Avaya B179 SIP Conference Phone
Installation and Administration Guide
16-603915 Issue 6
June 2012
Avaya B179 SIP Conference Phone Installation and Administration Guide
ABOUT THIS DOCUMENT
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2
Avaya B179 SIP Conference Phone Installation and Administration Guide
ABOUT THIS DOCUMENT
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Avaya B179 SIP Conference Phone Installation and Administration Guide
ABOUT THIS DOCUMENT
This document only includes setup, registration of accounts and conguration. The use
of the conference phone is described in the Avaya B179 SIP Conference Phone - Quick
Reference Guide (16-603916) and the Avaya B179 SIP Conference Phone - User Guide
(16-603918). The latest version of all documentation can be downloaded from support.
avaya.com.
Please note that there are also supporting Application Notes describing the steps to
congure the Avaya B179 SIP Conference Phone to work with certain systems and also
how to congure the systems (eg. administer SIP extensions).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CONTENT
Description 2
Display information and web interface 3
Display screen ................................................................3
Navigation and selection in menus ...................................4
Using the web interface ...................................................6
Installation 7
Connecting.....................................................................7
Obtaining a network address ........................................... 8
Login .............................................................................8
Software upgrade and basic settings ................................9
Registering an account .................................................. 12
Settings 14
Login ........................................................................... 14
Basic ........................................................................... 14
SIP .............................................................................. 17
Network .......................................................................23
Media .......................................................................... 26
LDAP ..........................................................................29
Web interface ............................................................... 31
Time & Region .............................................................32
Provisioning .................................................................33
System ........................................................................ 34
Headset and PA installation and settings 35
Connecting a wireless headset .......................................35
Connecting a PA interface box .......................................36
PA settings ..................................................................37
Hard system recovery 39
Provisioning – upgrade and conguration 40
Firmware upgrade on a single phone ...............................40
Using a conguration le ...............................................42
Using a device management server .................................48
Upgrading in IP Ofce...................................................52
How to do a downgrade ................................................ 52
Managing phone book and conf. guide 53
Importing and exporting contacts ...................................53
Importing and exporting conference groups .....................54
Checking status and logs 55
Device .........................................................................55
Network .......................................................................55
Time & Region .............................................................56
SIP ..............................................................................56
Media .......................................................................... 57
Log .............................................................................58
Technical data 60
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Avaya B179 SIP Conference Phone Installation and Administration Guide
DESCRIPTION
Maintenance
Clean the equipment with
a soft, dry cloth. Never use
liquids.
Display screenSpeaker Microphone
Keypad
Network cable port
LEDs
Flashing blue Incoming call
Steady blue light Call in progres s
Flashing red On hold,
Steady red light Mute,
SD memory card port
Expansion microphone port
microphone and
speakers turned off
microphone turned
off
Expansion
microphone
port
AUX port
Security lock po rt
Power supply por t
UP ARROW
Navigation in menus
Display of call list
Cancel
No/end/back
Start /stop
recording
Increase volume
Decrease volume
Mute
Hold
Some Avaya B179 have a different keypad with other symbols. This does not affect the functions of the buttons.
Hold down a button for 2 seconds
Menu
Settings
Alphanumerical buttons
to open the phon ebook
DOWN ARROW
Navigation in menus
Display of call list
OK
Yes/conrm choice
Answer/connect new line
During a call: Press to call
a new person
Hang up/end line
Conference
Automatic dialling of
conference groups
(See note on page 54)
One press of this button
will always connect all parties to a conference call
Line selection
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Avaya B179 SIP Conference Phone Installation and Administration Guide
DISPLAY INFORMATION AND WEB INTERFACE
DISPLAY SCREEN
On Hook
Press to display this screen.
Date
Time
Display text (can be changed)
Registere d
Not registered
Account name (can be changed)
Off Hook
Press to display this screen.
Information tex t (see below)
Phone lines (L1–L4)
Line status (see below)
Call duration
Secure connection (see page 20)
Line status:
Line free (Before account name – telephone not registered)
Line connected (Before account name – telephone registered)
Line on hold (“HOLD” displayed on the screen – all calls on hold)
Line (called party) bus y
Own line put on hold by other p arty
Recordin g call
Secure connection
Information tex t displays one of the following:
• Number or name of e ach phone line
(The name will be displayed if a numb er is in the phone book)
• Explanation of what you should do (For example ENTER NUMBER)
• Status (For example when you pla ce all calls on hold)
Line menu
Press to switch to and from this menu.
Line/number/name
New line
Option for creating or splitting conference calls
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Avaya B179 SIP Conference Phone Installation and Administration Guide
DISPLAY INFORMATION AND WEB INTERFACE
Menu
Press to switch to and from a menu.
Current menu
Submenu
Marked option – op en by pressing OK button
Scrolling lis t
(indication of where the marked optio n is in the list or menu)
List of setting options:
Existing settings
Marked option – select by pressing OK button
List of names:
Marked name – selec t by pressing OK but ton
NAVIGATION AND SELECTION IN MENUS
Press .
Select the option you want from the menu using the arrow buttons.
Conrm by pressing OK to select the marked option.
Cancel the setting or go back one level in the menu by pressing .
Quit the menu by pressing again.
Note that after you have made changes to a setting, you must press OK to activate the
setting.
It is possible to open a menu option directly by pressing the number button that cor-
responds to the position of the option in the menu (e.g. 2 to open PHONE BOOK and
then 3 to select EDIT CONTACT).
Writing style in instructions
In the instructions, > SETTINGS (6) means you should:
Press .
Mark the SETTINGS option using the arrow buttons and conrm by pressing OK to
open the menu (or press button number 6).
Correspondingly, Phone book > Conference Guide in the web interface means you should
select Menu Phone book and the Conference Guide tab.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
DISPLAY INFORMATION AND WEB INTERFACE
Menu tree
1 2 3 4 5 6 7 8
PROFILES
DEFAULT
PROFILE 1
PROFILE 2
PROFILE 3
PROFILE 4
PHONE BOOK EXT. PHONE BOOK
SEARCH
CONTACT
ADD
CONTACT
EDIT
CONTACT
ERASE
CONTACT
ERASE
ALL
STATUS
CONF GUIDE
SEARCH
GROUP
ADD
GROUP
EDIT
GROUP
ERASE
GROUP
ERASE
ALL
STATUS
PLAYBACK
FILE
RENAME
FILE
ERASE
FILE
ERASE
ALL
SETTINGS
STATUS
SETTINGS SYSTEM STATUSRECORDING
BASIC
ADVANCED
BASIC
SETTINGS
LANGUAGE
KEY TONE
RING LEVEL
EQUALIZER
AUX PORT
PA
TIME
FORMAT
DATE
FORMAT
SCREEN
TEXT
DEFAULT
RESTART
REBOOT
FACTORY
RESET
ACCOUNTS
NETWORK
NAT
TRAVERSAL
MEDIA
TIME
DEVICE
Menu tree, advanced settings
The advanced settings are protected by administrator’s PIN code. The default value is 1234.
SETTINGS
BASIC
PIN
ADVANCED
ACCOUNTS
ACCOUNT 1
ACCOUNT 2
TRANSPORT
NETWORK NAT
ETHERNET
802.1X
ENABLE/
DISABLE
ACCOUNT
EDIT
ACCOUNT
TRAVERSAL
IP
VLAN
AUTH
STUN
OFFER ICE
The simplest way to make settings and edit contacts is using a PC and the Avaya
B179 web interface.
MEDIA TIME REGION WEB
CODEC
VAD
DTMF
SIGNALLING
SECURITY
SRTP
SIGNALLING
NTP
TIME DATE
TIME ZONE
DAYLIGHT
SAVING
G722
PCMA
PCMU
G729
INTERFACE
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Avaya B179 SIP Conference Phone Installation and Administration Guide
DISPLAY INFORMATION AND WEB INTERFACE
USING THE WEB INTERFACE
You can use the web browser of a PC connected to the same network to manage contacts,
conference groups and settings in the Avaya B179.
For security reasons, recordings can only be managed directly on the Avaya B179. All
other settings that can be made directly on the Avaya B179 can also be made via the web
interface. It is also possible to import and export contacts and conference groups, name
user proles and change PIN codes, which can only be done via the web interface. The
administrator can also view logs, update software and create a conguration le.
The default setting for the PIN code is 0000 for the user account (Default, Prole 1,
Prole 2, Prole 3 and Prole 4) and 1234 for the administrator’s account (Admin). We
recommend that you change the PIN codes in order to protect the settings. The code may
consist of eight digits. The administrator can always view and change the PIN codes to the
user accounts. The administrator’s PIN code can only be reset with a complete reset to
factory settings.
Checking IP address
Press and select the sub menu STATUS > NETWORK (8,2).
Check the conference phone’s network address under the heading IP ADDRESS.
Use this address to log into the web server in the conference phone.
Login
Log into the web server in Avaya B179 by entering the phone’s network address in
your computer’s web browser.
Select Admin as Prole and enter your PIN.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
The printed Installation Guide provides brief and simplied installation instructions. The
guide includes the basic settings for a quick start and works in most cases.
CONNECTING
Connect the Avaya B179 to the network as illustrated below.
Plug the Avaya B179 into the mains using the power adapter as illustrated below.
The Avaya B179 can be driven directly from the network (Power over Ethernet,
ClassIII) if the network supports this.
Place the conference phone in the middle of the table.
The Avaya B179 must obtain a network address and be registered in a SIP PBX before it
can be used. The easiest way to register an account and make the settings in the Avaya
B179 is using a computer connected to the same network and via the integrated web
server.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
OBTAINING A NETWORK ADDRESS
Connecting to a network with DHCP
See “Check IP address” under “USING THE WEB INTERFACE” on page 6.
Connecting to a network with static IP addresses
You need the IP address, host name, domain, netmask, gateway, DNS 1, and DNS 2. The
host name can be set freely. The domain and secondary DNS can be left blank.
Press and select SETTINGS > ADVANCED (6,2).
Enter the PIN code.
The default code is 1234.
Select NETWORK (2)
Select S TATIC IP.
Enter values for the IP ADDRESS.
Enter three digits (begin with 0 if necessary), press OK, enter three digits, and so on.
Enter HOST NAME
Default is avaya.
Enter DOMAIN
Enter NETMASK
Ente r GAT EWAY
Enter DNS 1
Enter DNS 2
The display shows DONE.
LOGIN
See “Login” under “USING THE WEB INTERFACE” on page 6.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
SOFTWARE UPGRADE AND BASIC SETTINGS
The following settings should be done during installation.
See page 48 about Device management if you are responsible for installing or upgrad-
ing many phones.
Note that all settings on the Basic tab also affect the user prole Default. Other user
proles can be changed individually. The settings on the Basic tab, except the name and
PIN for Admin, can be modied by any user. Other settings require a login as Admin.
Upgrade software
See the heading “PROVISIONING – UPGRADE AND CONFIGURATION” on page 40 for a
detailed description and upgrading options.
Select Settings > Provisioning.
Click on Check Now.
Compare the latest version with the current version (shown on the web page).
If you want to upgrade, select the desired version in the list box and click on Upgrade.
The browser window and the display on the Avaya B179 shows that the upgrade has begun.
The download and installation can take several minutes. Do not interrupt the upgrade
and do not disconnect plugs to the Avaya B179 during the upgrade. Interrupting the
upgrade may render the conference phone inoperable.
When installation is complete, the text “Upgrade Complete. The unit will be reboot-
ed.” is shown in your browser, and after a while you hear the Avaya music signature,
which indicates that the conference phone has started.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
Setting time and region
Select Settings > Time & Region.
Select the time zone and, if you wish, correction for DST (Daylight saving).
It is also possible to set the time and date manually or choose a different time server.
Select the region where you are.
This setting affects the frequency and duration of the signaling tones (ring signal, busy
tone, etc).
Save the setting.
The Avaya B179 reboots with the new settings.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
Changing the language
Select Settings > Basic.
Select the desired language in the list box after Language and save the setting.
Note, this only affects the phone language, not the web interface.
Changing the PIN
We recommend that you change the PIN code for Admin from the default setting to
protect the settings. Make a note of the new PIN code and keep it in a safe place. The
administrator’s PIN code can only be reset by a full factory reset!
Select Settings > Basic and click the Edit button on the Admin line.
Enter a new PIN.
The PIN code may consist of 8 digits.
Click on the Set and Save buttons.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
REGISTERING AN ACCOUNT
The conference phone supports two accounts. The second account is a fallback account
that is automatically used if the phone fail to register to the main account. Note that this
account then will be used as long as the fallback account is available. If the phone fails to
register to the fallback account, it will try to use the main account again.
To register your phone, you must have access to the account information and all necessary settings that the SIP PBX requires.
See the heading “SIP” on page 17 for a detailed description of all settings.
Select Settings > SIP.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
INSTALLATION
Click Yes at Enable account under Main account.
Enter the account information you have received.
Account name can be chosen freely and is the name or phone number you want to
appear in the phone display.
Leave the default values if you have no other information.
Select a method of NAT traversal if you have received this information.
Select a different transport protocol if you have received this information. See page
20 about using a secure transport protocol.
Save the settings by clicking the Save button.
The Avaya B179 responds by showing REGISTERING. If registration is successful, your selected
account name will appear at the bottom of the display screen next to a shaded square.
Make media settings
Select a different codec priority, if you do not accept the default settings. See page
26.
Select a SRTP option if you need a secure media protocol. See page 27. Note that this
also requires a corresponding transport setting on the SIP tab.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Almost all settings can be done directly on the Avaya B179. See “NAVIGATION AND
SELECTION IN MENUS” on page 4 for using the menu system. We explain how to make
settings using the web interface as this is the easiest method.
For safety reasons, recordings can only be managed directly on the Avaya B179. All
other settings can be changed via the web interface. The web interface also allows you
to import and export contacts and conference groups, rename user proles and change
PIN codes. As an administrator, you can also study logs, upgrade the software and
create an XML based conguration le for easier management of a set of phones.
LOGIN
See “USING THE WEB INTERFACE” on page 6 for a description of how to log in to the
web server in the Avaya B179.
BASIC
Select Settings > Basic.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
These settings affect the Admin and Default proles. To change the basic settings of a
user prole, you need to log in with that prole.
Proles – edit name and PIN
We recommend that you change the PIN code from the default setting to protect the
settings.
Select Settings > Basic and click the Edit button on the account you want to change.
Enter a new PIN code.
The PIN code may consist of 8 digits.
You can also choose to change the name of a user prole.
Click on the Set and Save buttons.
Make a note of the new PIN code and keep it in a safe place.
The administrator’s PIN code can only be reset with a complete reset to factory set-
tings!
Language
Select phone language using the list box and click on the Save button.
On phone: > SETTINGS > BASIC > LANGUAGE (6,1,1).
Ring level
There are six volume levels plus a silent mode. You will hear the ring tone for each level
you select. If you select silent mode, only the blue LEDs on the phone ash when an
incoming call is received.
Select level using the list box and click on the Save button.
On phone: > SETTINGS > BASIC > RING LEVEL (6 ,1,3).
Key tone
You can select whether or not you want a tone to be heard when you press a button.
Select On or Off and click on the Save button.
On phone: > SETTINGS > BASIC > KEY TONE (6,1,2).
Recording
It is possible to turn off the recording feature. This setting can only be done by the administrator and affects all proles.
Select On or Off and click on the Save button.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Recording tone
A short beep is heard every 20 seconds so that all the parties in the call know it is being
recorded. This feature can be turned off.
Select On or Off and click on the Save button.
On phone: > RECORDING TONE > SETTINGS (5,5).
Settings when connecting external equipment (Aux)
The Avaya B179 can be connected to a wireless headset or an external PA system. An
optional PA interface box is required for PA system connection.
Select the PA option to activate features for external microphone mixer and PA
system.
Do not select the PA option unless a PA system is connected. This option turns off the
internal microphone and internal speakers as default. The HEADSET option may be
selected whether or not a headset is connected.
Time format
Select 12 hour or 24 hour and click on the Save button.
On phone: > SETTINGS > BASIC > TIME FORMAT (6,1,7).
Date format
Select date format and click on the Save button.
On phone: > SETTINGS > BASIC > DATE FO RM AT (6,1, 8).
Equalizer
The sound reproduction can be adjusted to the required pitch (SOFT, NEUTRAL or
BRIGHT).
Select Soft, Neutral or Bright and click on the Save button.
On phone: > SETTINGS > BASIC > EQUALIZER (6,1,4).
Screen text
The text on the display screen is shown when the Avaya B179 is in stand-by mode (on
hook).
Enter your new text in the text box and click on the Save button.
On phone: > SETTINGS > BASIC > SCREEN TEXT (6,1,9).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
SIP
Select Settings > SIP
The conference phone supports two accounts. The second account is a fallback account
that is automatically used if the phone fail to register to the main account. Note that this
account then will be used as long as the fallback account is available. If the phone fails to
register to the fallback account, it will try to use the main account again.
Main account and Fallback account
Enable account It is possible to store account information for future use, but
temporarily disable it.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Account name This is the name displayed on the screen. It can be set according
to company standards.
User The account (customer) name.
Registrar Shall contain the IP address or the public name of the SIP server
where the account is registered (e.g. 10.10.1.100 for a local SIP
server or sip.company.net for a public VoIP service provider)
Proxy Shall contain the proxy server used for Internet communication, if
any. Can be left blank.
Realm The protection domain where the SIP authentication (name and
password) is valid. This is usually the same as the registrar. If left
blank, or marked with a “*”, the phone will respond to any realm.
If specied, the phone will only respond to the specic realm when
asked for credentials.
Authentication name The name used for the Realm authentication. This may be the
same as the user name, but must be lled in.
Password The password used for the Realm authentication.
Registration Interval This is a request to the SIP server for when the registration should
expire. Avaya B179 automatically renews the registration within the
time interval if the phone is still on and connected to the server.
The default value is 1800 seconds.
On phone: > SETTINGS > ADVANCED > (PIN) > ACCOUNTS (6,2,1).
Nat traversal
NAT (Network Address Translation) is a rewall or router function that operates by rewriting
the IP addresses in the IP headers as packets pass from one interface to the other. When
a packet, for example, is sent from the inside, the source IP address and port are rewritten from the private IP address space into the address space on the outside (Internet).
NAT rewrites the addresses but leaves the packets themselves untouched. This kind of
translation works ne for many protocols, but causes a lot of trouble for SIP packets that
contain address information in their content (for example an INVITE request from one IP
address to another).
NAT traversal solves this problem, providing a “view from the outside” that makes it possible to replace the IP address in the SIP requests with the address shown on the other
side of the rewall.
Note that in some cases NAT traversal is not necessary. Some public service providers
of IP telephony keep track of the actual IP address used to register a phone, and the one
used in the SIP requests from the same phone, and then replaces the addresses in the
SIP messages.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
STUN STUN (Simple Traversal of UDP through NATs) is a protocol that
assists devices behind a NAT rewall or router with their packet
routing. STUN is commonly used in real-time voice, video, messaging, and other interactive IP communication applications..
The protocol allows applications operating through a NAT to discov-
er the presence and specic type of NAT and obtain the mapped
(public) IP address (NAT address) and port number that the NAT
has allocated for the application’s User Datagram Protocol (UDP)
connections to remote hosts. The protocol requires assistance from
a 3rd-party network server (STUN server).
STUN should be activated if an external SIP server cannot connect
to the Avaya B179 behind a rewall NAT function and the SIP
server supports STUN. A suitable STUN server is usually provided
by the VoIP service provider.
Note: STUN might also be referred to as Session Traversal Utilities
fo r NAT.
STUN host The IP address or public name of the STUN server.
Offer ICE ICE (Interactive Connectivity Establishment), is a STUN addition
that provides various techniques to allow SIP-based VoIP devices to
successfully traverse the variety of rewalls that may exist between
the devices. The protocol provides a mechanism for both endpoints
to identify the most optimal path for the media trafc to follow.
TURN TURN (Traversal Using Relay NAT ) TURN is an extension of the
STUN protocol that enables NAT traversal when both endpoints are
behind symmetric NAT. With TURN, media trafc for the session
will have to go to a relay server. Since relaying is expensive, in
terms of bandwidth that must be provided by the provider, and
additional delay for the media trafc, TURN is normally used as a
last resort when endpoints cannot communicate directly.
TURN User User authentication name on the TURN server.
TURN host The IP address or public name of the TURN server.
Password User authentication password on the TURN server.
On phone: > SETTINGS > ADVANCED > (PIN) > NAT TRAVERSAL (6,2,3).
Advanced
Enable SIP Replaces Default is Yes. Setting this option to No, will instruct the PBX not
to use the SIP replace header. Some PBXes try to take over the
bridging functionality from Avaya B179 using this command, which
causes the calls to interrupt.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Enable Blind Transfer Default is Yes. Setting this option to No, will disable the transfer
function ( > TRANSFER) during a call. This may be used if the
PBX does not support blind transfer.
Allow contact rewrite Default is Yes. When enabled, the B179 will store the IP address
from the response of the register request. If a change is detected,
the phone will unregister the current sip URI (contact), and update
the sip URI with the new address.
Transport
The transport setting only concerns the protocol to be used for SIP messages between the
devices involved. These settings do not include the media (the actual call). The settings
on the Media tab should be set accordingly.
Note that if you choose to use a secure connection, both units must support it. Otherwise
they cannot negotiate a connection. If an incoming call demands a secure TLS or SIPS
connection, the Avaya B179 uses the appropriate protocol even if you have set the phone
to use UDP.
Protocol UDP (User Datagram Protocol) is a protocol on the transport layer
in the Internet Protocol Suite. It is a stateless protocol for short
messages – datagrams. Stateless implies that it does not establish
any connection between sender and receiver in advance. UDP
does not guarantee reliability or ordering in the way that TCP does.
Datagrams may arrive out of order or go missing without notice.
The advantages it offers are speed and efciency.
UDP is the default protocol for SIP.
TCP (Transmission Control Protocol ) is a protocol on the transport
layer in the Internet Protocol Suite. TCP is the standard protocol
for Internet communication. TCP keeps track of all individual
packets of data, ensuring that they reach the receiver and are put
together properly. TCP is not the default protocol for SIP, because
it is slower and uses more bandwidth than UDP.
With UDP and TCP, SIP packets travel in plain text. TLS (Transport
Layer Security) is a cryptographic protocol that provides security
and data integrity for communications over TCP/IP networks. TLS
encrypts the datagrams of the transport layer protocol in use. The
secure connection may be to the end device or to the rst server
(usually the SIP server where the phone is registered). There is
no guarantee that there is a secure channel to the end point, but
because the SIP server is the only part receiving the user authentication, this is still a rather secure solution.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
SIPS (Secure SIP) is a security measure that uses TLS to provide
an encrypted end-to-end channel for the SIP messages. To use SIPS,
however, both VoIP devices and the SIP server must support it.
Even if Transport is set to TLS or SIPS, the Avaya B179 still accepts incoming UDP or
TCP signalling.
On phone: > SETTINGS > ADVANCED > (PIN) > ACCOUNTS > TRANSPORT (6,2,1,3).
TLS Settings
If you select TLS or SIPS under the transport setting, this additional setting appears on
the page.
It may be possible to use secure communication without a certicate and make changes
to these settings. In some cases, if you choose TLS or SIPS, the SIP server requires a
certicate for user/client verication. This should be specied in the account information.
Youcan further increase security by requiring verication of the server, or the client when
the Avaya B179 acts as a server for incoming calls.
Method The TLS includes a variety of security measures. The methods are
dened in the versions of the standard (SSL, SSL v2, SSL v3, TLS
v1, TLS v2). The default method is SSLv23, which accepts both
SSL v2 and v3.
Negotiation timeout The TLS settings are negotiated during a call setup (both incoming
and outgoing). If this negotiation does not succeed within the
specied time (seconds) the negotiation is aborted. Timeout is
disabled with 0 (zero).
Verify client When set to On, the Avaya B179 will activate peer verication for
incoming secure SIP connections (TLS or SIPS).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Require client certicate
When set to On, the Avaya B179 rejects incoming secure SIP
connections (TLS or SIPS) if the client does not have a valid
certicate.
Verify server When the Avaya B179 is acting as a client (outgoing connections)
using secure SIP (TLS or SIPS) it will always receive a certicate
from the peer. If Verify server is set to On, the Avaya B179 closes
the connection if the server certicate is not valid.
Certicate Here you can upload a certicate to the Avaya B179 to be used for
TLS or SIPS communication.
A certicate is a le that combines a public key with information
about the owner of the public key, all signed by a trusted third
party. If you trust the third party, then you can be sure that the
public key belongs to the person/organization named in that le.
You can also be sure that everything you decrypt with that public
key is encrypted by the person/organization named in the certicate.
Root certicate The public key in the root certicate is used to verify other certi-
cates. A root certicate is only needed if you have selected client
or server verication.
A root certicate is signed by the same public key that is in the
certicate, a so-called “self-signed” certicate. A typical root
certicate is one received from a Certicate Authority.
Private key Here you can upload a private key to the Avaya B179 to be used for
TLS or SIPS communication.
A private key is one of the keys in a key-pair used in asymmetric
cryptography. Messages encrypted using the public key can only be
decrypted using the private key.
Private key password The password used for encryption of the private key, if it is
encrypted.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
NETWORK
Select Settings > Network.
DHCP Dynamic Host Conguration Protocol is used by network devices
(clients) to obtain the parameters necessary for operation in the IP
network. This protocol reduces system administration workload, allowing devices to be added to the network with little or no manual
conguration.
DHCP should be set to On if no other information is given. When
set to On, all information on this page will be set automatically.
IP address IP address of the device (Avaya B179). The address is provided by
the network administrator or service provider if DHCP is not in use.
Hostname Set to avaya as default. Can be changed to suitable name.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Netmask Usually set to 255.255.255.0 to limit network trafc to the
subnet.
Domain The domain where the device is located. May be left blank.
Gateway The device or server used for Internet communication.
Primary DNS The address to the primary DNS (Domain Name System) server - a
program or computer that maps a human-recognisable name to its
computer-recognisable identier (IP address).
Secondary DNS The address of an optional secondary DNS server.
On phone: > SETTINGS > ADVANCED > (PIN) > NETWORK (6,2,2).
Quality of Service
Quality of service is used in IP networks to provide different priority to different applications, users, or to guarantee a certain level of performance to a critical data ow such as
voice or video. Differentiated Services or DiffServ is a networking architecture that speci-
es a simple mechanism for classifying network trafc using a 6-bit eld in the header of
the IP packets. VLAN (Virtual LAN) is a technology to logically divide a physical network
into several logical nets and thus to differentiate trafc.
SIP DiffServ Enter a value between 0 and 63 to prioritize the SIP messages.
Media DiffServ Enter a value between 0 and 63 to prioritize the media packets
(voice).
VLAN By enabling this option, all communication to and from Avaya B179
is done via the VLAN specied under VLAN ID. Note that this
VLAN also must be used to communicate with Avaya B179 via the
web interface.
VLAN ID The ID number to be used for the IP telephony VLAN.
VLAN map enable Enabling VLAN priority mapping from the DiffServ setting.
VLAN prio SIP Set a value between 0 and 7 to prioritize the SIP messages in the
VLAN.
VLAN prio media Set a value between 0 and 7 to prioritize the media packets in the
VLAN.
On phone: > SETTINGS > ADVANCED > (PIN) > NETWORK > ETHERNET > VLAN (6,2,2,2,1).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
802.1x
IEEE 802.1X is an IEEE Standard for port-based Network Access Control and is part of
the IEEE 802.1 group of networking protocols. It provides an authentication mechanism
to devices wishing to attach to a LAN or WLAN.
Enable 802.1x By enabling this option, Avaya B179 asks an authentication server
for permission when connected to the LAN.
EAP method Select which EAP (Extensible Authentication Protocol) method to
use: MD5 or TLS.
Username The device identity in the network.
MD5 password Password for the device identity when using MD5.
Certicate Here you can upload a certicate to the Avaya B179 to be used for
authentication when using TLS.
Root certicate The public key in the root certicate is used to verify other certi-
cates when using TLS.
Private key Here you can upload a private key to the Avaya B179 to be used for
authentication when using TLS.
TLS password The password used for encryption of the private key when using
TLS.
On phone: > SETTINGS > ADVANCED > (PIN) > NETWORK > ETHERNET > 8 02.1X AUTH
(6,2,2,2,2).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
MEDIA
Select Settings > Media.
The media settings determine how audio is sent between the devices. The devices negotiate via SIP before a call is connected. All devices must support the same media types,
codecs and security settings.
Codec
Codecs are used to convert an analog voice signal to a digitally encoded version and vice
versa. Codecs vary in the sound quality they deliver and the bandwidth required. The
Avaya B179 supports the most common codecs and each codec can be given a precedence depending on your requirements for high quality audio or low bandwidth use.
The priority can be set to from 4 (high) to 1 (low) or 0 (disabled)
G722 G.722 is an ITU-T standard codec that provides 7 kHz wideband audio
at a data rate within 64 kbit/s. It offers greately improved speech
quality compared with older narrowband codecs such as G.711, but
requires a high quality network connection between the devices.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
G711 Alaw G.711 is an ITU-T standard codec that uses audio companding.
Companding algorithms reduce the dynamic range of an audio
signal. In analog systems, this can increase the signal-to-noise
ratio achieved during transmission and, in the digital domain, can
reduce the quantization error.
Two main compression algorithms are dened in the standard,
the µ-law algorithm (used in North America and Japan) and A-law
algorithm (used in Europe and the rest of the world).
G711 Ulaw See G711 µ-law above.
G729 G.729 is an ITU-T standard codec that operates at 8 kbit/s. It is
mostly used in VoIP applications with low bandwidth requirement.
On phone: > SETTINGS > ADVANCED > (PIN) > MEDIA > CODEC (6,2,4,1).
Security
The media in VoIP calls is usually sent using the RTP protocol (Real-time Transport
Protocol ). RTP is a standardized packet format for delivering audio and video over the
Internet.
SRTP (Secure Real-time Transport Protocol) is an extension of RTP to provide encryption,
message authentication and integrity for the audio and video streams.
All devices must support SRTP to establish a connection. It is therefore possible to set
SRTP as disabled, optional or mandatory.
RTCP can be used to control the RTP session. When using SRTP, the same securityrelated features are added to the control protocol. SRTCP (Secure Real Time Control
Protocol ) can be either encrypted or not encrypted.
SRTP If set to disabled, the media is sent using RTP. Note that despite
this setting, the Avaya B179 will still use a secure channel if the
opposite device demands it.
If set to optional or mandatory, a padlock will be shown in the
bottom right-hand corner of the screen. If the other devices support SRTP, the padlock will be locked. Otherwise, an open padlock
will be displayed.
If set to mandatory, the call will not be connected if the other
devices do not support SRTP.
SRTCP Is automatically used when SRTP is enabled. Can be set to use
encryptation or not.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Secure signalling The SIP messages (signalling) and the SRTP cipher key are sent
on a different channel than the media and are not affected by the
RTP/SRTP setting. To ensure a secure connection, the signalling
must be secured using TLS or SIPS, see page 20. Note that the
SIP transport setting must be set accordingly.
On phone: > SETTINGS > ADVANCED > (PIN) > MEDIA > SECURITY (6,2,4,4).
VAD
Voice Activity Detection (speech detection) is a technique used in speech processing to
detect the presence or absence of human speech in regions of audio. In VoIP applications,
VAD is mainly used to avoid unnecessary coding and transmission of silence packets,
saving on computation and network bandwidth.
On phone: > SETTINGS > ADVANCED > (PIN) > MEDIA > VAD (6,2,4,2).
DTMF
DTMF (Dual-tone multi-frequency) signalling is used for telephone signalling over the line
to the phone switch or PBX.
If the device itself generates the tones and they are sent in the voice-frequency band, the
method is called Inband. This is not the best method when using VoIP. Low bit rate codecs may corrupt the signalling tones and make it difcult for the switch to identify them.
RFC 2833 is a method of carrying DTMF signals in RTP packets using a separate RTP
payload format. With this method a PSTN gateway reproduces the DTMF tones sent from
the end device.
With SIP Info the DTMF signals are sent as SIP requests. The SIP switch creates the
tones if the call is transferred to the PSTN.
Use RFC 2833 or SIP Info as preferred methods. Switch to inband only if you encounter
problems using DTMF signalling with your PBX/SIP switch.
On phone: > SETTINGS > ADVANCED > (PIN) > MEDIA > DTMF SIGNALLING (6,2,4,3).
Advanced
First RTP port If the RTP packets must be directed to a specic port series, the
rst port number is set here.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
LDAP
Select Settings > LDAP.
Avaya B179 has support for an external phone book, which means it can communicate
with a directory server using LDAP (Lightweight Directory Access Protocol). The built in
search function dynamically lters the content from the LDAP database, based on the
search characters the user enter.
To make the LDAP phone book available, the administrator has to activate and congure
the LDAP feature.
Enable LDAP The LDAP feature is disabled by default because it has to be
congured.
Name lter Denes how the entered search characters are used. The lter is
designed conforming to the string representation of LDAP search
lters described in RFC2254. The character % in the lter string
will be replaced with the search character entered by the user.
Example:
(|(s n =%* ) (cn=%*)) - All entries with the search characters in the
beginning of the sn OR cn attribute are presented to the user.
Server URL The IP address of the LDAP server host. Supports ldap and ldaps.
Search base The DN (distinguished name) of the search base
Example:
dc=domain, dc=com.
Username Leave this eld blank if the LDAP server does not require a username.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Password Leave this eld blank if the LDAP server does not require username
and password.
Max hits The maximum number of hits to return for each LDAP search.
Display name Species how the search hits shall be presented on the display in
Avaya B179.
Example:
%cn - shows the cn attribute.
%givenName %sn - shows the givenName attribute and the sn
attribute with a space in between.
Sort results Sorts the search hits based on the Display name.
Number attributes Here you dene the attributes that shall be displayed for a selected
search hit.
Example:
mobile telephoneNumber - shows the mobile phone number and
ofce phone number on separate rows for the selected Display
name. (Refer to the LDAP administrator for the actual names of the
elds in the LDAP database.)
Country code By entering the country code where the phone is located, the coun-
try code in any phone number attribute is ignored, if it is identical.
Area code By entering the area code where the phone is located, the area
code in any phone number attribute is ignored, if it is identical.
External prex If a special prex is needed to dial external numbers, it should
be added here. Use this if you for example need to dial 0 to get a
dialing tone.
Min length for external prex
Restricts the external prex to be added only if the phone number
is longer than the min length. This makes it possible to use short
internal numbers.
Exact length for no external prex
The external prex in not added if the phone number has exactly
the entered length.
Number prex for no external prex
All numbers that starts with this number will not have the external
prex added. Useful if you know that all internal numbers start
with a certain number.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
WEB INTERFACE
Select Settings > Web interface.
The web server in the Avaya B179 supports secure connections using HTTPS.
Enable HTTPS Set Enable HTTPS to On if you need a secure communication
between the PC used for setup and the phone.
Certicate To use HTTPS you need to upload a certicate to the phone.
On phone: > SETTINGS > ADVANCED > (PIN) > WEB INTERFACE (6,2,7).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
TIME & REGION
Select Settings > Time & Region.
Time
Enable NTP NTP (Network Time Protocol) is a protocol for distributing the
Coordinated Universal Time (UTC) by means of synchronizing the
clocks of computer systems over packet-switched, variable-latency
data networks.
Time This eld shows the actual time if NTP is enabled. Otherwise enter
the correct time (hh:mm:ss) and save the setting.
Date This eld shows the actual date if NTP is enabled. Otherwise enter
the correct date (yyyy-mm-dd) and save the setting.
Timezone Select the UTC time zone in your country.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
Daylight saving Select the Yes radio button if DST (Daylight Saving Time or
Summer Time) is currently used in your country. Note that this
setting only adjusts the time by one hour and does not change the
time automatically when the DST starts and ends.
NTP Server The NTP pool is a dynamic collection of networked computers that
volunteer to provide highly accurate time via NTP to clients worldwide. These computers are part of the pool.ntp.org domain and
part of several subdomains divided by geographical zones. They are
distributed to NTP clients via round robin DNS.
On phone: > SETTINGS > ADVANCED > (PIN) > TIME (6,2,5).
Region
Select the region where you are. This setting determines the signalling (disconnect tone,
busy tone, etc).
On phone: > SETTINGS > ADVANCED > (PIN) > REGION (6,2,6).
Daylight saving
Enable DST Select the Yes radio button if DST (Daylight Saving Time or
Summer Time) is used in your country.
DST Timezone Select the offset from UTC time when daylight saving is in use.
DST Mode When set to Automatic, OmniTouch 4135 IP uses dates stored
in the phone to adjust for DST. When set to Manual, you need to
manually set the offset two times a year.
Start/Stop xed date Set to Yes if DST changes the same date every year in your coun-
try. Then select the time and date it changes.
Set to No if DST changes a specic week and day each eyar. (For
instance third sunday in March.) Then select the month, week and
time it changes.
PROVISIONING
See “PROVISIONING – UPGRADE AND CONFIGURATION” on page 40.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
SETTINGS
SYSTEM
Select Settings > System.
Application restart
The Restart button restarts the phone application. This takes less than 30 seconds.
On phone: > SYSTEM > RES TAR T ( 7, 1 ).
System reboot
The Reboot button reboots the conference phone. The starting procedure may take about
two minutes.
On phone: > SYSTEM > REBOOT ( 7, 2).
Factory reset
The Reset button resets the Avaya B179 to factory default settings. All personal settings,
including account information, are erased.
On phone: > SYSTEM > FA CT ORY RE SE T ( 7, 3 ).
Hard reset to factory settings
See page 39 about resetting the phone if you have forgotten the Admin PIN code.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
HEADSET AND PA INSTALLATION AND SETTINGS
CONNECTING A WIRELESS HEADSET
Connect the headset to the Aux port on Avaya B179.
The microphones from the Avaya B179 and the wireless headset will work simultaneously
and transmit the call to other participants in the phone conference.
Please refer to the headset manual for further information.
Turning off the internal speakers when using a headset
The internal speakers can be turned off temporarily if you wish to use the Avaya B179 as
a personal telephone with a headset.
During a call, select > HEADSET.
Select YES when asked “SPEAKER OFF?”.
The speakers come on automatically when the call is ended.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
HEADSET AND PA INSTALLATION AND SETTINGS
CONNECTING A PA INTERFACE BOX
The Avaya B179 can be connected to an external PA system using a PA interface box.
To amplier/speakersTo mixer/microphone2.5 m connection cable
PA interface box
AUX port
Always disconnect the power supply from the
electrical outlet before disconnecting or conne cting equipment to the Avaya B179.
Connect the PA-box to the AUX port on Avaya B179 with the included cable.
Connect the external amplier to the RCA connector marked with a speaker.
Connect the microphone mixer to the RCA connector marked with a microphone.
Changing the auxilary port setting
Select Settings > Basic.
Select PA under the heading Auxilary port to aktivate the functions for external micro-
phones and speaker system.
Click on Save.
Do not select the PA option unless a PA system is connected. This option turns off the
internal microphone and internal speakers as default.
On phone: > SETTINGS > BASIC > AUX PO RT (6 ,1,5).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
HEADSET AND PA INSTALLATION AND SETTINGS
PA SETTINGS
To match several types of situations and equipment, there are some settings available in
the Avaya B179 menu.
Activating internal microphone and speakers
These settings are not available via the web interface.
Select > SETTINGS > BASIC > PA (6,1,6).
Select INTERNAL MIC and press OK to switch between on (shaded box) and off.
To ensure maximum audio quality, do not use the internal microphone and external
microphones connected via the PA box at the same time.
Only the internal microphone is turned off. Any external microphones connected to the
Avaya B179 are still turned on.
Select INTERNAL SPKR and press OK to switch between on (shaded box) and off.
To ensure maximum audio quality, do not use the internal speakers and external
speakers connected via the PA box at the same time.
Adjusting microphone volume from PA
During a call, select > PA > PA MONITOR.
Adjust the microphone volume from the mixer so that the level on the display screen is
around 10–12 when speaking in a normal tone.
Adjusting PA calibration manually
It is possible to calibrate the duplex performance of the conference phone when it is connected to a PA system. The calibration level can be set automatically by the Avaya B179
or adjusted manually to any value between 0 and 5 (0 being full duplex).
• Increase the calibration if the other party experiences disturbing echo.
• Decrease the calibration if the other party experiences low duplex, i.e. your voice is
muted or clipped when the other party is speaking.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
HEADSET AND PA INSTALLATION AND SETTINGS
The position of the PA system’s microphones and speakers and the amplier’s settings
may affect full duplex performance.
Select > PA > CALIBRATIO N.
AUTO is the default setting and is recommended in most cases. The gure shown in
brackets is the measured calibration value.
Select different levels and compare the audio quality to achieve your preferred setting.
NB. You must ask the person you are calling to assess the effect of the adjustments
you make.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
HARD SYSTEM RECOVERY
Reset conguration
If you have forgotten the Admin PIN code, the only way to reset it to default is to do a
hard factory reset. This is the same as the Factory reset in the system menu ( > SYSTEM
> FACTORY RESET).
This erases all settings including account information and contacts!
Disconnect the power supply cable. Note that this is the same as the network cable if
the phone uses Power over Ethernet.
Press and hold the button while you connect the cable again (i.e. starts the Avaya
B179). Hold the button pressed until the SYSTEM RECOVERY menu is shown on the
display.
You can press any other button than 1, 2, or 3 to start the phone without resetting.
Press 1 to select Reset conguration and conrm with OK.
Upgrade to the latest version of the software when the phone has started and redo the
setup of account and other settings (see page 7).
Restore rmware
This replaces the current software with the one supplied with the phone. All settings are
erased.
Export a conguration le (page 47) if local settings should be saved and export the
contacts (page 53) if these are stored in the unit.
Disconnect the power supply cable. Note that this is the same as the network cable if
the phone uses Power over Ethernet.
Press and hold the button while you connect the cable again (i.e. starts the Avaya
B179). Hold the button pressed until the SYSTEM RECOVERY menu is shown on the
display.
You can press any other button than 1, 2, or 3 to start the phone without restoring the
rmware.
Press 3 to select Restore rmware and conrm with OK.
All content in the phone’s memory is erased and the rmware supplied with the phone is written to
the memory.
Upgrade to the preferred version of the rmware when the phone has started.
Import the local conguration (page 47) and previous contacts (page 54) or do a
manual account setup (page 7).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
FIRMWARE UPGRADE ON A SINGLE PHONE
The easiest way to upgrade the Avaya B179 is via a computer connected to the same
network. Via the web interface, you can check for a more recent version and then automatically install it.
It is also possible to download the latest version, via the Avaya website (support.avaya.
com), and then install the le via the web interface or using a SD card.
Using the web interface
Select Settings > Provisioning.
Click on the Check Now button.
Compare the latest software version with the current version (shown on the same
page).
If you choose to upgrade, select version in the list box and click on the Upgrade button.
The browser window and the Avaya B179 display shows that the upgrading has begun.
The download and installation can take several minutes. Do not interrupt the upgrade
and do not disconnect plugs to the Avaya B179 during the upgrade. Interrupting the
upgrade may render the conference phone inoperable.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
When installation is complete, the text “Upgrade Complete. The unit will be reboot-
ed.” is shown in your browser, and after a while you hear the Avaya music signature,
which indicates that the conference phone has started.
Upgrading from downloaded le
It is possible to download a rmware le from support.avaya.com and install it on the
Avaya B179 from the local hard disk.
Download the rmware le from support.avaya.com.
Click on the Browse… button and locate and select the downloaded le.
Click on the Upgrade button.
Upgrading from SD card
Upgrading from SD card may be suitable if you have many phones to upgrade. The phones
do not have to be connected to the network.
Download the latest rmware as above and save it on a SD card.
Put the SD card in the phone you want to upgrade.
Disconnect the power supply cable. Note that this is the same as the network cable if
the phone uses Power over Ethernet.
Press and hold the button while you connect the cable again (i.e. starts the Avaya
B179). Hold the button pressed until the SYSTEM RECOVERY menu is shown on the
display.
You can press any other button than 1, 2, or 3 to start the phone without any change.
Press 2 to select SD-card upgrade.
The Avaya B179 is upgraded with the rmware le on the SD card and starts when the upgrade is
done.
After upgrading
If DHCP is used in the network, the IP address may have been changed. If the web
browser loses contact with Avaya B179, check the IP address on the conference phone
(see “USING THE WEB INTERFACE” on page 6).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
USING A CONFIGURATION FILE
It is possible to save a conguration xml le to be used as:
• Backup (i.e. if the system has been reset to factory default)
• Conguration interface (there are some settings that are not congurable via the web
interface)
• Management tool (export, edit and import settings to a set of phones instead of doing
the settings on each phone)
• Use with a Device Management server, see page 48.
The structure of the xml le is as follows:
<locale>
<region>
<recording>
<enable>
<logging>
<level> The phone application logs me ssages to log facility LOCAL0.
<log_sip> Log SIP message s to log facility LOCAL1. Default is true.
<remote_log> Log message s to a remote log ser ver. Default is false.
<remote_host /> Remote log server.
<network>
<net>
<dhcp> Specify if DHCP should be used to obtain network settings. If so, the other
<ip> Specify the IP address of the Avaya B179.
<netmask> The netmask of the IP address.
<gateway> Specify the default gateway to be used.
<dns1> Specify at most two Domain Name Servers to be used.
<dns2>
<hostname> Specify host name.
<domain /> Specify domain name.
<vlan>
<enable> Virtual LAN enabled if set to true
<id> VLAN ID.
<s td_prio_map>
<sip_ priority >
<media_priority>
<ether_8021x>
<enable>
Log level 1–5 (equivalent to Fatal–Trace)
network settings won’t be used.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
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<username />
<eap_md5>
<enable>
<password>
<eap_tls>
<enable>
<password>
<qos>
<dscp_sip>
<dscp_media>
<time>
<ntp>
<timez one>
<daylight _save>
<ntps>
<sip>
<udp_transport> Specify if UDP shall be used as transport.
<udp_port> Specify the UDP port to listen to.
<tcp_transport> Specify if TCP shall be used as transport.
<tcp_port> Specify the TCP port to listen to.
<tls_transport> Specify if TLS shall be used as transport.
<sips_transport> Specify if SIPS shall be used as transport.
<tls_port> Specify the TLS port to listen to.
<rtp_port> Specify the start port for RTP trafc.
<outbound_proxy /> Specify the URL of outbound proxies to visit for all outgoing requests. The
<use_stun> Use Simple Traversal of UDP through NATs (STUN) for NAT traversal.
<stun_domain /> Specify domain name to be resolved with DNS SRV resolution to get the ad-
<stun_host /> Specify STUN server to be used in ”HOST[:PORT]” format. If port is not
<use_turn> Use Traversal Using Relay NAT (TURN) for NAT traversal. Default is no.
<turn_host /> Specify TURN relay server to be used.
<turn_tcp> Use TCP connection to TURN server. Default is false.
<turn_user /> TURN username.
<turn_passwd /> TURN password.
<nat_type_in_ sdp> Support for adding and parsing NAT type in the SDP to assist troubleshoot-
outbound proxies will be used for all accounts and will be used to build the
route set for outgoing requests. The nal route set for outgoing requests will
consist of the outbound proxies and the proxy congured in the account.
Default is no.
dress of the STUN servers. Alternatively application may specify stun_host
and stun_ relay_ho st instead.
specied, default por t 3478 will be used.
ing. The valid values are:
0: no information will be added in SDP and parsing is disabled
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
1: only the NAT type number is added
2: add both NAT type number and name
<require_100rel> Specify whether support for reliable provisional response (100rel and
<use_srtp> Specify default value of secure media transport usage. Note that this setting
<srtp_ secure_signaling> Specify whether SRTP requires secure signalling. This option is only used
<codec>
< type> Codec type
<name> Codec name
<prio> Codec priority (0-4)
<dtmf> DTMF signalling. Default is 2.
<no_vad> Disable VAD. Default is VAD enabled.
<ec_tail> Echo c anceller tail length, in milliseconds.
<enable_ice> Enable ICE?
<enable_relay> Enable ICE relay?
<enable_presence> Enable the use of presence signalling.
<enable_sip_replaces>
<enable_blind_transfer>
<allow_contact_rewrite>
<tls>
<tls_password /> Password for the private key
<tls_method> TLS protocol method from pjsip_ssl_method, which can be:
<tls_verify_server> Verify server certicate.
<tls_verify_client> Verify client certicate.
<tls _require_ client_cer t> Require client ce rticate.
<tls_neg_timeout> TLS negotiation timeout in seconds to be applied for both outgoing and
PRACK) should be required by default. Note that this setting can be further
customized in account conguration.
can be further customized in account conguration.
0: SRTP will be disabled, and the transpor t will reject RTP/SAVP offer.
1: SRTP will be advertised as optional and incoming SRTP of fer will be
accepted.
2: The transport will require that RTP/SAVP media shall be used.
when use_ srtp option above is non-zero. Note that this setting can be
further customized in account conguration.
0: SRTP does not require secure signalling
1: SRTP requires secure transport such as TLS
2: SRTP requires secure end-to-end transport (SIPS)
0: In-band
1: SIP message
2: RTP message
0: Default (SSLv23)
1: TL S v1
2: SSLv 2
3: SSLv3
23: SSLv23
incoming connections. If zero, no timeout is used.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
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<account1>
<valid> If this account information is valid or not.
<name> User dened name of the account
<id> The full SIP URL for the account.
<registrar > This is the URL to be put in the request URI for the registration.
<publish_enabled> If this ag is set, the presence information of this account will be published
<initial_auth> If this ag is set, the authentication client framework will send an empty
<initial_algo /> Specify the algorithm to use when empty Authorization header is to be sent
<pidf_tuple_id /> Optional PIDF tuple ID for outgoing PUBLISH and NOTIFY. If this value is
<force_contact /> Optional URI to be put as Contact for this account. It is recommended that
<require_100rel> Specify whether support for reliable provisional response (100rel and
<proxy_uri /> Optional URI of the proxies to be visited for all outgoing requests that are
<reg_ timeout> Optional interval for registration, in seconds. If the value is zero, default
<cred> Array of credentials. Normally, if registration is required, at least one
<realm> Realm. Use ”*” to make a credential that can be used to authenticate any
< scheme /> Scheme (e.g. ”digest”).
<username> Authentication name
<cred_data _type> Type of data (0 for plaintext password).
<cred_data> The data, which can be a plaintext password or a hashed digest.
<auto_update _nat> This option is useful for keeping the UDP transpor t address up to date
<ka_interval> Set the interval for periodic keep-alive transmission for this account. If this
<ka_data /> Specify the data to be transmitted as keep-alive packets. D efault: CR-LF.
<use_srtp> Specify whether secure media transport should be used for this account.
to the server where the account belongs.
Authorization header in each initial request.
for each initial request (see above).
not specied, a random string will be used.
this eld is left empty, so that the value will be calculated automatically
based on the transport address.
PRACK) should be required for all sessions of this account.
using this account (REGISTER, INVITE, etc).
interval will be used.
credential should be specied to successfully authenticate the service
provider. More credentials can be specied, for example when it is expected
that requests will be challenged by the proxies in the route set.
challen ges .
with the NAT public mapped address. When this option is enabled and
STUN is congured, the librar y will keep track of the public IP address
from the response of REGISTER request. Once it detects that the address
has changed, it will unregister current Contact, update the UDP transport
address and register a new Contact to the registrar.
value is zero, keep-alive will be disabled for this account. The keep-alive
transmission will be sent to the registrar’s address after succes sful registration.
0: SRTP will be disabled and the transport will reject RTP/SAVP offer.
1: SRTP will be advertised as optional and incoming SRTP of fer will be
accepted.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
2: The transport will require that RTP/SAVP media is used.
<use_srtcp> Species whether encr yption or no encryption is used for the control
<srtp_ secure_signaling> Specify whether SRTP requires secure signalling. This option is only used
<account2>
Same as above for account 2
<provisioning>
<upgrade>
<url> Place to nd software upgrades. The supported URL types are: HTTP, FTP,
<dev_mgnt>
<enable> Device management enabled, true or false.
<use_dhcp_option> Use DHCP option for DM server address.
<dhcp_option> Specication of which DHCP option to use.
< le_server_address> DM server address if not provided by DHCP option.
<pagename /> Base name of conguration les to download
< type /> Conguration le type specication
<update_interval> Timing for downloading les. Shall be entered in crontab format:
<https_check_srv_cer t> Controls server cer ticate, true or false.
<https_protocol> Possibility to set https protocol if open-ssl auto detection fails.
<www>
<enable_https> Secure communication to the Avaya B179 web server. Default is false.
<pa>
<enable_pa> PA enabled, true or false
<enable_internal_mic> Internal mic enabled when PA set to true.
<enable_internal_ spkr> Internal speakers enabled when PA set to true.
<calibration> Calibration value. Note that 0 is auto, 1 is calibration value 1, 2 is calibra-
<ldap>
<enable> LDAP enabled, true or false.
<name_lter> Name lter according to RFC2254
<server_url> LDAP server address
<search_base> The DN (distinguished name) of the search base
<username />
protocol. Valid values: true or false.
when use_ srtp option above is non-zero.
0: SRTP does not require secure signalling
1: SRTP requires secure transport such as TLS
2: SRTP requires secure end-to-end transport (SIPS)
and TFTP.
* * * * * where the * stands for minute (0–59), hour (0–23), day of month
(1–30), month (1–12), day of week (0–7) (Sunday =0 or 7)
Example:
0 6 * * * = the les are downloaded daily at 6:00.
tion value 1, etc.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
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<password />
<max_hits>
<country_code>
<area_code>
<external_prex />
<min_length_for_ext_prex />
<exact_length_for_no_ext_ prex />
<number_prex_for_no_ext_prex />
<number_attributes>
<display_name >
<sort_results>
Export conguration
Select Settings > Provisioning.
Click on the Export button under Conguration.
The conguration le is shown in the web browser.
Choose to save the page as an xml le.
The xml le is as default saved in your folder for downloaded les.
If necessary, edit the xml le in a suitable editor.
Import conguration
Click on the Browse… button under Conguration.
Select the xml le and choose to open it.
Click on the Import button.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
USING A DEVICE MANAGEMENT SERVER
Using Device management facilitates the upgrading and conguration of multiple conference phones. To use this feature, the Device management needs to be enabled (default)
and congured and the appropriate les must be located on a server reachable from all
phones, here called a device management server.
The conguration and rmware download are controlled with a congurable frequency.
The default value is once every 30 minutes. (Note: The interval can only be edited directly
in the conguration le.)
Conguration priorities
Because the same conguration parameters can be entered in multiple locations, there is
a need for priorities. The local conguration les have the highest priority followed by the
global conguration le. Conguration entered on the unit itself, via the web interface or
directly on the phone, is overridden the next time the conguration les are downloaded.
Note one exception. Phone language entered on the unit will take precedence.
Files on the device management server
Global conguration le
The global conguration le contains the basic conguration – all settings that are
common for all conference phones on your location. The easiest way to create this le
is simply to congure one phone and export the conguration le or use the built in
conguration le creator.
The default name for this le is avaya.xml, but it is possible to create a custom name by
using the pagename element in the conguration le. It is also possible to refer to a cgi,
php, asp, js or jsp le, instead of the xml le, if this is declared using the type element in
the conguration le.
Avaya B179 searches for conguration les in the following order:
"
%"%
4
5 E
:
=
E %
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
Local conguration le
The local conguration le contains conguration parameters that are unique for every
conference phone. The settings in this le takes precedence over the settings in the global
conguration le.
The default name for this le is avaya-<MAC>.xml, where <MAC> is the MAC address of
the specic conference phone. The MAC address should be written without colons.
It is possible to create a custom name by using the pagename element in the conguration le. It is also possible to refer to a cgi, php, asp, js or jsp le, instead of the xml le,
if this is declared using the type element in the conguration le.
Avaya B179 searches for conguration les in the following order:
"
4
5
:
=
Firmware binary
Contains the rmware binary that will be downloaded and installed by Avaya B179 if the
metadata le shows that this is a newer version than the present installed. The binary le
can be downloaded from support.avaya.com.
Firmware metadata le
A metadata le in xml format with information of the rmware version in the binary le.
The le is used to check if the binary le should be downloaded to the phone or not.
The name of this le shall be avaya_fw_version.xml. The le shall contain the following
elements in xml format.
<rmware _version>
<version>X.X.X </version> Eg. 2.3.0
< lenam e>xxx x </ len am e> Eg . AVAYA_B179 _v 2.3.0 .k t
<checksum>XX XX < /checksum> MD5 checksum of the rmware binar y
</rmware_version>
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
Device management conguration in Avaya B179
Select Settings > Provisioning.
Device management
Enable On enables Device management.
Use DHCP option Set to on if you want to use DHCP option for DM server address.
DHCP option Select the DHCP option used for the DM server address.
242: Avaya specic option (default)
43: Vendor specic
56: DHCP message
60: Class Id
61: Client Id
66: Server-name
67: Bootle-name
File server address DM server address if not provided by DHCP option.
HTTPS protocol Default is auto, but can be set to SSLv2 or SSLv3 if open-ssl auto
detection fails.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
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Check server cert. Enable authentication with certicate.
Certicate Here you can upload a certicate to the Avaya B179 to be used for
authentication when using Device management.
Root certicate The public key in the root certicate is used to verify other certi-
cates when using Device management.
Private key Here you can upload a private key to the Avaya B179 to be used for
authentication when using Device management.
Read more about certicates on page 22.
Setting up a Device management server
This is a description of a manual method to create the conguration les.
Select Settings > Provisioning.
Enable Device management and enter the server information.
Creating a global conguration le
Congure one phone with the basic conguration.
Click on Export to create a conguratuion le.
If necessary, edit the xml le in a suitable editor.
Some parameters can’t be entered via the web interface (update frequency, page-
name, and letype).
To avoid confusion, it may be wise to delete the local information from the le (eg.
account information).
Save the le with the name avaya.xml on the File server address specied above.
Creating local conguration les
Save a copy of the conguration le for each conference phone on your location with
content only in the elements that shall be unique for each conference phone (eg.
account information).
The default name for each le is avaya-<MAC>.xml, where <MAC> is the MAC address
of the specic conference phone.
Place the conguration les on the File server address specied above.
Firmware binary
Place the rmware binary le on the Provisioning server.
Create a Firmware metadata le according to page 49 and place it on the File server
address specied above.
Depending on the server used, and the security settings, there might be necessary
to add the le type .kt to the MIME settings on the server. This is easily checked by
trying to download the kt le from a web browser.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
PROVISIONING – UPGRADE AND CONFIGURATION
UPGRADING IN IP OFFICE
Avaya B179 supports the IP Ofce check-sync message with le id 4 for rmware
upgrades. Please see the IP Ofce documentation for more information.
HOW TO DO A DOWNGRADE
The only way to “downgrade” – install a previous version of the rmware - is to restore the
rmware to factory default (page 39) and then install the preferred rmware.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
MANAGING PHONE BOOK AND CONF. GUIDE
IMPORTING AND EXPORTING CONTACTS
You can import contacts from a comma separated values (CSV) le. One way of creating a
CSV le is using Microsoft Excel and saving the le in CSV format.
Enter the names of the contacts in the rst column and their phone numbers or URIs in
the second. Do not use hyphens or spaces in the number. Note that Excel ignores zeros at
the beginning of numbers. The cells must therefore be formatted as text.
It is normally possible to export contact books stored in your PC in CSV format.
The way the number can be written may depend on the SIP PBX being used, but
normally you can use:
• Complete phone number, including country code
• Phone number, including area code
• Local phone number only
• Internal speed dial number (with company’s own PBX)
• URI, e.g. sip:user@company.com
• URI with IP address, e.g. sip:10.10.1.10 0 (within a local network)
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Avaya B179 SIP Conference Phone Installation and Administration Guide
MANAGING PHONE BOOK AND CONF. GUIDE
Importing contacts
Select Phone Book.
Click on the Scroll… button under the heading Import in the web window.
Open your CSV le.
Click on Import.
The name is limited to 15 characters, since the Avaya B179 screen cannot display
more than 15 characters.
Exporting contacts
You can export your contacts as a CSV document in order to import them into another
phone.
Click on Export.
Save the document.
IMPORTING AND EXPORTING CONFERENCE GROUPS
The Conference group feature requires that your system allows multiple call appear-
ances. Avaya Aura Communication Manager supports this while Avaya IP Ofce and
Communication Server 1000 do not.
The conference groups can be imported and exported in the same way as the contacts in
the phone book, but use a three column csv instead of a two column csv.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CHECKING STATUS AND LOGS
The tabs under Status show the settings on corresponding tabs plus device info and logs.
DEVICE
Select Status > Device.
The Device tab shows phone information including serial number, network port and current software version.
On phone: > STATUS > DEVICE (8,6).
NETWORK
Select Status > Network.
On phone: > STATUS > NETWORK (8,2).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CHECKING STATUS AND LOGS
TIME & REGION
Select Status > Time & Region.
On phone: > STATUS > TIME (8,5).
SIP
Select Status > SIP.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CHECKING STATUS AND LOGS
Account 1 and Account 2
On phone: > STATUS > ACCOUNTS (8,1,1 an d 8,1, 2).
NAT traversal
On phone: > STATUS > NAT TRAVERSAL (8,3).
Transport
On phone: > STATUS > ACCOUNTS> TRANSPORT (8,1, 3).
MEDIA
Select Status > Media.
On phone: > STATUS > MEDIA (8,4).
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CHECKING STATUS AND LOGS
LOG
Select Status > Log.
The Log tab contains ve types of log messages and can be useful for trouble shooting.
Select the log you want to review and click on the Change button.
The Refresh button adds all new messages sent since the present log was chosen.
Application log
This shows the phone application messages. The log can be ltered from “Fatal” (only the
fatal error messages) to “Trace” (all messages).
The Clear log button erases all content in the log.
SIP Trace
The SIP Trace logs the communication between the phone and the SIP PBX.
The Clear log button erases all content in the log.
It is possible to disable the SIP trace log
Select SIP logging Off and click on the Set button.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
CHECKING STATUS AND LOGS
System log
Shows the phone system messages.
Device management log
Logs the device management activities.
Upgrade log
Logs the upgrade procedure.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
TECHNICAL DATA
Size Diameter 240 mm, height 77 mm
Weight 1 kg
Color Liquorice black
Display screen Illuminated graphics (LCD), 128x64
Keypad Alphanumerical 0–9, *, on, off, mute, hold, volume up, volume
down, 5 buttons for menu navigation, line mode, conference
guide
Anti-theft protection Kensington security slot
Memory Support for SD and SDHC memory cards
(Recording capacity 35h/GB)
Connectivity
Network connection Modular 8P8C (RJ45), Ethernet 10/100 Base T
Power supply AC adapter 100–240 V AC/14 V DC
IEEE 802.3af Power over Ethernet, Class III.
Extra microphones 2 modular 4P4C
Auxiliary Modular 4P4C for wireless headset
Network and communication
Network addressing DHCP and static IP
NAT traversal STUN, ICE and TURN
Connection protocol SIP 2.0 (RFC 3261 and companion RFCs)
Transport UDP, TCP, TLS and SIPS
Security SRTP and TLS
Quality of Service DiffServ, VLAN and 802.1x
Audio support Codecs: G722, G711 A-law, G711 µ-law, G729ab
DTMF tone generation RFC, SIP INFO, In-band
Time servers NTP and SNTP
Daylight saving: Congurable for automatic adjustments.
Conguration Via integrated web server
Conguration and provisioning
Conguration Via integrated web server, HTTP or HTTPS
Separate user and administrator login for secure conguration.
Device management Support for device management for easy conguration and
updating of multiple conference phones.
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Avaya B179 SIP Conference Phone Installation and Administration Guide
TECHNICAL DATA
Directory
Internal phone book 1,000 entries per prole
(4 password protected proles)
Export/import of directory
Call list
External directory Support for LDAP
Sound
Technology OmniSound® Wideband
Microphone Omni-directional
Reception area Up to 30 metres2, >10 people
Speaker Frequency band 200–7000 Hz,
Volume 90 dB SPL 0.5 m
Equalizer Three pitches: soft, neutral, bright
Environment
Temperature: 5°–40°C
Relative humidity: 20-80% condensation free
Recommended acoustic conditions:
Reverberation period: 0.5 S Rt 60
Background noise: 45 dBA
Approvals
Electrical safety EN 60950-1:2006,
ANSI/UL 60950-1-2002,
CAN/CSA-C22.2, No. 60950-1-03
EMC/Radio EN 301 489-3 V1.4.1 (2002-08),
EN 301 489-1 V1.6.1 (2005-09),
FCC Part 15 subpart B class A,
FCC Part 15 subpart C,
EN 300220-1:2000,
EN 300220-2:2000
RoHS
61
110231- 61-001 R ev 4 C
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