = National
Code is truncated,
TON is set to Subscriber.
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4. 3. 2Presentation of the CLIP on the Terminal
Call from the Public Network
If a call originates from the public network, the prefix for Exchange access, business
followed by a hyphen is added to the CLIP number (e.g. 0-333 33 33) so that the called
party can call back simply by dialling the number displayed.
Call from a PISN User in a Virtual Network
If a call originates from a PISN user in a virtual network, the call number to the PISN
user is used to convert the CLIP number into the PISN user number and NPI is set to
PNP (see also examples on page 101
Destination is not a system phone
If the destination is not a system phone; the CLIP number is handled in the same way
as with system phones but without adding a hyphen.
Calls with suppressed CLIP (CLIR)
If a caller uses the CLIR function to suppress his CLIP display to the called party, the
system phone displays Number suppressed instead of the CLIP.
Calls without CLIP
Number unknown is displayed on the system phone for calls without CLIP.
).
4. 3. 3Replicating the Name Display in the Communication
Server
The communication server will try to assign a name to the CLIP number of an incoming
call from the public network and to display that name on the system phone (CNIP). A
search is therefore carried out in the communication server directories for a match for
the CLIP number. The directories are searched in the following sequence:
• PISN user list
• Abbreviated dialling list
• Local directories of the system phones
A name will be displayed depending on the search result as shown in Fig. 33
CNIP and CONP are supported in the private leased-line network under QSIG. They
are both accepted and do not need to be recreated in the communication server.
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Identification elements
•
•
E.164
[
2
]
[
1
]
[
2
]
[
2
]
[
2
]
[
2
]
[
2
]
= E. 164 ?
CNIP is created by
complementing the
internal CLIP No. with the
stored name
Does
the CLIP No.
match an abbreviated
dialled
number
?
NPI of
the CLIP No.
CLIP No. is forwarded
to the terminal
without the CNIP
CNIP is created by
complementing the
internal CLIP No. with the
stored name
no
no
no
yes
yes
to the terminal
to the terminal
to the terminal
to the terminal
CLIP is forwarded
to the terminal
wihout CNIP
CLIP No. is forwarded
to the terminal
without the CNIP
to the terminal
to the terminal
CNIP is created by
complementing the
internal PISN no. with the
stored name
NPI of the CLIP No. is set to
unknown
yes
yes
no
Does
the CLIP No.
match the stored call
number of a PISN
user block
?
Does
the CLIP No.
match the stored call
number of a
PISN user
?
Does
the CLIP No.
match an abbreviated
dialled number
?
Does
the CLIP No.
match a PISN
No.?
no
no
no
Trunk
group settings
“Type of
network = ?”
NPI of the CLIP No.?
Unknown
Public
Private
Private
Matching digits are
removed
Matching digits are
removed. NPI of CLIP
No. is determined
no
yes
Do
the rst
digits of the CLIP No.
match an own
prex?
Call with CNIP
Call without CNIP
Do
the rst
digits of the CLIP No.
match the digits under
"Truncate
CLIP"?
yes
yes
yes
4. 3. 4Flow charts for name identification (CNIP)
[1] Possible prefixes: own prefix, country code, area code or own regional prefix.
[2] Continues on Fig. 34
Fig. 33Analysis and processing of an incoming call in the communication server
78
.
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No. in call
list?
No. on
the key?
No. in
phone book?
System
generates CNIP?
System phone
No CNIP
CNIP = name from call list
CNIP = name on
the key
CNIP = name in
phone book
no
no
no
yes
yes
yes
no CNIP
CNIP = unchanged
no
yes
ISDN terminal
which
terminal?
CLIP/CNIP from communication server[1]
Identification elements
[1]From Fig. 33.
Fig. 34Presentation of the CLIP / CNIP of an incoming call on the terminal
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4. 4CLIP with Outgoing Calls
With an outgoing call the CLIP number is transmitted along with the NPI and TON information. In principle there are two possible variants for creating a CLIP number:
• The communication server creates the CLIP number automatically, based on the origin and routing of the call.
• A number is entered permanently as the CLIP number in the user configuration.
4. 4. 1Creating the CLIP in the communication server
If in the user configuration the setting Create CLIP number automatically is activated, the communication server generates a CLIP number. If there is a suitable DDI
number for the calling user, that number will be used.
A suitable DDI number is a number in a direct dialling plan which
• is linked directly or through a user group to the calling user via a call distribution element, and
• is linked with the same trunk group via which the outgoing call is routed.
If there is more than one suitable DDI number, the lowest one is used.
The trunk group settings are used as the numbering plan identifier and type of number.
If there is no suitable DDI number, the trunk group settings are used for calls into the
public network (Fig. 35), for calls into the private leased-line network it also depends on
how the automatic CLIP is set in the trunk group configuration (=bg) (Fig. 37
).
4. 4. 2Entering a fixed CLIP
In practice a permanent CLIP number is used if the CLIP of the user concerned is always to remain the same in the public network, regardless of the path used for routing
an outgoing call. Break-out is a typical application (see page 251).
If a call goes out to the public network, the permanent CLIP number is retained unchanged together with the numbering plan identifier NPI and the type of number TON,
even if the call is routed via another PINX (see example on page 98
The required CLIP number, the Numbering plan identifier (NPI) and Number
type (TON) are entered in the user configuration. The setting Create CLIP number
automatically must be deactivated.
For the Numbering plan identifier (NPI)E.164 is set in normal case.
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200
624 32 00
032 624 11 11
200
E.164
60
E.164
624 32 00
Users:
Automatic
CLIP
CLIP sent
CLIP No., NPI
and TON from
user
CLIP created
automatically [1]
CLIP
NPI =
TON = Subscriber
Example
no
yes
Conguration: User Trunk groups
CLIP:
NPI: Unknown
TON: Subscriber Unknown
DDI number:
Region:
4. 4. 3Suppressing CLIP / COLP (CLIR / COLR)
If Restrict call identification (CLIR) has been activated in the caller’s user configuration, the information sent along with the CLIP and COLP numbers specifies that they
are not to be displayed to the call’s recipient (CLIR: Calling Line Identification Restriction, COLR: Connected Line Presentation Restriction). In this case the network provider does not forward the CLIP number to the recipient (the CLIP number may nonetheless be sent to a number of public authorities, such as the police, see also "Display
CLIR", page 92).
The same setting is also used to prevent the name being displayed to the call's recipient. The suppression of CNIP (Calling Name Identification Presentation) and CONP
(Connected Name Identification Presentation) is called CNIR (Calling Name Identification Restriction) and CONR (Connected Name Identification Restriction).
Depending on the network provider it may be necessary to subscribe to CLIR.
For each user CLIR can only be activated permanently or temporarily for one call (see
"Suppression of the call number display", page 454
).
4. 4. 4CLIP flowcharts for Outgoing Calls
[1] Continues on Fig. 37.
Fig. 35CLIP of an outgoing call to an external user in the public network
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200
[1]
624 32 00
032 624 11 11
200
E.164
60
E.164
624 32 00
CLIP from user
PISN user: CLIP
selection
CLIP created
automatically [1]
Users:
Automatic
CLIP
CLIP sent
CLIP No., NPI
and TON from
user
CLIP
NPI =
TON = Subscriber
Conguration: User Trunk groups
CLIP:
NPI: Unknown
TON: Subscriber Unknown
DDI number:
Region:
Example
ye
s
no
normal
[1] Continues on Fig. 37.
Fig. 36CLIP of an outgoing call to a PISN user
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[1]
032 624 11 11
60 200
60 200
200
NPI = PNP
TON = level 1
CLIP No. =
NPI = unknown
TON = Unknown
CLIP No. =
NPI = unknown
TON = Unknown
CLIP No.
NPI = PNP
TON = level 1
CLIP No. =
NPI = unknown
TON = Unknown
CLIP sent
NPI and TON
from trunk
group
no
Trunk
groups:
Automatic
CLIP?
CLIP No., NPI and
TON from trunk
group
CLIP No. = own region
prex + user number
Private
Public
Trunk
groups: Net-
work type
CLIP No. = DDI No.
NPI and TON from
trunk group
Suitable DDI
number?
no
ye
s
yes
yes
[1]From Fig. 35 or Fig. 36
Fig. 37Creating an automatic CLIP for outgoing calls
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4. 4. 5CLIP Display with a Virtual Network PISN User
A public network user can be set up as a virtual PISN user in the communication
server. Internal users will then perceive the user as another internal user: A call is signalled with the internal ringing pattern. The internal number can also be dialled for outgoing calls. Individual mobile user or entire number blocks can be integrated in this
way.
Setting Up a Virtual Network PISN User
A PISN user is set up for this purpose (see "Numbering plan", page 92
lic network user’s full number under External Number. For outgoing calls the configured number will be dialled via the configured route instead of the dialled PISN user
number. This mechanism is similar to the one used for abbreviated dialling.
When the user calls up from the public network, his CLIP number will be compared
with the numbers of all the PISN users. If there is a match, the called user is shown the
PISN user number by way of CLIP instead of the CLIP sent from the public network.
). Enter the pub-
4. 5Display for Call Forwarding Unconditional
When Call Forwarding Unconditional is activated, it is useful for users to know that the
call was redirected, by whom and to whom. This means the called user is able to answer the call on behalf of the user who redirected the call to him. With this information
the calling user is better prepared for the call. This redirecting information is available
on system phones and ISDN terminals both internally and in private networks. If the
public network provider supports the function (special arrangement), the redirecting information is also available to virtual PISN users and users in the public network.
4. 5. 1Information displayed to the called user
The called user sees not only the caller's name and number but also that the call was
redirected and who redirected it (redirecting information).
Example:
User A calls user B, who has redirected to user C. The display on an system phone at
user C reads:
<CNIP A> / <CLIP A> forwarded from <CNIP B> / <CLIP B>
This redirecting information at user C is available for CFU, CFB, CFNR and Call De-
flection (CD). (With CD forwarded from is displayed instead of deflected from.)
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4. 5. 1. 1Outgoing call with local call forwarding
The configuration possibilities for the redirecting information depend on the destination
user:
If the destination user is
• an internal user in the local PINX, the redirecting information is always transmitted
to the called user.
• a PISN user, a PISN user in a virtual network, an integrated mobile/external phone
user or a public network user, you can select in the trunk group configuration
(=bg) whether the redirecting information is to be sent to the called user or suppressed (Send redirecting information).
• a public network user and if CLIR is activated at the user who carried out the redirecting, the called user will see neither the originator of the call nor that it has been
redirected. This even though the calling user did not activate CLIR. To prevent this,
you can deactivate the (=bg) CLIR for redirected calls parameter in the trunk
group configuration.
In a call forwarding chain with several users the name/number of the first user in the
chain is displayed as redirecting information to the called user.
4. 5. 1. 2Incoming call with CDE overflow
If in the event of a CDE overflow the call is routed from one call distribution element to
another due to the entries under (=dh) with CDE if busy or CDE if no answer,
the redirecting information provided to the called user depends on the new destination:
If the destination is
• an internal user or a user in a private QSIG network, the name/number of the CDE
is transmitted.
• a virtual network PISN user, the direct dial number to which the call is made is transmitted.
• an external user in the public network, no redirecting information is transmitted.
4. 5. 1. 3Incoming call that is already redirected
The redirecting information is also available to the called user in the case of an incoming call redirected via a PISN user or a user in the public network. If the call is routed
via a call distribution element, it is useful in certain cases if the name/number of the
CDE is displayed instead of the redirecting information. Moreover, in the CDE configuration (=dh) deactivate the parameter Show forwarding information instead of
CDE name (default value = activated).
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4. 5. 2Information displayed to the calling user
The calling user sees not only the called user’s name and number but also that the call
is being redirected and to whom (redirecting information).
Example:
User A calls user B, who has redirected to user C. The display on an system phone at
user A reads:
This redirecting information at user A is available for CFU, CFB and Call Deflection
(CD). (With CD forwarded to is displayed instead of deflected to.)
4. 5. 2. 1Incoming call with local call forwarding
The caller's configuration possibilities for the redirecting information depend on the
call's origin:
If the caller is
• a user in the local PINX, the redirecting information is always transmitted to the user
who is calling.
• a PISN user, a PISN user in a virtual network or a user in the public network, you
can select in the trunk group configuration (=bg)whether the redirecting information should be sent to the calling user or suppressed ((Send redirecting informa-
tion)).
• a public network user or if the user who redirected the call has activated COLR, the
caller will not see that he is being redirected. If this setting is required only for internal redirected calls but not external ones, in the trunk group configuration(=bg),
the COLR for redirected calls parameter can be deactivated.
In a call forwarding chain with several users the name/number of the last user in the
chain is displayed as redirecting information to the calling user.
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A
B
Trunk groups
PISN users
CLIP basic settings
CLIP A
4. 5. 2. 2Incoming call with CDE overflow
If in the event of a CDE overflow the call is routed from one call distribution element to
another due to the entries under (=dh) with CDE if busy or CDE if no answer,
the redirecting information provided to the called user depends on the new destination:
If the destination is
• an internal user or a user in a private QSIG network, the name/number of the CDE
is transmitted.
• a virtual network PISN user or an external user in the public network, no redirecting
information is transmitted.
4. 5. 2. 3Outgoing call with non-local redirection
The redirecting information is also available to the calling user in the case of an outgoing call that is not redirected via his own communication server but via a PISN user, an
integrated mobile/external phone user, a virtual network PISN user or a public network
user.
4. 6CLIP / COLP Settings
The following settings affect the CLIP and, by analogy, the COLP, too.
Fig. 38CLIP incoming
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D
C
C’s CLIP
User
PISN users
Trunk groups
Fig. 39CLIP outgoing
Tab. 11CLIP related settings
Configuration
Element
UserCreate CLIP number automatically
PISN userCall number
Trunk groupRing NPI 'Unknown'
ParameterAffect on CLIP
Restrict call identification (CLIR)
CLIR for redirection
Restrict call identification while connected (COLR)
COLR for redirection
Numbering plan identifier (NPI)
Type of number (TON)
CLIP number
CLIP selection (Normal, CLIP from user)
Create CLIP number automatically
Restrict call identification (CLIR)
CLIR for redirection
Restrict call identification while connected (COLR)
COLR for redirection
Numbering plan identifier (NPI)
Type of number (TON)
CLIP number
Truncate CLIP
Send redirection/redirecting information
ECT information
Transit CLIP format
Transit exchange access prefix
Send incoming CLIP for exchange-to-exchange connections
Call to the public network with exchange access prefix via a trunk group with Net-
work type = Public:
If the setting Create CLIP number automatically is activated, the DDI number is
used as CLIP, providing the user is himself reachable by incoming calls via the path
trunk group → DDI plan → CDE. If there is no direct dialling plan or corresponding DDI
number, the CLIP number entered in the trunk group will be used instead.
The numbering plan and type of number are always taken from the trunk group.
If the setting Create CLIP number automatically is deactivated, the configured number is used without further alterations.
Internal call to a PISN User
The creation of the CLIP number depends on the configured PISN user. If the PISN
user has the setting CLIP selection = Normal, the DDI number is used as CLIP, providing the user is himself reachable by incoming calls via the path trunk group → Direct
dialling plan→ CDE.
If there is no direct dialling plan or corresponding DDI number (which is normally the
case), the user’s internal call number is used instead.
If the PISN user has the setting CLIP Selection = CLIP from user, the CLIP number
is created in the same way as for a call to the public network. This means that a permanently defined CLIP number can also be transmitted in the private network.
Internal call to an integrated mobile/external phone user
The CLIP is built according to the configuration of the mobile/external phone assigned
to the integrated user.:
•If CLIP selection = Normal is configured for the user in the terminal interface settings, the direct dialling number of the calling user is used as the CLIP, regardless of
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his settings. If there is no corresponding direct dialling number, the internal call number is used in its place.
• If the terminal interface has the setting CLIP Selection = CLIP from user, the
CLIP number is created in the same way as for a call to the public network. In this
case, the caller's setting Create CLIP number automatically is decisive.
Call to the private network with route selection
Call to the private network with route selection via a trunk group with Network type =
Private:
By analogy with the call to a PISN user with the setting CLIP selection = Normal.
4. 6. 2PISN user
External call number setting
The call number entered under External call number is compared with the CLIP
number of an incoming call. If the two numbers match up, the PISN user number is displayed as the CLIP with Numbering plan identifier (NPI) = Private and Number
(TON) = Level 0.
CLIP selection settings
See "Internal call to a PISN User", page 89
.
4. 6. 3Trunk group
Call if NPI unknown setting
If a call with Numbering plan identifier (NPI) = Unknown is received, it is signalled
with the internal or external ringing pattern on the basis of the setting Call if NPI ’Un-
known’. It is also decided at the same time whether the exchange access prefix (0-)
should precede the CLIP number.
CLIP cut setting
A digit sequence can be configured under CLIP cut. If the sequence matches the initial digits of the CLIP number received, the digits will be truncated. This setting is normally used to remove any superfluous "0".
Setting Create CLIP number automatically
The setting Create CLIP number automatically only has an impact if the trunk group
configuration (=bg) Network type = Private is set.
If the setting is activated, the numbering plan identifier and type of number are left unchanged.
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If the setting is deactivated, the numbering plan identifier and type of number are taken
from the trunk group setting, but not the actual CLIP number. This may be necessary in
cases where connected third-party systems do not process numbering plan identifiers
and types of number correctly.
Numbering plan identifier (NPI), Number type (TON), CLIP number
These settings are used if the CLIP number could not be created automatically. This is
the case when there is no suitable DDI number available with a call to the public network.
ECT information
If the parameter ECT information is activated, the new CLIP is also transmitted in
the event of a call transfer to the exchange, provided the network interface involved is
in the same trunk group.
Example:
Internal user A calls internal user B, who transfers to external user C. After the call
transfer, C is presented with A’s new CLIP instead of B’s old CLIP.
The same applies accordingly with COLP, if the caller is an external user.
Example:
External user A calls internal user B, who transfers to internal user C. After the call
transfer, A is presented with C’s new COLP instead of B’s old COLP.
Note:
With some carriers there are problems in connection with ECT information. Transmission of this
information can therefore be suppressed with the deactivation of the parameter ECT infor-
mation.
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4. 6. 4CLIP/CLIR settings
These settings are used to truncate prefixed access digits so that the CLIP number is
as short as possible.
To enable the communication server to interpret CLIP numbers correctly, own local
prefixes must be entered in the location-related regional settings (=fz):
• International and national prefixes for the locations (example: "00" and "0" for Switzerland, "00" and "-" for France)
• Country code and toll area code of the location (example: for Switzerland "41", for
Geneva "22"", see also "Numbering Plan Identifier E.164", page 55
Mitel Advanced Intelligent Network:
In an AIN the nodes may be spread over different regions or even countries. Some settings do
not apply throughout the system but only to one region. A region is assigned to one or more AIN
nodes. An region can also be assigned for each trunk group. The trunk group allocation takes
priority over the node-specific allocation.
Display CLIR
When CLIR is activated (Restrict call identification (CLIR)) the public network provider will still send a CLIP to special customers, for instance the fire brigade and the
police. The CLIR information will, however, include the CLIP (see also "Suppressing
CLIP / COLP (CLIR / COLR)", page 81).
In the private leased-line network a CLIP is always sent with an activated CLIR. It is
also provided with the CLIR information.
With the activation of the Ignore call identification restriction (ignore CLIR) setting, a
clip with CLIR information is still displayed with incoming calls.
In internal traffic, a suppressed CLIP is always displayed.
).
4. 6. 5Numbering plan
The CLIP number is prefixed with the regional prefix for outgoing calls to a PISN user
or via a trunk group with Network type = Private.
For incoming calls, the regional prefix is removed from the CLIP number (provided it
begins with that digit sequence).
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201
340
310
A
B
200
C
330
D
E
F
G
022 222 22 22
066 333 33 33
066 666 66 xx
055 555 55 xx
055 777 77 77
Region 60
PINX 3
PINX 2
PINX 1
Region 50
PISN
4. 7Examples of CLIP Displays in the PISN
Various scenarios are used in a sample network to illustrate how CLIP displays are
handled in a PISN. Fig. 40 shows the sample network.
Fig. 40Sample network: PISN with two regions and one virtual network user
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340
340
340
C
A
201
50340
Destination PINX
Source PINX
PINX 2
PINX 1
4. 7. 1PISN-Internal Calls
Ordinary PISN-Internal Call
User C (340) on PINX 2 calls user A on PINX 1 by a direct route. Both users belong to
the same region.
Fig. 41Example 1: User C calls user A (excerpt from Fig. 40)
Tab. 12Example 1: Creating and presenting user C’s CLIP number
StepCLIP numberNPITONDescription
1340PNPLevel 0User C → PINX 2
250340PNPLevel 1PINX 2
3340PNPLevel 0PINX 1
4340PINX1
• There is no suitable DDI number.
→
PINX 1
• The system’s own regional prefix is deleted
• TON is adapted.
→
user A
• Presentation on the system phone
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340
340
340
201
50340
50340
A
C
PINX 3PINX 2
PINX 1
Region 60
Region 50
Transfer PINX
Region 50
Source PINX
Destination PINX
PISN - Internal Call with Overflow Routing
User C (340) on PINX 2 calls user A on PINX 1 via PINX 3 as all the available communication channels between PINX 2 and PINX 1 are busy. PINX 3 belongs to Region 60.
Fig. 42Example 2: User C calls user A, overflow routing (excerpt from Fig. 40)
Tab. 13Example 2: Creating and presenting user C’s CLIP number
StepCLIP numberNPITONDescription
1340PNPLevel 0User C → PINX 2
250340PNPLevel 1PINX 2
350340PNPLevel 1PINX 3
450340PNPLevel 1PINX 3
5340PNPLevel 0PINX 1
6340PINX 1
• There is no suitable DDI number.
→
PINX 3
• There is no suitable DDI number.
→
PINX 1
• The system’s own regional prefix is deleted
• TON is adapted.
→
user A
• Presentation on the system phone
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340
F
C
340
50340
54
055 555 55 54
055 555 55 54
022 222 22 22
Source PINXGateway PINX
PINX 1
PINX 2
4. 7. 2Outgoing Calls to the Public Network
Call to the Public Network via a Gateway PINX
User C (340) on PINX 2 calls user F on the public network via PINX 1.
PINX 1 has a DDI number for user C (54).
The following CLIP characteristics are set in the trunk group configuration of PINX 1:
•CLIP number = 50
•Numbering plan identifier (NPI) = Unknown
•Number type (TON) = Unknown
Fig. 43Example 3: User C calls user F in the public network (excerpt from Fig. 40)
Tab. 14Example 3: Creating and presenting user C’s CLIP number
StepCLIP numberNPITONDescription
1340PNPLevel 0User C → PINX 2
250340PNPLevel 1PINX 2
3340PNPLevel 0PINX 1
454UnknownUnknownPINX 1
5055 555 55 54Public exchange
• There is no suitable DDI number.
→
PINX 1
• The system’s own regional prefix is deleted
• TON is adapted.
→
Public exchange
• There is a suitable DDI number, which is used as a
CLIP number and sent to the public network.
• Presentation on the terminal
96
→
user F
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340
F
C
340
50340
60
055 555 55 54
066 666 66 60
022 222 22 22
Gateway PINXSource PINX
PINX 3
Region 50
PINX 2
Region 60
Call to the Public Network via a Gateway PINX with Overflow Routing
User C (340) on PINX 2 calls user F on the public network via PINX 3 as all the available communication channels between PINX 2 and PINX 1 are busy. PINX 3 does not
have a DDI number for user C.
The following CLIP characteristics are set in the trunk group configuration of PINX 3:
•CLIP number = 60
•Numbering plan identifier (NPI) = Unknown
•Number type (TON) = Unknown
Fig. 44Example 4: User C calls user F via an alternative path (excerpt from Fig. 40)
Tab. 15Example 4: Creating and presenting user C’s CLIP number
StepCLIP numberNPITONDescription
1340PNPLevel 0User C → PINX 2
250340PNPLevel 1PINX 2
350340PNPLevel 1PINX 3
460UnknownUnknownPINX 3
5066 666 66 60Public exchange
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• There is no suitable DDI number.
• There is no suitable DDI number.
• The CLIP number entered in the trunk group configuration is sent to the public network.
• Presentation on the terminal
→
PINX 3
→
Public exchange
→
user F
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330
F
B
330
055 555 55 53
55 555 55 53
022 222 22 22
55 555 55 53
55 555 55 53
Region 60
Region 50
PINX 2
PINX 3
Gateway PINX
Source PINX
Call to the Public Network via a Gateway PINX with Overflow Routing and nonautomatic CLIP
User B (330) on PINX 2 calls user F on the public network via PINX 3 as all the available communication channels between PINX 2 and PINX 1 are busy.
PINX 3 does not have a DDI number for user B.
In the user configuration of user B, the setting Create CLIP number automatically is
deactivated. The CLIP settings of the user configuration are used:
• CLIP number = 55 555 55 53
•Numbering plan identifier (NPI) = E.164
•Number type (TON) = National
Fig. 45Example 5: User B calls user F (excerpt from Fig. 40)
Tab. 16Example 5: Creating and presenting user B’s CLIP number
StepCLIP numberNPITONDescription
1330PNPLevel 0User B → PINX 2
255 555 55 53E.164NationalPINX 2
355 555 55 53E.164NationalPINX 3
455 555 55 53E.164NationalPINX 3
5a055 555 55 53Public exchange
5b066 666 66 60Public exchange
98
• A suitable DDI number is not searched for.
→
PINX 3
• CLIP number is buffered unchanged
• A suitable DDI number is not searched for.
→
Public exchange
• CLIP number is sent unchanged to the public network.
→
• Presentation on the terminal if special arrangement is
available (see page 74
• Presentation on the terminal if special arrangement is
not available (see page 74
user F
).
→
user F
).
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340
C
G
066 333 33 33
0-066 333 33 33066 333 33 33066 333 33 33
66 333 33 33
Destination PINX
PINX 2PINX 1
Gateway PINX
4. 7. 3Incoming calls from the public network
User G on the public network calls user C on PINX 2 via PINX 1. He dials 055 555 55
54.
Fig. 46Example 6: User G calls user C (excerpt from Fig. 40)
Tab. 17Example 6: Creating and presenting user G’s CLIP number
StepCLIP numberNPITONDescription
1066 333 33 33E.164UnknownUser G → Exchange → PINX 1
266 333 33 33E.164NationalPINX 1
366 333 33 33E.164NationalPINX 1
466 333 33 33E.164NationalPINX 2
50-066 333 33 33
1)
In PINX 3’s trunk group configuration 066 666 60 is entered as the master number.
1)
• Prefix is truncated
• TON is set to National
→
PINX 2
• CLIP number is not altered.
PINX 2 → user C
• Presentation on the system phone
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340
CG
066 333 33 33
0-066 333 33 33
066 333 33 33
066 333 33 3366 333 33 3366 333 33 33
Region 50
PINX 2
Destination PINX
Transfer PINX
Gateway PINX
PINX 3PINX 1
Region 50
Region 60
Call from the Public Network with Overflow Routing
User G on the public network calls user C on PINX 2 via PINX 1 and PINX 3 as all the
available communication channels between PINX 2 and PINX 1 are busy. He dials 055
555 55 54.
Fig. 47Example 7: User G calls user C via PINX 3 (excerpt from Fig. 40)
Tab. 18Example 7: Creating and presenting user C’s CLIP number
StepCLIP numberNPITONDescription
1066 333 33 33E.164UnknownUser G → Exchange → PINX 1
266 333 33 33E.164NationalPINX 1
366 333 33 33E.164NationalPINX 1
4333 33 33E.164SubscriberPINX 3
566 333 33 33E.164NationalPINX 3
666 333 33 33E.164NationalPINX 2
70-066 333 33 33PINX 2
• Prefix is truncated
• TON is set to National
→
PINX 3
• Long-distance code is truncated as it is the same as
the system’s own long-distance code
• TON is set to Subscriber.
→
PINX 2
• CLIP number is not altered.
→
user C
• Presentation on the system phone
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