GAI-Tronics VoIP Telephone User Manual
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GAI-TRONICS
A division of Hubbell Ltd
VoIP Telephones
Configuration Guide:
Firmware version 3
(The previous versions of this manual, for firmware versions 1 & 2, remain available)
Document Ref: 502-20-0119-001 Issue 3. Apr 2012. CN33927-001
GAI-TRONICS
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VoIP Telephones
Configuration Guide
Firmware version 3
CONFIDENTIAL
The contents of this publication are confidential, are the property of GAI-Tronics, and may not
be reproduced, wholly or in part, without their written permission.
TRADEMARKS and LICENCES
Windows is a trademark of Microsoft Corporation, registered in the United States and other
countries.
All other product and brand names are trademarks of their respective owners.
Software licences and notices are available on the GAI-Tronics website at www.gai-
tronics.co.uk/voipsupport.htm
POLICY
The policy of GAI-Tronics is one of continual development and improvement of products and
we reserve the right therefore to alter specifications without notice.
GAI-Tronics
Brunel Drive
Stretton Park
BURTON-UPON-TRENT
Staffordshire
England
DE13 0BZ
Tel.: +44 (0)1283 500500
Fax.: +44 (0)1283 500400
www.gai-tronics.co.uk
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Contents
1.Introduction ......................................................................................................................... 5
2.What's new ? ...................................................................................................................... 5
2.1New in Version 3 ........................................................................................................ 5
2.2New in Version 2 ........................................................................................................ 6
3.How the product is intended to work .................................................................................. 6
3.1Operating Sequence. ................................................................................................. 7
3.2Dictionary of terms ..................................................................................................... 7
4.Setting up and Configuring the telephones. ....................................................................... 9
4.1Quick Start ................................................................................................................. 9
4.2Frequently Asked Questions (FAQs) ......................................................................... 9
4.2.1What network facilities do I need to provide? ...................................................... 10
4.2.2How do I set up dialling and memory lists? ......................................................... 10
4.2.3Can I set the phone to make calls without a proxy (ie peer-to-peer)? ................. 11
4.2.4How do I set up Real-time alarm reporting via email or syslog? ......................... 11
4.2.5How can I set up an external beacon to flash when the phone is ringing? ......... 12
4.2.6How do I set up a door-entry system? ................................................................. 13
4.2.7How can I use the phone to make paging or PA announcements? .................... 14
4.2.8What additional features are available with CMA? .............................................. 14
5.Web pages in detail .......................................................................................................... 14
5.1Login ........................................................................................................................ 15
5.2Home Page .............................................................................................................. 16
5.3IP settings ................................................................................................................ 17
5.3.1Note about Syslog: .............................................................................................. 18
5.4SIP settings .............................................................................................................. 19
5.4.1SIP Info sub-pages: ............................................................................................. 21
5.5Unit settings ............................................................................................................. 22
5.5.1Audio Path Test ................................................................................................... 24
5.6Access settings ........................................................................................................ 25
5.7Serial settings .......................................................................................................... 26
5.8Email settings .......................................................................................................... 27
5.9Clock settings .......................................................................................................... 28
5.10Dialling & Memories ................................................................................................. 29
5.10.1Memories sub-page ......................................................................................... 30
5.10.2Memory Lists sub-page. .................................................................................. 31
5.10.3Basic Info sub-page. ........................................................................................ 32
5.11Key mapping ............................................................................................................ 33
5.11.1Activating relays using DTMF codes ............................................................... 35
5.12Current status .......................................................................................................... 36
5.13Audio settings .......................................................................................................... 37
5.14Alarm settings .......................................................................................................... 39
5.15Tone settings ........................................................................................................... 42
5.15.1Suggested Tone Settings for Various Countries: ............................................ 44
5.16LED settings ............................................................................................................ 45
5.17Logic settings ........................................................................................................... 48
5.18Multicast settings ..................................................................................................... 50
6.Configuration File update ................................................................................................. 53
6.1Configuration File Syntax......................................................................................... 55
6.2Configuration File Commands ................................................................................. 56
7.Time Zone Table .............................................................................................................. 66
8.Example Configuration File .............................................................................................. 68
9.Command Line Interface .................................................................................................. 72
9.1CLI Syntax ............................................................................................................... 73
9.2ACCESS Module Command Line Syntax ................................................................ 74
9.3ALARMS Module Command Line Syntax ................................................................ 75
9.4KEY Module Command Line Syntax ....................................................................... 76
9.5LED Module Command Line Syntax ....................................................................... 76
9.6DIALPLAN Module Command Line Syntax ............................................................. 77
9.8CLOCK Module Command Line Syntax .................................................................. 78
9.9AUDIO Module Command Line Syntax ................................................................... 79
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9.10TONES Module Command Line .............................................................................. 79
9.11IP Module Command Line Syntax ........................................................................... 80
9.12LOCAL Module Command Line Syntax ................................................................... 80
9.13LOGIC Module Command Line Syntax ................................................................... 81
9.14SIP Module Command Line Syntax ......................................................................... 82
9.15SMTP Module Command Line Syntax .................................................................... 83
9.16Status Module Command Line Syntax .................................................................... 83
9.17UNIT Module Command Line Syntax ...................................................................... 84
9.18Multicast Module Command Line Syntax ................................................................ 85
10.Reset to Defaults ......................................................................................................... 86
11.Troubleshooting ........................................................................................................... 87
11.1Is the unit powered up? ........................................................................................... 87
11.2I can't access the web pages ................................................................................... 87
11.3I can't make calls ..................................................................................................... 87
11.4Calls connect but there is no speech (or sound is garbled) .................................... 88
12.Licensing Notices ......................................................................................................... 88
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1. Introduction
This guide provides information on the operation and configuration of GAI-Tronics' range of
rugged VoIP telephones with firmware version 3, released in January 2012.
There are significant changes to some of the web pages and commands from those in
previous versions. Issue 1 and issue 2 of this manual will remain available on the GAITronics UK website (www.gai-tronics.co.uk/voipsupport.htm
versions.
The firmware version of each unit is displayed at the bottom of its home web page, and as
part of the welcome message following login via a Telnet or serial connection.
In each case the firmware version is a series of 3 numbers separated by dots (periods). The
main firmware version is the first number. For example:
Upgrading to the latest version is possible in most circumstances, but please note that certain
new features may not be enabled on upgraded phones - contact GAI-Tronics for details.
GAI-Tronics VoIP telephones are available in a variety of model styles, including handset and
hands-free models, but the programming and configuration methods are common to all.
Please note that the features may depend on the model type, and that therefore this guide
may describe features not available on the particular model being configured.
Features of the GAI-Tronics range of VoIP telephones include:
• SIP compatible (RFC3261) only
• Registration with multiple SIP proxies (new in v2)
• Configurable via web pages, serial link or downloading a configuration file
• Outgoing cascading call lists
• Real-time alarm reporting via email or Syslog
• 4 auxiliary inputs, 2 volt-free contact outputs (revised in v2)
• Remote operation of contacts ("door opening" function)
• 3 “autoanswer” modes, including paging mode (revised in v2)
• Compatible with GAI-Tronics' Call Management Application (CMA)
This guide does not include information on:
• Installation, cabling and connections (see guide 502-20-0115-001 for non ATEX
phones and 502-20-0133-001 for Auteldac4 VoIP)
• Setting up, configuring and operating a network for VoIP. Please ensure that the
network is configured to allow VoIP communications (using the SIP protocol) between
the desired locations before attempting to configure GAI-Tronics telephones.
) as a reference for earlier
1.2.13 indicates firmware version 1
2.1.6 indicates firmware version 2.
3.0.0 indicates firmware version 3.
2. What's new ?
2.1 New in Version 3
Version 3 added the following features from version 2:
Acoustic Path Testing (APT)
APT allows remote testing of handsets, microphones and speakers. APT can be used to
verify that a phone is functioning acoustically. The test can be run on demand or on a
scheduled basis, reporting its results via Syslog and / or email. See section 5.5.1.
Multicast
(Only applicable to hands-free products). Multicast allows a single audio stream to be
received by multiple endpoints simultaneously, to achieve multi-point paging or Public
Address functionality over IP. (Requires a multicast compliant SIP server). 8 definable
multicast address ranges, with individual priority levels, for zoning. Assignable relay outputs
and splash tones. See section 5.18.
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Reset to factory defaults
The unit can be restored to factory defaults by pressing an internal reset button on the PCB.
This can be used to recover a phone where the user name or password have been lost. See
section 10.
Relay operation from DTMF tones
For door opening purposes, output relays can now be triggered by the receipt of a
programmable DTMF tone sequence. This means that a phone's relays could potentially be
triggered from any phone on the system. See section 5.11.1.
Default autoanswer mode
The unit can now be set to automatically answer an incoming call, with a choice of 3 different
autoanswer modes to suit applications such as paging and intercom. See section 5.5.
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2.2 New in Version 2
Version 2 firmware added the following features from version 1:
Multiple SIP proxies
The unit can now hold up to 4 alternate addresses for the SIP proxy and registrar with a
prioritised failover sequence between them. This means that if it fails to register with the first
server it will attempt to do so with the next and so on.
The unit can be set to automatically refresh its registration at a predetermined interval to
ensure that registration is maintained at all times (or if not raise an alarm).
This provides a high degree of resilience across the network and reduces the possibility of a
single point of failure jeopardising the operation of the whole system.
Additional functions for Relay outputs and LEDs
In addition to the functions in version 1, the unit can now trigger its output relays and / or
LEDs on:
• PAGE (activated by PAGEMODE, see section 5.5). For example a relay could be used to
activate a public address amplifier, allowing the unit to be used as a mini PA.
• EMERGENCY (if an outgoing call is designated as an emergency call), where for
example a relay could be used to activate an emergency beacon, and
• REGISTERED, where for example an LED could indicate that the unit is available for use
(i.e. it can make a call).
Additional LED drive
Version 2 allows 3 programmable LEDs instead of 2. Note that the number of LEDs fitted
varies with model type. Some standard models have no LEDs fitted.
Page Mode
Auto-answer mode 3 is now explicitly referred to as PAGE MODE to highlight its potential use
as a PA or paging system. Functionally it is unchanged, except for the LED and relay triggers
described above.
3. How the product is intended to work
The VoIP telephone has been designed to mimic the behaviour of a traditional, analogue
telephone, specifically based on the GAI-Tronics range of rugged telephones, to give
continuity where VoIP and analogue units are used in similar situations.
Accordingly, traditional telephone terminology is used throughout the manuals and
documentation, and many of the features are designed to mimic analogue telephone
behaviour.
A major difference between analogue telephones and VoIP is that, with analogue units, most
signalling and tones such as ringing, dial tone, busy tone etc., are provided by a telephone
exchange (PABX), whereas the VoIP unit must generate these itself. The telephone provides
features to change the various tones to emulate those of different countries or PABXs, to give
familiar operation in its intended location.
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3.1 Operating Sequence.
Typical sequences of events for various model types are explained below:
Handset models (Titan, Commander)
Placing a call
• Lift handset (off hook)
• Dial tone in receiver
• Dial number - confidence tones in receiver
• Call progress tone in receiver (e.g. ring tone)
• Call is answered by remote party
• Normal voice call
• Replace handset (on hook)
• Call terminates.
Receiving a call
• Telephone rings
• Lift handset (off hook)
• Normal voice call
• Replace handset (on hook)
• Call terminates.
Hands-free models (VR, Help Point)
Placing a call
• Press button
• Dialling confidence tones heard from speaker (wake and dial)
• Call progress tone heard from speaker (e.g. ring tone)
• Call is answered by remote party
• Normal voice call
• Call terminates. (On hook)
Receiving a call
• Ringing heard from speaker
• Press any button to answer call (off hook)
• Normal voice call
• Call terminates. (On hook)
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3.2 Dictionary of terms
Busy tone
A tone played to the user to indicate that a call has failed because the called party is engaged
Call progress tone
One of a number of different tones played to the user to indicate the status of a call. Dial
tone, busy tone and NU tone are all examples of call progress tones.
Confidence tones
Tones played to the user to indicate that dialling is in progress, by imitating DTMF tones used
by analog telephones.
Dial tone
A tone played to the user to indicate that the telephone is ready to dial – ie it is off hook and
waiting for a button to be pressed to initiate a call.
Dialling
Used to describe the process of initiating a call, usually by pressing a memory button or a
series of digit buttons.
DTMF
Standing for “dual tone multi-frequency”, the dialling digit tones produced by a touch-tone
phone. Commonly used for signalling in analogue systems.
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Handset phone
Used to denote a telephone from the GAI-Tronics Titan or Commander product ranges, with a
separate handset attached to the main telephone body by a heavy duty flexible cord. No
separate loudspeaker is fitted to these models.
Hands-Free phone
Used to denote a telephone from the GAI-Tronics Help Point or Vandal Resistant product
ranges, with a microphone and speaker integrated into a flat panel. No corded handset is
fitted to these models.
LNR
Standing for “last number redial”, this is a button provided on some models of GAI-Tronics
phone to redial the last manually dialled number.
Memory dial number
On an analogue or cellular phone, memory numbers are pre-stored digit sequences used to
start calls. With VoIP these can also be URI’s rather than numbers, but are still referred to in
the same way.
Mute
A function to temporarily mute the microphone so that the remote party cannot hear. On GAITronics telephones this function is provided by the "S" button.
NU tone
Number unobtainable tone – used to indicate that a call cannot connect due to the end point
not being recognised.
Off hook
Used to denote the state of a telephone during an active call, or when a call has been
initiated. For a handset phone, off hook usually means that the handset is lifted.
On hook
Used to denote a telephone in the idle state – no call started or answered. A telephone is still
on hook when it is ringing on an incoming call. For a handset phone, on hook usually means
the handset is not lifted. If a call is terminated whilst the handset is still lifted (for example by
the CALL LIMIT timer), the telephone is placed into the on hook state. For a hands-free
phone, on hook means that no ON or WAKE & DIAL button has been pressed following a
terminated call or reset.
Recall
On analogue phones, the Recall button is used to activate exchange signal, usually to
transfer a call. The GAI-Tronics VoIP telephone does not have a recall facility, but the “R”
button (where fitted) can be used to activate an output on a remote phone, for example as a
door release.
Register Fail tone
A tone played to the user initiating a call to indicate that the telephone is not currently
registered with a registrar, meaning that a call cannot be made.
Ring tone
A tone played to the user initiating a call to indicate that the call has been placed but not yet
answered. This usually signifies that the remote end is ringing.
Ringing
A loud alert tone made by the telephone indicating that an incoming call is ready to be
answered.
Secrecy (mute)
A function to temporarily mute the microphone so that the remote party cannot hear. On GAITronics telephones this function is provided by the "S" button.
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Sidetone
On handset phones, part of the microphone signal is fed to the earpiece so that the user can
hear his or her own voice during the call. This makes it a more natural experience, and has
been a feature of analogue telephones since their invention. Not used on hands-free phones.
4. Setting up and Configuring the telephones.
Each telephone must be configured for use on the intended network. Most models have
memory-dial locations, which will need to be set up. The telephone also has a range of
customisable features. All of these can be set up using one of 4 different methods:
• Web pages (the simplest and quickest method for configuring an individual phone)
• Downloading configuration files (the most efficient method for multiple updates)
• Command-line commands via direct serial link
• Command-line commands via Telnet session
Note:
All the above access methods require you to know the unit's username and password.
All methods, except direct serial link, also require you to know the unit's IP address.
Please ensure these details are recorded securely once set or changed.
All of the telephone's features can be configured using any of the above methods, but the
most complete description of features is contained in the web page section (Section 5).
4.1 Quick Start
The factory defaults will generally be sufficient in most cases, but the following steps must be
taken as a minimum:
• Provide an Ethernet connection and power (either 24-48Vdc or PoE)
• Using a web browser, browse to the default IP address 192.168.1.2
• Enter the user name and password (Defaults: user & password)
• Set an IP address and net mask (or set DHCP) on the IP page
• From the SIP settings page, select the SIP1 Info sub-page, check that ENDPOINT is
ENABLED
• On the SIP1 Info sub-page, give the phone a LOCALID (usually its extension number)
• On the SIP1 Info sub-page, set DOMAIN, PROXY and REGISTRAR all to the
address of the SIP server. If registrar authentication is in use, also set a user name
and password.
• Program any dial memories using the Dialling & Memories pages
With these basic steps the telephone will be able to make and receive calls in most cases.
Check the Current Status page to help diagnose problems - this will show whether or not the
phone is registered and what is happening during calls (refresh the page to see changes).
NOTE:
Make sure each unit is given at least a basic configuration before installing it. All units have
identical settings as factory defaults, so each one must be individually configured to give it a
unique identity on the network. This may be difficult to do after the units are installed.
1,2
4.2 Frequently Asked Questions (FAQs)
Note: a more up-to date list of questions and answers may be available on the GAI-Tronics
website. See www.gai-tronics.co.uk/voipsupport.htm for more details.
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9GF8QLWVDUHPDUNHGDFFRUGLQJO\QH[WWRWKHSRZHUWHUPLQDOVVHHLQVWDOODWLRQJXLGH
RUIRU$XWHOGDF9R,3IRUGHWDLOV
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WKHGDWDSDLUV6HHLQVWDOODWLRQJXLGHIRUGHWDLOV
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4.2.1 What network facilities do I need to provide?
This may vary widely depending on how your network is constructed and what else it is
carrying, but as a general guide you will probably need:
• A SIP proxy server (to route calls)
• A SIP registrar server (frequently combined with the proxy server) to resolve URIs to
IP addresses
• A TFTP server (for downloading configuration files).
• A TCP Syslog server (for reporting alarms and external inputs)
• An SMTP server (for reporting via email)
• An STNP server (to synchronise the internal clock)
Dedicated systems, such as Gatekeepers, VoIP-enabled PABXs or soft PABXs may also
provide these functions. Bear in mind that GAI-Tronics telephones only support Session
Initiation Protocol (SIP) to RFC3261, as opposed to H.323 or SCCP VoIP protocols for
example.
Note that the performance of VoIP telephones depends on the provision of sufficient
bandwidth and prioritisation on the network to give the quality of service required.
4.2.2 How do I set up dialling and memory lists?
Let's assume you have a telephone with 2 buttons: memory 1 for information, memory 2 for
emergency. You want the emergency button to call "888" only. You want the information
button to call the information desk, or if that is busy the security office, or failing that the
administration centre on 223344.
First set up the 4 possible user agents (end points) as memories on the memories page (it
doesn't matter which end point is in which memory):
Note that comfort strings have been set to give the user confidence that "dialling" is taking
place when the button is pressed.
Then set up 2 memory lists, one for each button:
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Memory list 1 relates to memory button 1, and will dial memories 2, 3 and 4 in cascade.
Memory list 2 is for memory button 2, and will dial memory 1 only.
Note that, in this case, WAKEANDDIAL is set for both - the normal case for help point and
hands-free telephones.
Refer to the Dialling & Memories pages in section 5.10 for more details.
4.2.3 Can I set the phone to make calls without a proxy (ie peer-topeer)?
There are two ways of setting the phone to make peer to peer calls.
The first is where there is no proxy server on the system at all. In this case:
1. Set the ENDPOINT field on SIP 1 Info page to ENABLED, but make sure DOMAIN,
PROXY and REGISTRAR are blank. Set ENDPOINT on SIP info 2, 3 and 4 to
DISABLED.
2. Make each entry on the Memories page the address of an endpoint or phone, in the
form 1@192.168.1.2. Note that the number before the "@" symbol is not normally
significant
address of the end point.
3. Note that peer-to-peer calls can only be made by using a memory - not by manually
dialling from a numeric keypad. All phones have at least one memory list (the
OFFHOOK list). Refer to section 5.10 for details on setting up memories.
The second way is where one or more proxy servers are in use, but you want to be able to
make a peer-to-peer call if no proxy is available. This is referred to as "failover to peer-topeer". In this case:
1. Set the proxy address on one of the 4 SIP info pages (usually the one with the lowest
priority) to be the IP address of an endpoint, in the form 192.168.1.2
REGISTRAR address to be blank.
2. If all attempts to make calls to higher priority proxies fail, the phone will attempt to
place a call to this IP address as a peer-to-peer entity, regardless of what number is
dialled or what entry is selected from a memory list.
NOTE: you cannot make a peer-to-peer call by entering an IP address on a numeric
keypad - peer-to-peer calls can only be made using a memory dial.
3
- there just needs to be a number, followed by "@", followed by the IP
4
, but set the
4.2.4 How do I set up Real-time alarm reporting via email or syslog?
To do this you will need to set up email and/or syslog facilities within the phone, then set up
the alarm itself, using the following 3 web pages:
• Refer to the Email page to enter the required SMTP server settings for email.
• Refer to the IP settings page to set up Syslog server settings.
• Refer to the Alarms page to set which alarm events will report.
In the example shown below, a syslog message will be generated if the telephone has a cold
reset (ie recovers from a power failure) or has an integrity loop fault (ie the handset has been
detached). In addition, it will send an email to the security office if the handset is detached.
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1RWHWKLVFRXOGDOVREHD)4'1IXOO\TXDOLILHGGRPDLQQDPHLIWKHUHLVD'16DYDLODEOHRQ
WKHQHWZRUN
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4.2.5 How can I set up an external beacon to flash when the phone is ringing?
Traditional telephone beacons and sounders, with ring detectors, will not work on VoIP
because there is no ring signal. You will therefore need a powered beacon or sounder
instead, and use the telephone's volt-free contacts to activate it. These beacons or sounders
must be provided with a separate power supply - they cannot be powered from the telephone.
Having connected an external device to an output (say Output 1), the next step is to set the
output to activate it when required.
Enter the keyword "RING" for the relevant output. The example above shows the output set
with a cadence of 10:0, meaning continuously on. This would be suitable for a beacon,
because beacons usually flash (once per second) when permanently energised. It might not
suit a sounder, however, because it would emit a continuous tone, which might not be
recognisable as a phone ringing. For a sounder on its own, the keyword "RINGCADENCE" is
a better option, causing the sounder to be energised in time with the normal phone ringer.
For a beacon and sounder together, it is often best to use a separate output for each as
shown:
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In this example, Output 1 is set to activate a flashing beacon, whilst Output 2 is set to activate
a sounder in sync with the cadence of the ring signal (set on the Tone settings page). In both
cases the outputs are energised when the phone is ringing with an incoming call, and deenergised when the call is answered or disconnected.
Refer to the Logic Settings page (section 5.17) and Tone settings page (section 5.15) for
more details.
4.2.6 How do I set up a door-entry system?
A common application is to have a single button hands-free telephone mounted outside a
door, and a 15 button Commander model at a remote security point. Visitors arriving at the
door use the hands-free unit to call the security point. A security guard answering the call can
release the door lock by pressing the "R" button on the Commander unit.
To achieve this, connect one of the volt-free outputs on the hands-free telephone (say output
1) to the electronic door release mechanism. Using the Logic settings page, set this output to
PULSE:
Note that the TIMER is set to 3, meaning that the output will remain active for 3 seconds after
being activated.
To activate this output from the security office, set the RECALL setting on the Key mapping
page of the Commander unit to the IP address of the hands-free unit. So, for example, if the
IP address of the hands-free were 192.168.9.2, the setting would be:
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Refer to the Logic settings page (section 5.17) and the Key mapping page (section 5.11) for
more details.
4.2.7 How can I use the phone to make paging or PA announcements?
If you are using GAI-Tronics CMA, simply set the PAGEMODE field (on the UNIT page) to
"aa3". CMA has a page button that will place a call to the unit in page mode, i.e. an
announcement tone will be heard from the unit, following which the CMA operator will be able
to make a page through the unit's speaker (see section 5.5).
It may be possible to activate this feature from systems other than CMA - contact GAI-Tronics
for details.
Note page mode is usually implemented using handsfree models (VR and Help Point for
example) but it may also be possible with other models, depending on application. The
integral relays can also be set to activate during a page, and this feature could be used to
trigger an external public address amplifier. Contact GAI-Tronics for details.
4.2.8 What additional features are available with CMA?
GAI-Tronics CMA is a security call centre application for Windows XP™ designed for use with
GAI-Tronics analogue and VoIP telephones, providing powerful features such as:
• Automatic call answering
• Text-to-speech auto announcements
• Location ID linked to a user-definable mapping application
• Call recording and incident logging
• Call queueing
One of the system's most important functions is to give callers the reassurance that their call
is being dealt with and that their location is known.
The ANI field on the UNIT page is used as an identifying token to CMA. Using this the
telephone can automatically announce location information (using text-to-speech) to the user
and the call centre operator when a call is made. It is also used to locate the phone on a map
to help the operator identify its location and give assistance to the caller.
CMA can also activate 3 special auto-answer modes on hands-free VoIP telephones if
required by using codes also entered on the UNIT page:
Stealth mode, where the operator can listen discreetly to the telephone (the ANSMODE1 field
should be set to "aa1").
Intercom mode, where the operator can make a call to a telephone and start two-way voice
communication immediately, without the user having to answer (the ANSMODE2 field should
be set to "aa2").
Page mode, where the operator can make an announcement directly to the telephone, but not
listen (the PAGEMODE field should be set to "aa3").
5. Web pages in detail
The following sections describe the embedded web pages in detail.
Once past the login screen, all the pages have a similar layout.
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Edit button
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Module
name
Navigation
pane
The left hand navigation pane gives direct access to each of the 16 main pages, grouped by
functional headings of Network, Phone functions and Signals & Audio, plus the home page.
Most pages have an "Edit" button that allows the changing of parameters.
Some pages have entry dialog boxes that accept certain predefined values. These values
are listed in the sections below.
Some pages have links to related sub pages.
Each page displays its module name near the top for ease of navigation.
Note that these pages have been developed and tested on Microsoft Internet Explorer (v6).
Screen layout may appear differently using other browsers.
5.1 Login
To access the web pages, navigate to the unit's IP address using a web browser such as
Internet Explorer.
The factory default setting is for static IP addressing, with an address of:
Note that the unit's default subnet mask is 255.255.0.0.
The Phone will request a user name and password as shown.
Links to subpages
Page values
192.168.1.2
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The default user name and password are
user
password
(lower case)
The user name and password can be changed using the Access Settings page.
On accepting the username and password, the phone's home page is displayed.
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5.2 Home Page
No settings can be changed directly from the home page.
The Web support page link defaults to http://www.gai-tronics.co.uk/voipsupport.htm, but can
be changed on the Unit Settings page (section 5.5).
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At the bottom of the home page (you may need to scroll down, depending on screen
resolution) there is a list of information about the phone including serial numbers of the unit
and its PCBs, software versions and MAC ID.
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5.3 IP settings
The IP settings page is used to display or change various settings for connection to the IP
network.
DHCP: Enables or disables the use of DHCP for the assignment of IP parameters. If this
value is set to OFF the telephone will use the Static IP values. (Values available: ON or OFF,
default value is OFF)
ADDRESS: Sets the static IP Address of the unit. (Default value is 192.168.1.2) Do not enter
a value here if DHCP is set to ON.
MASK: Sets the static sub-net mask. (Default value is 255.255.0.0) Do not enter a value here
if DHCP is set to ON.
GATEWAY: Sets the static default gateway address (Default value is 0.0.0.0)
DNS1: Sets the IP address of the primary static DNS server. If DHCP is enabled then this
DNS server will not be used. (Default value is 0.0.0.0 )
DNS2: Sets the IP address of the secondary static DNS server for redundancy. If DHCP is
enabled then this DNS server will not be used. (Default value is 0.0.0.0 )
LOCALDOMAIN: Sets the domain name of the telephone on the network, as used by DNS.
May be assigned by DHCP.
WEB: Enables or disables access to the web server (Values available: ON or OFF, default
value is ON)
WEBPORT: Sets the TCP port through which the Telephone Web server can be accessed
(Default Value is 80)
TELNET: Enables or disables access to the telnet server (Values available: ON or OFF,
default value is ON)
TELNETPORT: Sets the TCP port through which the Telephones telnet server can be
accessed (Default Value is 23)
SYSLOG: Sets the destination address for syslog server messages. (Valid values: IP
address or FQDN. Default value: blank)
SYSLOGPORT: Sets the port number to be used for syslog messages. The default value is
514
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SYSLOG2: Sets the destination address for a second syslog server for redundancy. (Valid
values: IP address or FQDN. Default value: blank)
SYSLOGPORT2: Sets the port number to be used for syslog messages (second syslog
server). The default value is 514
SYSLOGFACILITY: Sets the SYSLOG message facility level, as per RFC3164. (Default
value: 14)
SYSLOGSEVERITY: Sets the SYSLOG message severity level, as per RFC3164. (Default
value: 5)
STUN: Sets the IP address or URL for the STUN server that will be used to resolve STUN
requests. Leaving this field blank will disable the STUN facility. (Default value: blank)
At the bottom of the IP settings page are 2 action buttons, each with an entry box. The entry
boxes will accept either an IP address or FQDN. These buttons provide useful diagnostic
functions:
PING: Sends an ICMP ping to the entered address, providing a results page.
TRACEROUTE: Executes a series of PING messages with varying HOP numbers in order to
determine the routing used to reach the destination address. A results page is displayed.
5.3.1 Note about Syslog:
GAI-Tronics VoIP products send Syslog messages using TCP (as opposed to UDP). Please
make sure that Syslog servers support TCP.
SYSLOG over TCP ensures reliable delivery, and utilises port number 514 by default. Note
that in the event of a TCP session failure there is no higher layer protocol acknowledging the
receipt of the message, but each message has an Event Count parameter that will indicate if
a previous message has been lost
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5.4 SIP settings
The SIP settings page is used to view or change parameters specific to the SIP signalling
protocol. GAI-Tronics VoIP phones can hold details of up to 4 SIP proxies. If the phone is
unable to register or make a call it can fail over to the next in a prioritised sequence. There is
a SIP Info page for each of the 4 possible endpoints, and a General SIP Info page containing
details common to all. The 4 endpoint pages are sub pages of the General page shown
below:
LOCALPORT: Configures the port number used for the local SIP signalling socket.
Default value: 5060
PROXYFAILOVERSTATUSES: This field contains a list of SIP error codes that will trigger a
fail over from one proxy to the next. Codes are 3 digits and the wildcard character “x” can be
used (ie 5xx would include any code from 500 to 599 inclusive). Codes are separated by
commas. Maximum field length 79 characters, ie 20 codes. The default list is 5xx, 6xx, 49x,
403, 406, 9xx. Codes are as defined in RFC3261 except 9xx, which is defined as "time-out"
and should always be included in the list.
Note that there are two failover mechanisms: one for proxies (defined here) and a second for
memories (defined in section 5.10.3). If a call fails due to a proxy error, the phone will then try
to place the call to the same number on the next proxy. If the call fails due to an endpoint
problem (for example "busy"), the phone will try the next number in the list, on the current
proxy.
DONTSTARTMEDIAATRING: This setting is not normally required. It can be used to delay
the sending of media packets to end points until the call has been answered. Only required if
problems are encountered with certain types of end point. Default value: OFF.
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SENDDTMFLAST: This setting is not normally required. It can be used to reorder the codec
sequence to end points, so that the DTMF codec is sent last. Only required if problems are
encountered with certain types of end point. Default value: OFF
RTPTOS: Sets the value of the TOS/Diffserv field in the UDP packets carrying RTP data. This
value prioritises traffic over the network to provide QoS (Quality of Service) for voice, see
RFC2474. Valid values are 1->63 (Default value = 46
SINGLEPTIME: Certain endpoints can only accept a single audio packet time regardless of
CODEC (see AUDIO page). This field forces a single packet time to the value set in ms.
Valid values are 0 to 100, where 0 disables the feature allowing codecs to use the packet
times set on the AUDIO page. Default value 0.
SENDMULTIPARTMIME: This option is for future enhancement and should always be set to
‘OFF’. Default value OFF.
NEWBRANCHONAUTHBYE: This is a legacy option that is no longer used, and must always
be set to 'ON'. Default value 'ON'.
MODE: This field sets whether multiple proxies and registrars are used serially or
concurrently. If set to SERIAL the phone will attempt to register with the next priority registrar
if registration with the current one fails. If set to MULTIPLE it will attempt to maintain
registration with all enabled registrars, and will use the priority sequence for outbound call
failover. Default value: SERIAL. When only a single proxy / registrar is enabled, set this
value to SERIAL to ensure any registration failure is detected quickly.
REGTIMEOUT: Sets the Registration timeout value (in seconds) that will be suggested by the
telephone to a Registrar. Following the expiry of this timeout, the telephone will be
deregistered and then automatically attempt to re-register. (Value range: 0 to 2
value: 3600) The registration server can ignore or override this suggested time.
REREGTIMEOUT: Sets a period in seconds after which the phone will force a re-registration
period and the server cannot override it. Disabled if set to zero. Default value 0. This field
can be used to ensure that registration is maintained for this particular phone, regardless of
the general settings on the registration server. For example, if this were an emergency
phone, setting this field to 30 would force re-registration every 30 seconds even if the server
normally only refreshes registration once an hour. In this way, if the proxy server fails or
becomes unavailable, the phone can detect it quickly and either attempt to register with the
)
32
-1, default
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next server in the priority list (if MODE is set to SERIAL) or direct calls to the next priority
server (if MODE is set to MULTIPLE).
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Note that, if the current registrar becomes unavailable, the telephone may not be able to
make a call until it re-registers with the next.
5.4.1 SIP Info sub-pages:
Each of the 4 sub pages is identical, and is used to set parameters for each of 4 possible
proxies.
LOCALID & DOMAIN: together these set the URI (uniform resource identifier) of the phone.
In the example shown above the URI would be sip:12345@mydomain.com.
These values are used in the To:, From: and Contact: headers, and also in the registration
process with a registrar.
They will accept any alphanumeric string and their default values are both blank.
PROXY: Sets the IP address or the FQDN of the SIP proxy server to be used for
incoming/outgoing calls. Default value: blank
PROXYPORT: Sets the port number on the proxy used for SIP protocol signalling.
Default value: 5060
PRIORITY: Sets the failover sequence between the 4 pages.
REGISTRAR: Sets the address of the Registrar, either as an IP address or FQDN. The
registrar address and the proxy may or may not be the same, but the address for registration
must be set here. Default value: blank
REGISTRARPORT: Sets the port number to send the requests to. Is 5060 by default or if
unspecified.
USERNAME: Sets the username for the registrar authorisation realm. (Default value: blank)
PASSWORD:
ENDPOINT: Sets whether the subpage is ENABLED or DISABLED. (Default value:
ENABLED for SIP1, all others DISABLED).
Sets the password for the registrar authorisation realm. (Default value: blank)
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Note that the Proxy address could also be that of a peer-to-peer entity, allowing the unit to
make a direct peer-to-peer to connection. This can provide an extra level of resilience,
allowing the unit to fall back to a peer to peer call in the event that all proxy servers become
unavailable
5.5 Unit settings
The Unit page is used to set parameters for how the unit interfaces to the network, including
configuration file updates.
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HOSTNAME: Sets the unit host name. Maximum 15 alphanumeric characters (a-z, A-Z , 0-9).
Default Value is a unique string starting with "GT" and followed by the serial number of the
main circuit board inside the phone (referred to as the "Board serial" on the home page). The
host name identifies the unit on the network, and is also used in email and syslog messages
to identify the source of the message. If using DHCP, this field must be kept unique for each
phone on the system.
UPDATE SERVER: Sets the address of the host running the TFTP server. (Valid values: IP
address or FQDN. Default value: blank)
UPDATE FILE: The name of the update control file on the update server. This name may
contain the macro symbols %m, %h and %i. These symbols are expanded to the MAC
address, host name and IP address respectively. (Default value: blank)
UPDATE INTERVAL: Forces the unit to attempt a file download every X hours where X can
be an integer value between 0 and 1000. A value of 0 disables the periodic update request.
The default value is 1. Any non-zero value will cause the unit to attempt a configuration file
download at boot time.
HELPSERVER: Sets the default address for the Help web page reached from the link on the
home page. The default value is http://www.gai-tronics.co.uk/voipsupport.htm, but it can be
changed to any appropriate page available on the network.
LAN SPEED: Sets the speed or auto negotiation status for the WAN Ethernet port. Valid
values: 10, 100 or AUTO. Default value: AUTO. If the speed is auto negotiated the duplex
setting has no effect.
LAN DUPLEX: Sets the duplex value for the WAN Ethernet port. Valid values: FULL or HALF.
Default value: FULL.
CONFIGID: Used by the configuration upgrade script to determine if the local configuration is
the same as the one it wants to upgrade to. If this matches the CONFIGVERSION line in the
update control file, no download will take place. Default value: blank.
ANI: Used as an identifying token to GAI-Tronics CMA Call Management Application. Default
value: "GAIPHONE". Maximum 12 characters.
DEFAULT_ANS_MODE: Sets the default answer mode. This mode will be used to answer a
call when ANSMODE1, ANSMODE2 and PAGEMODE are not triggered. Values available are
RING, PICK-UP, PAGE and STEALTH. RING is normal phone operation, where a button
must be pressed or handset lifted to answer an incoming call. PICK-UP is as described in
ANSMODE2 below. PAGE is as described in PAGEMODE below. STEALTH is as described
in ANSMODE1 below. Default value: ‘RING’.
The next 3 fields set “passwords” that can be used by GAI-Tronics CMA to activate 3 special
auto-answer modes, usually for hands-free telephone types.
ANSMODE1: Stealth auto-answer mode, where the telephone provides no indication of the
incoming call and immediately auto answers the call. The speaker is muted, and the
microphone gain is enhanced. Sending a DTMF ‘*’ during a call will change the unit to
ANSMODE 2. For activation from CMA, set this field to "aa1"
ANSMODE2: Sets Intercom auto-answer mode, where the telephone auto answers and
provides normal duplex audio, preceded by an announcement tone. For activation from CMA,
set this field to "aa2"
PAGEMODE: Where the unit auto answers and disables the microphone. A "splash" tone
(tone 9) is emitted from the speaker to alert those nearby of an impending page
announcement. The output level of the speaker is increased to its maximum level. For
activation from CMA, set this field to "aa3"
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At the bottom of the UNIT page are three action buttons:
Update Now: Causes the phone to fetch the update file immediately.
Reboot Now: Causes the unit to reboot.
Reset to defaults: Causes the unit to reset to a predefined default configuration, effectively
returning it to the settings it had when it left the factory. WARNING: this will undo any
configuration changes, including returning the IP address and net mask to their default, static
values of 192.168.1.2 and 255.255.255.0. It will also reset the username and password to
defaults.
5.5.1 Audio Path Test
Audio Path Test (APT) is a factory option which must be specified at order time. If the phone
was not ordered with APT, these controls will not appear on the UNIT page.
The purpose of APT is to send a specific audio tone from the earpiece or speaker of a
telephone and then check that it is correctly received by the microphone. This will then verify
that both microphone and speaker are functioning.
APT appears as an alarm on the ALARMS page, and can be set to report via Syslog and / or
email like any other alarm, with some differences as listed below.
The test can be set to run automatically or triggered manually using the controls below:
APTENABLE sets whether APT is on or off
APTTIME sets a start time (24h clock) and test interval (in hours). The field should contain
first the time in hours and minutes separated by a colon (:), followed by a comma,(,) followed
by the interval in hours (range 1-24). Automatic testing will start at the specified time and
repeat every specified interval until 00:00 midnight the next day. The cycle will then repeat
the next day and so on. Default is 00:00,24 meaning that the test will perform once per day at
midnight.
APTCOUNT sets the number of tests that will be performed at each interval. Range is 1-10.
Default is 1, but it can be increased to repeat the test at each interval.
APTOKCOUNT sets the number of tests that must pass at each interval to be classed as a
successful test. Default value is 1. APTOKCOUNT must always be <= APTCOUNT. For
example if APTCOUNT were set to 3 and APTOKCOUNT to 2, the test would be deemed to
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have passed if 2 pass readings out of 3 were recorded. This feature is to allow for potential
disruption in areas of high ambient noise.
APTREPORT sets whether or not APT will send reports every time the test passes. Normal
alarms only report if they change state; setting APTREPORT to ON will cause the phone to
send a regular report confirming that it's acoustic components are healthy. By inference this
report also confirms that the phone is powered, running and connected to the network so it
also provides a useful general health check. If the test fails, the phone will not send repeated
reports until at least APTOKCOUNT tests pass again.
APT now will start an APT test within 60 seconds. This button will only start a test if
APTENABLE is set to ON.
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5.6 Access settings
The Access settings page allows the user name and password to be changed.
USERNAME: Can be up to 30 characters long, and can contain only the alphanumeric
characters a-z, A-Z , 0-9 . The default value is “user”. The Username cannot be blank.
IMPORTANT: The word ‘root’ is a reserved username and must not be used or assigned a
password. Setting a user name of "root" will make it impossible to access the phone, and will
require a reset to factory defaults.
PASSWORD: Can be up to 30 characters long, and can contain only the alphanumeric
characters a-z, A-Z , 0-9 . The default value is "password". Password can be blank if
required.
Note: please make sure to record the user name and password securely. They will be
required to access the phone every time, whether by web page, command line or
configuration file. In the event that the username and password are lost, the unit will
need to be reset to factory defaults. This can be done by holding down a button on the
main circuit board. See section 10.
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At the bottom of the Access page are a series of counters showing how many unsuccessful
access attempts have been made to this phone, and how many times it has been rebooted.
The counters can be reset using the "Reset counters" button.
5.7 Serial settings
The Serial settings page is used to set the speed for communication on the serial port.
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Speeds available (from a drop-down list) are: 9600, 19200, 38400, 56700 & 115200 baud.
The default value is 115200.
The other parameters for serial comms are: 8 data bits, 1 stop bit, no parity.
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5.8 Email settings
The telephone can report various alarm and input conditions via email (see the ALARMS and
LOGIC pages in sections 5.14 and 5.17). The Email settings page is used to set the
parameters required.
SERVER1: Sets the primary SMTP server, as an IP address or a FQDN
SERVER2: Sets the secondary SMTP server, as an IP address or a FQDN, for redundancy.
TOADDRESS, CCADDRESS & FROMADDRESS: Set the email addresses that will appear
in the message. Note that the phone can send the message to two separate addresses (TO
& CC) Each of these fields can contain a single email address of the form abc@xyz.com
SUBJECT: Sets the subject that will appear with each email message from this unit.
SMTP: enables or disables email.
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