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VoIPMaster™
Version 4.x
VoIP to GSM gateway
Connecting Cellular Phones directly to
‘Voice over IP’
worldwide networks
User Manual
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MasterVoIP VoIP to GSM Gateway
Usage Warnings
1) High voltage transients, surges, and other power irregularities can cause
extensive damage.
It is the user's responsibility to provide a power protection system.
2) It is the user's responsibility to install, operate, and maintain the system in
accordance with all
applicable codes, regulations, and safety measures.
Trademark and Patents
All trademarks, patents and copyrights apply.
General Manual Notes
Without notice and without obligation, the contents of this manual may be revised to
incorporate
changes and improvements.
Every effort, has been made to ensure that the information in this manual is most
complete and
accurate while writing this time of publication.
Nevertheless, Mega 2000 AS cannot be held responsible for errors or commissions.
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Februaray-2006 Page 3 / 42 MasterVoIP VoIP to GSM Gateway
Dear Customer,
We thank you for purchasing our VoIP Master VoIP to GSM Gateway.
The information in this manual does not constitute a warranty of performance,
although the
information has been compiled and checked for accuracy by Eurotech
Communication Ltd.
All our products are developed and produced by experienced engineers, who
aspire to achieve customer
satisfaction, utility value and reliability of products.
Warranty Policy
The Dual Cell to BRI Gateway product you have purchased is under warranty of
12 months from the
date of purchase, by the original purchaser. In case of defects of materials or
workmanship, Eurotech
Communication will replace it free of charge. This warranty applies to
hardware/software but does not
include SIM Cards.
This warranty will not be honoured if the device has been mishandled in any
way.
We hope you enjoy our product and we will be happy to receive any comments
you may have. This will
enable us to improve our products and the Technical Support that we give to
every customer.
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TABLE OF CONTENTS
1 Getting Started................................................................................................................................................. 5
2 Check your package Items............................................................................................................................... 7
3 VoIP Client ATA ............................................................................................................................................... 8
3.1 Product Overview............................................................................................................................................... 8
3.1.1 Key Features............................................................................................................................................ 8
3.1.2 Hardware Specification............................................................................................................................. 9
3.1.3 Basic Operations .................................................................................................................................... 10
3.1.3.1 Getting Familiar with the Key Pad and the Voice Prompt........................................................................ 10
3.1.4 Placing Phone Calls................................................................................................................................ 11
3.1.4.1 Calling phone or extension numbers..................................................................................................... 11
3.1.4.2 Direct IP calls...................................................................................................................................... 11
3.1.4.3 Blind Transfer...................................................................................................................................... 12
3.1.4.4
3.1.5
3.1.6
3.1.7
3.2 Configuration Guide......................................................................................................................................... 15
3.2.1 Configuring VOIP Client with a Web Browser........................................................................................... 15
3.2.1.1 Access the Web Configuration Menu .................................................................................................... 15
3.2.1.2 End User Configuration........................................................................................................................ 15
3.2.1.3 Advanced User Configuration............................................................................................................... 19
3.2.1.4 Saving the Configuration Changes........................................................................................................ 29
3.2.1.5 Remotely rebooting VoIP Client ATA.................................................................................................... 29
3.3 Restoring the Factory Default Settings .............................................................................................................. 30
3.4 VoIP Master..................................................................................................................................................... 31
3.4.1 What is the VoIP Master and how it works................................................................................................ 31
3.4.2 Set-up and Installation ............................................................................................................................ 33
3.4.3 Installing the Manager Application ........................................................................................................... 35
3.4.4 Define the Com port Connection.............................................................................................................. 36
3.4.5 Port and SIM Settings ............................................................................................................................ 39
3.4.5.1 Dial Settings for the GSM Port.............................................................................................................. 39
3.4.5.2 SIM Settings........................................................................................................................................ 40
3.4.5.3 Follow Me Settings .............................................................................................................................. 41
3.4.5.4 Call Back Settings ............................................................................................................................... 42
Attended Transfer................................................................................................................................ 12
Call Features.......................................................................................................................................... 13
Fax Support ........................................................................................................................................... 13
LED Light Pattern Indication.................................................................................................................... 14
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1 Getting Started
Eurotech Communication
for greatly saving your call costs. We will do our best to make your installation efforts as well as day-to-day
configuration and monitoring tasks be ple asant tasks as possible. We wish you a smooth operation while greatly saving
your office mobile phone calls.
This chapter is your “
on each stage as well as references for more elaborated explanations, drawings and examples in other chapters. The
following is a list of tasks you shall perform, where you shall go o ver it sequentially or skip tasks that are optional and
not required for your current needs. It is advised that you will use the following tasks as
your
Do To List
As a start
Later proceed with the
resides in the VoIPMaster gateway and enables a Web based configuration interface.
The VoIP Client ATA provides VoIP call origination and termination with PSTN network, with some add-on
supplementary services which are reviewed at Chapter 3.
The
The following topics of the VoIP Client ATA are reviewed in Chapter 3:
Client ATA Product Overview, K ey Features, Hardwa re Specification
Check your package Item s
VoIPMaster ga teway
Getting Familiar with the Key Pad and the Voice Prompt
Placing Phone Calls
Calling phone or extension numbers
Direct IP calls
Blind Transfer
Attended Transfer
Call Features
Fax Support
LED Light Pattern Indication
Map for installation, configura ti on and m onitori ng tasks”
.
team is glad you have chosen to use the
at Chapter 2 “
Gradstream HanyTone 286 VoIP Client ATA (Analog to Telephone Adaptor)
adds new capabilities of GSM to VoIP calls origination and termination to the client ATA.
Check your packa ge Item s”.
Eurotech’s VoIP Master
and includes a short explanat ion
and
Basic Operations
GSM to VoIP gateway
as follows
which
:
Now you shall start configure the
The following configuration actions shall be performed with several guidelines for optional actions:
Configuring VOIP Client with a Web Browser
Access the Web Configuration Menu
End User Configuration
Advanced User Configuration
Saving the Configuration Changes
Remotely rebooting VoIP Client ATA
Restoring the Factory Default Settings
After you have completed the
the VoIPMaster concept rule in the network and the way it works at “
chapter.
VoIP Client ATA
VoIP Client ATA
following the
you can start configuring the
Config uration Guide
What is the VoIP Master and how it works
at chapter 3.2.
VoIP Master,
starting with learning
”
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MasterVoIP VoIP to GSM Gateway
Completing this you shall start the installation procedure of the VoIPMaster following the
Set-up and Installation
as follows:
Installing the Manager Application
Define the Com port Connection – to enable a PC to VoIP master proper connection
Port and SIM Settings – to associate and set SIM and Ports accordingly
Dial Settings for the GSM Port – to define policies and profile of behaviour when dialling
SIM Settings – regarding with usage limits and other optional modes
Call follow-me settings – to let the system call you while you are away from office as if you
where in office
Call Back Settings – to let waiting lines make the call when line is available again.
At menu: Cellular Gateways
Please Give Us feedback to improve your BRI Gateway product
Please let us know your feedback and enhancement ideas to improve the product to your
best value.
Email: Support@mega2000.no
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2 Check your package Items
Please verify your package contains the following components (some were ordered specific) before installation:
Main Hardware Device – The
110/220V Electric Power converter to 24V with cables suppli ed
VoIP master softwa re Installation CD
Manual file and additional auxiliary ut ilities.
GSM Antenna
RS-232 Serial PC COMport connection cable
– To be installed to the VoIP Master Gateway
VoIP Master Gateway
- Installation kit for MS-Windows Management Application, this User
we’ll be referred as
Compor t ca ble
in this manual
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3 VoIP Client ATA
Before using this device please perform the following actions:
1. Connect the VoIPMaster (Which include the VoIP Client ATA as a built-in module) to the IP network via the RJ-45
connection near the 2 LEDs and power supply side. You must have an account with a VoIP termination service provider
or you should register an extension with a SIP Gateway/Server. Get all needed data from your provider (such as: user
name, Password, server IP addresses ports etc.).
2. Connect a regular Analog telephone (RJ-11 connection) to the system and configure it first as a regular VoIP client.
That configuration is done using a web interface. You will find instructions on page 14 of this manual.
3. Test that you can make and receive calls using your regular phone set.
4. Run the GSM management software, and configure it according to the manual and the interface menu.
3.1 Product Overview
The report will include various standards been used in each demo and any interoperability issue need to
be considered regarding the need for certain standard support, what section of the standard are mandatory
and what standards implementation are recommended as an implementation reference.
3.1.1 Key Features
The document will be prepared as contribution of all partners where Albatronics will integrate the contributions.
Each partner will contribute information for its demo provided equipment regarding with standards support details.
Supports SIP 2.0(RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server),
NTP, PPPoE, STUN, TFTP, etc.
Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive jitter control and packet
loss concealment technology
Supports various codecs including G.711 (PCM a-law and u-law), G.723.1 (5.3K/6.3K), G.726, (40K/32K/24K/16K), as
well as G.728, G.729 and iLBC.
Supports Caller ID/name display or block, Call waiting caller ID, Hold, Call Waiting/Flash, Call
Transfer, Call Forward, in-band and out-of-band DTMF, Dial Plans, etc.
Supports Caller ID/name display or block, Call waiting caller ID, Hold, Call Waiting/Flash, Call
Transfer, Call Forward, in-band and out-of-band DTMF, Dial Plans, etc.
Supports fax pass through (for PCMU and PCMA) and T.38 FoIP (Fax over IP).
Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation
(G.168), and AGC (Automatic Gain Control)
Supports standard encryption and authentication (DIGEST using MD5 and MD5-sess)
Supports for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Supports automated NAT traversal without manual manipulation of firewall/NAT
Supports device configuration via built-in IVR, Web browser or Central configuration files through TFTP or HTTP server
Supports firmware upgrade via TFTP or HTTP with encrypted configuration files.
Supports PSTN pass through, able to make and receive VoIP or PSTN calls using same connected analogue phone.
Ultra compact (wallet size) and lightweight design, great companion for travelers.
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Compact, lightweight Universal Power adapter
3.1.2 Hardware Specification
The following table describes the hardware specification of VoIP Client ATA
Model VoIP Client (ATA)
LAN interface
1xRJ45 10Base-T
Button 1
LED GREEN & RED color
Universal
Power Adaptor
Input: 100-240VAC
Output: +5VDC, 1200mA
UL certified
Dimension
65mm (W
93mm (D)
27mm (H)
Weight
Operating
Temperature
Humidity
32 - 104oF
0 - 40oC
10% - 95%
(non-condensing)
Compliance FCC/CE/C-Tick
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3.1.3 Basic Operations
3.1.3.1 Getting Familiar with the Key Pad and the Voice Prompt
VoIP Client ATA has a stored voice prompt menu for quick browsing and simple configuration.
To enter this voice prompt menu, simple pick up the phone and press the button on the VoIP Client ATA;
or pick up the phone and dial “***”. The following table shows how to use the voice prompt menu to configure the device for
required voice prompts.
Menu Voice Prompt User’s Options
Main Menu “Enter a Menu Option” Enter ‘*’ to next option and “#” back to main menu, or
Dial 01 – 06, 47, 86 or 99 Menu option
01 “Static IP Mode”, or
“Dynamic IP Mode”
Dial ‘9’ to toggle the selection.
If user selects “Static IP Mode”, user will need to configure
the all IP address information through menu 02 to 05. If user
selects “Dynamic IP Mode”, the device will retrieve all IP
address information from DHCP server automatically when
user reboots the device.
02 “IP Address” + IP address The current WAN IP address is announced. Enter 12-digit
new IP address if in Static IP Mode.
03 “Subnet” + IP address Same as Menu option 02
04 “Gateway “ + IP address Same as Menu option 02
05 “DNS Server” + IP address Same as Menu option 02
06 “TFTP Server “ + IP address Same as Menu option 02
TFTP server is used to update the firmware of the device.
47 “Direct IP Calling” When entered, user will be prompted by dial tone, dial the
12-digit IP address to make a direct IP call. (For details, see
“4.2.2 Make a Direct IP Call”.)
86 “No Voice Messages”; or
“Voice Messages Pending”
If there are voice messages, user can dial ‘9’ and dial preconfigured phone number to retrieve voice message.
99 “RESET” Dial ‘9’ to confirm the RESET; or
Enter MAC address to restore factory
default setting (For detail, see section 8)
“Invalid Entry” Automatically return to Main Menu
Notes:
Once the LED button is pressed, it enters the voice prompt main menu. If the button is pressed
again while it is already in the voice prompt menu state, it will jump to the “Direct IP Calling” option - dial tone
plays in this state.
• “*” shifts down to the next menu option
• “#” returns to the main menu
• “9” functions as the ENTER key in many cases to confirm an option
• All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address.
Once all digits are accumulated, it automatically processes them.
• Key entry cannot be deleted but the phone may prompt error once it is detected
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3.1.4 Placing Phone Calls
3.1.4.1 Calling phone or extension numbers
There are currently two methods to make an extension number call:
1. Dial the extension number directly and wait for 4 seconds. (Default “No Key Entry Timeout”).
Or:
2. Dial the number directly, and press # (assuming that “Use # as dial key” is selected in the web configuration).
Other functions available during the call are call-waiting/flash, call-transfer, and call-forwarding supplementary call
services.
3.1.4.2 Direct IP call s
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy.
VoIP calls can be made between two phones, if:
Both VOIP Client ATA and the other VoIP device (i.e., another VOIP Client ATA or other SIP products) have public IP
addresses, or
Both VOIP Client ATA and the other VoIP device (i.e., another VOIP Client ATA or other SIP produces) are on the same
LAN using private or public IP addresses, or
Both VOIP Client ATA and the other VoIP device (i.e., another VOIP Client ATA or other SIP products)
can be connected through a router using public or private IP addresses.
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog
phone, then access the voice menu prompt by dial “***” or press the button on the HT286, and dial
“47” to access the direct IP call menu. User will hear a voice prompt “Direct IP Calling” and a dial
tone. Enter a 12-digit target IP address to make a call.
The follow is a table of the encoding scheme for the most commonly used characters:
INPUT Encoding
00 0
01 1
02 2
03 3
04 4
05 5
06 6
07 7
08 8
09 9
*0 . (dot character)
*4 : (column character)
Examples:
If the target IP address is 192.168.0.160, the dialing convention is
Voice Prompt with option 47, then 192168000160
followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this case, the
default destination port 5060 is used if no port is specified.
If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:
Voice Prompt with option 47, then 192168001020*45062
followed by pressing the “#” key if it is
configured as a send key or wait for 4 seconds.
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3.1.4.3 Blind Transfer
Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.
2. Then “A” dials *87 then dials C’s number, and then # (or waits for 4 seconds)
3. “A” can hang up.
Note: Call Feature has to be set to YES.
“A” can hold on to the phone and wait for one of the three following behaviors:
A quick confirmation tone (temporarily using the call waiting indication tone) followed by a
dial tone. This indicates the transfer is successful (transferee has received a 200 OK from
transfer target). At this point, “A” can either hang up or make another call
A quick busy tone followed by a restored call (on supported platforms only). This means the
.
transferee has received a 4xx response for the INVITE and we will try to recover the call. The
busy tone is just to indicate to the transferor that the transfer has failed.
Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the
transferee and decided to time out. Note: this does not indicate the transfer has been successful, nor does it indicate the
transfer has failed. When transferee is a client that does not support the second NOTIFY (such as our own earlier
firmware), this will be the case. In bad network scenarios, this could also happen, although the transfer may have been
completed successfully.
3.1.4.4
Attended Transfer
Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:
1.
“A” presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone
2. “A” then dial C’s number then # (or wait for 4 seconds). “A” and “C” now are in conversation.
3. “A” can hang up.
Note:
When intended Transfer failed, if “A” hangs up, the HandTone-496 will ring user “A” again to
remind “A” that “B” is still on the call, by pressing FLASH or Hook again will restore the
conversation between “A” and “B”.
.
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3.1.5
Call Features
The Following table shows the call features of VoIP Client ATA.
Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call)
*82 Send Caller ID (per call)
*50 Disable Call Waiting (for all subsequent calls)
*51 Enable Call Waiting (for all subsequent calls)
*70 Disable Call Waiting. (Per Call)
*71 Enable Call Waiting (Per Call)
Unconditional Call Forward.
*72
To use this feature, dial “*72” and get the dial tone. Then dial the forward number and “#” for a dial tone, then
hang up.
*73
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get the dial tone, then hang up.
Busy Call Forward
*90
To use this feature, dial “*90” and get the dial tone. Then dial the forward number and “#” for a dial tone, then
hang up.
*91
*92
Cancel Busy Call Forward
To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then hang up
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Then dial the forward number and “#” for a dial tone, then
hang up.
*93
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then hang up
Flash/Hook When in conversation, this action will switch to the new incoming call if there is a call waiting indication. When in
conversation without an incoming call, this action will switch to a new channel for a new call.
3.1.6
Fax Support
VoIP Client ATA supports FAX in two modes: T.38 (Fax over IP) (and fax pass through. T.38 is the
preferred method because it is more reliable and works well in most network conditions. If the service
provider supports T.38, please use this method by selecting Fax mode to be T.38. If the service
provider does not support T.38, pass-through mode may be used. To send or receive faxes in fax pass
through mode, users will need to select all the Preferred Codecs to be PCMU/PCMA.
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3.1.7
Following are the LED light pattern indications
LED Light Pattern Indication
.
RED LED
DHCP Failed or WAN No Cable
VOIP Client-486 fails to register
GREEN LED
Message Waiting Indication Button flashes every 2 seconds
RINGING Button flashes at 1/10 second
RINGING INTERVAL Button flashes every second
indicates abnormal status
flash every 2 seconds (if DHCP is configured)
flash every 2 seconds (if SIP is configured)
indicates normal working status
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3.2 Configuration Guide
3.2.1 Configuring VOIP Client with a Web Browser
VoIP Client ATA has an embedded Web server that will respond to HTTP GET/POST requests.
VoIP Client ATA is enabled with embedded HTML pages, which allow a user to configure the IP phone,
through a Web browser, such as Microsoft’s IE and AOL’s Netscape.
3.2.1.1 Access the Web Configuration Menu
First, get the IP address of the VOIP Client through section 2.1 with menu option 02. Then access the
VOIP Client’s Web Configuration Menu using the following URI:
Address
3.2.1.2 End User Configuration
Once this request is entered and sent from a Web browser, the IP phone will respond with the following login screen
is the IP address of the phone.
http://Phone-IP-Address
where the
Phone-IP-
:
The password is case sensitive with a maximum length of 25 characters. The factory default password
for End User is “admin”. After the correct password is entered in the login screen, the embedded Web server inside the IP
phone will respond with the following Basic Settings configuration page, which is explained in details below.
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The following table describes the various configurations to be performed:
End User
Password
This contains the password to access the Web Configuration Menu. This field is case sensitive
with max. 25 characters
IP Address There are 2 modes under which the IP phone can operate:
- If DHCP mode is enabled, then all the field values for the Static IP mode are
not used (even though they are still saved in the Flash memory) and the IP
phone will acquire its IP address from the first DHCP server it discovers on the
LAN it attaches to.
To use PPPoE feature please set the PPPoE account settings if the HT-286 is connected directly to
a DSL modem. The HT-286 will attempt to establish a PPPoE session if any of the PPPoE fields is
set. In this mode, the WAN side web access is disabled and TFTP upgrade for firmware is not
feasible and HTTP upgrade is the only available solution.
- If Static IP mode is selected, then the IP address, Subnet Mask, Default
Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields will need to be
configured. These fields are reset to zero by default.
Time Zone This parameter controls how date/time will be displayed according to the specified time zone.
Daylight Savings Time This parameter controls whether the displayed time will be daylight savings time or not. If set to
“Yes”, then the displayed time will be 1 hour ahead of normal time.
In addition to the Basic Settings configuration page, the end user also has access to the device Status page.
The following is a screen shot of the device Status page.
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Here are the status details shown
:
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MAC Address
The device ID, in HEX format. This is very important ID for ISP troubleshooting.
WAN IP Address
Product Model
Software Version
System Uptime
Registered
PPPoE Link Up
NAT
This field shows WAN port IP address.
This field contains the product model info.
Program:
This is the main software release. This number is always used for
firmware upgrade.
Bootloader
HTML:
VOC:
: This is normally not changed.
This is the user interface, normally not changed.
This is the codec program, normally not changed.
This shows system up time since last reboot.
This shows whether the unit is registered to service provider’s server.
This shows whether the
PPPoE
is up if connected
to DSL modem
This shows what kind NAT the VoIP Client ATA is connected to via its WAN port. It is
based on
STUN
protocol.
NAT Mapped IP
NAT Mapped Port
Statistical Status
WAN side public IP if connected to LAN of a SOHO router.
External port detected by
STUN
.
Self explainable. Please refer to the page displayed.
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3.2.1.3 Advanced User Configuration
To login to the Advanced User Configuration page, follow the instruction in section 3.2.1, they will lead
You to the following page: (The password is case sensitive with a maximum length of 25 characters and the factory default
password for Advanced User is “admin”).
Advanced User configuration page includes not only the end user configuration, but also some
advanced settings such as SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous settings. Following is the screen shot of the Advanced configuration page:
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The following window if for advanced configuration regarding IP, SIP, QoS, NAT, IP Telephony modes setting:
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Admin Password
Administrator password: Only the administrator can configure the “Advanced
Settings” page. Password field is purposely left blank for security reasons after Pressing
update and save. The maximum password length is 25 characters.
SIP Server
Outbound Proxy
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This field contains the URI string or the IP address (and port, if different from
5060) of the SIP proxy server. e.g., the following are some valid examples:
sip.my-voip-provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066
This field contains the URI string or the IP address (and port, if different from
5060) of the outbound proxy. If there is no outbound proxy, this field
SHOULD
proxy.
be left blank. If not blank, all outgoing requests will be sent to this outbound
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SIP User ID
This field contains the user part of the SIP address for this phone. e.g., if the
SIP address is: sip:my_user_id@my_provider.com, then the SIP User ID is:
my_user_id. Please do NOT include the preceding “sip:” scheme or the host
portion of the SIP address in this field.
SIP User ID
Authenticate ID
Authenticate
Password
Name
G723 Rate:
iLBC frame size
iLBC payload type
Preferred Vocoder
User account information, provided by VoIP service provider (ITSP), usually
has the digit form of a phone number (or is actually a phone number).
SIP service subscriber’s ID used for authentication. Can be
identical to or, different from SIP User ID.
SIP service subscriber’s account password for GXP-2000 to register to (SIP)
servers of ITSP.
SIP service subscriber’s name which will be used for Caller ID display.
This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is
chosen.
This defines the size of the iLBC codec frame. The default setting is 20ms.
This defines the iLBC payload type. The default setting is 97.
VoIP Client ATA supports up to 7 different vocoder types including G711-ulaw
(PCMU), G711-alaw (PCMA), G723, G729A/B, G726-32 (ADPCM), G728,
and iLBC. Depending on the product model, some of these vocoders may not
be provided in a standard release.
A user can configure vocoders in a preference list that will be included with the
same preference order in SDP message. The first vocoder in this list can be
entered by choosing the appropriate option in “Choice 1”. Similarly, the last
vocoder in this list can be entered by choosing the appropriate option in
“Choice 7”.
Silence Suppression
This controls the silence suppression/VAD feature of G723 and G729. If set to
“Yes”, when a silence is detected, a small quantity of VAD packets (instead of
audio packets) will be sent during the period of no talking. If set to “No”, this
feature is disabled.
Layer 3 QoS
This field defines the layer 3 QoS parameter which can be the value used for IP
Precedence or Diff-Serv. Default value is 48
Layer 2 QoS
This setting includes two fields. The 802.1Q/VLAN Tag contains the value
used for layer 2 VLAN tag. Default setting is blank. And 802.1p priority value
contains the value of the priority value.
Use DNS SRV
This parameter controls whether the IP phone supports the DNS SRV route
function.
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Voice Frames per
TX
This field contains the number of voice frames to be transmitted in a single
packet. When setting this value, the user should be aware of the requested
packet time (used in SDP message) as a result of configuring this parameter.
This parameter is associated with the first vocoder in the above vocoder
Preference List or the actual used payload type negotiated between the 2
conversation parties at run time.
e.g., if the first vocoder is configured as G723 and the “Voice Frames per TX”
is set to be 2, then the “ptime” value in the SDP message of an INVITE request
will be 60ms because each G723 voice frame contains 30ms of audio.
Similarly, if this field is set to be 2 and if the first vocoder chosen is G729 or
G711 or G726, then the “ptime” value in the SDP message of an INVITE
request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the
phone will use and save the maximum allowed value for the corresponding first
vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726,
it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64
(x10ms) and 64 (x2.5ms) frames respectively.
Fax Mode
User ID is phone
number
SIP Registration
Unregister On
Reboot
Registration
Expiration
Early Dial
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec
PCMU/PCMA)
If the VoIP Client ATA has an assigned PSTN telephone number,
then this field will be set to “Yes”. Otherwise, set it to “No”. If
“Yes”, a “user=phone” parameter will be attached to the
“From” header in SIP request.
This parameter controls whether the IP phone needs to send REGISTER
messages to the proxy server. The default setting is “Yes”.
Default is No. If set to Yes, the SIP user’s registration information will be
cleared on reboot.
This parameter allows the user to specify the time frequency (in minutes) the
phone will refresh its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes
(about 45 days).
This parameter controls whether the phone will attempt to send an early
INVITE each time a key is pressed when a user is dialing a number. If set to “Yes”, an
INVITE is sent using the dial-numbers collected so far; Otherwise, no
INVITE is sent until the “(Re-)Dial” button is pressed or after about 5 seconds
have elapsed if the user forgets to press the “(Re-)Dial” button.
The “Yes” option should be used ONLY if there is a SIP proxy configured and
the proxy server supports 484 Incomplete Address responses. Otherwise, the call
will most likely be rejected by the proxy (with a 404 Not Found error).
Please note that this feature is NOT designed to work with and should NOT be
enabled for direct IP-to-IP calling.
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Dial Plan Prefix
No Key Entry
Timeout
Use # as
Send Key
This value contains the dial plan prefix string (typically an ASCII numeric
string). If it is not blank, then this string will be used as a prefix to the target
URI string in the “To” header field of an INVITE message.
Default is 4 seconds.
This parameter allows the user to configure the “#” key to be used as the
“Send”(or “Dial”) key. Once set to “Yes”, pressing this key will immediately
trigger the sending of the
dialed string collected so far. In this case, this key is essentially equivalent to the “(Re)Dial”
key. If set to “No”, this # key will then
be included as part of the dial string to be sent out.
Local SIP port
Local RTP port
Use Random Port
keep-alive interval
Use NAT IP
Proxy-Require
NAT Traversal
This parameter defines the local SIP port the IP phone will listen and transmit
on. The default value is 5060.
This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit
on. It is the base RTP port for channel 0. When configured,
channel 0 will use this port value for RTP and the port_value+1 for its RTCP;
channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The
default value is 5004.
This parameter, when set to Yes, will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple IP phones are
behind the same NAT.
The VoIP Client ATA sends a UDP package to the SIP server periodically in
order to keep the port open on the router. This parameter defines the interval
time that HT286 send the UDP package. The default setting is 20 second.
NAT IP address used in SIP/SDP message. Default is blank.
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
This parameter defines whether the phone NAT traversal mechanism will be activated or
not. If activated (by choosing “Yes”) and a STUN server is also specified, then the phone
will behave according to the STUN client specification. Under this mode, the embedded
STUN client inside the phone will attempt to detect if and what type of firewall/NAT it is
behind through communication with the specified STUN server. If the detected NAT is a
Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will attempt to use
its mapped public IP address and port in all the SIP and SDP messages it sends out.
If this field is set to “Yes” with no specified STUN server, then the phone will
periodically (every 20 seconds by default) send a blank UDP packet (with no
payload data) to the SIP server to keep the “hole” on the NAT open.
Firmware Upgra de
This radio button will enable VoIP Client ATA to download firmware or configuration file
through either TFTP or HTTP.
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Via TFTP Server
This is the IP address of the configured tftp server. If it is non-zero or not
blank, the IP phone will attempt to retrieve new configuration file or new code
image (update) from the specified tftp server at boot time. It will make up to 3
attempts before timeout and then it will start the boot process using the existing
code image in the Flash memory. If a tftp server is configured and a new code
image is retrieved, the new downloaded image will be verified and then saved
into the Flash memory.
Note: DO NOT interrupt the TFTP upgrade process (especially the power
supply) as this will damage the device. Depending on the network environment
this process can take up to 15 or 20 minutes.
Via HTTP Server
Automatic HTTP
Upgrade
SUBSCRIBE for
MWI
Offhook
Auto-Dial
Enable Call Feature
The URL for the HTTP server used for firmware upgrade and configuration via
HTTP. For example,
http://provisioning.mycompany.com:6688/Grandstream/1.0.5.16
Here “:6688” is the specific TCP port that the HTTP server is listening at, it can
be omitted if using default port 80.
Note: If Auto Upgrade is set to “No”, VoIP Client ATA will only do HTTP
download once - at boot up.
Choose “Yes” to enable automatic HTTP upgrade and provisioning.
In “Check for new firmware every” field. Enter the number of days period.
VoIP Client ATA will check the HTTP server for firmware upgrade or
configuration after the defined number of days.
When set to “No”, VoIP Client ATA will only do HTTP upgrade once at boot
up.
Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
This parameter allows the user to configure a User ID or extension number to
be automatically dialed upon offhook. Please note that only the user part of a
SIP address needs to be entered here. The phone will automatically append the
“@” and the host portion of the corresponding SIP address.
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported
(locally).
Disable Call
Waiting
Send DTMF
DTMF Payload
Type
Default is No.
This parameter controls the way DTMF events are transmitted. There are 3
ways: in audio which means DTMF is combined with the audio signal (not very
reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
This parameter sets the payload type for DTMF using RFC2833
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Send Flash Event
This parameter allows the user to control whether to send an SIP NOTIFY
message indicating the Flash event, or just to switch to the voice channel when
the user presses the Flash key.
FXS Impedance
Caller ID Scheme
Onhook Voltage
Polarity Reversal
NTP server
Send Anonymous
Lock keypad
update
Selects the impedance of the analog telephone connected to the Phone port.
Select the Caller ID Scheme to suit the standard of different area.
• Bellcore (North America)
• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
• ETSI-DTMF (Finland, Sweden)
• DTMF (Denmark)
Select the onhook voltage to suit different area or PBX.
Select Polarity Reversal to adapt some call charge/billing system. Default is
No.
This parameter defines the URI or IP address of the NTP server which the IP
phone will use to display the current date/time.
If this parameter is set to “Yes”, the “From” header in the outgoing INVITE
message will be set to anonymous, essentially blocking the Caller ID from
being displayed.
If this parameter is set to “Yes”, the configuration update via keypad is
disabled.
Syslog Server
The IP address or URL of the System log server. This feature is especially useful
for ITSP (Internet Telephone Service Provider)
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Syslog Level
Select the ATA to report the log level. Default is NONE. The level is one of
DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on
the following events:
• product model/version on boot up (INFO level)
• NAT related info (INFO level)
• sent or received SIP message (DEBUG level)
• SIP message summary (INFO level)
• inbound and outbound calls (INFO level)
• registration status change (INFO level)
• negotiated codec (INFO level)
• Ethernet link up (INFO level)
• SLIC chip exception (WARNING and ERROR levels)
• memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it
contains the following components:
GS_LOG: [device MAC address][error code] error message
Here is an example:
May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet
link is up
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3.2.1.4 Saving the Configuration Changes
Once a change is made, the user should press the “Update” button in the Configuration Menu. The IP
phone will then display the following screen to confirm that the changes have been saved.
Users are recommended to power cycle the VOIP Client-488 after seeing the above message.
3.2.1.5 Remotely rebooting VoIP Client ATA
The administrator of the phone can remotely reboot the phone by pressing the “Reboot” button, at the Configurations menu
button. Once done, the following screen will be displayed to indicate that
rebooting is underway
.
At this point, the user can relogin to the phone after waiting for about 30 seconds.
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3.3 Restoring the Factory Default Settings
Warning !!!
Restoring the Factory Default Settings will DELETE all configuration information of the device.
Please backup or print out all the settings before attempting the following steps.
Please disconnect the network cable and power cycle the unit, before trying to reset the unit to the factory defaults.
The steps are as follows:
Step 1:
bottom of the device. It is a 12 digits hex’ number.
Step 2
0-9: 0-9
A: 22
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, for the MAC address: 00 0b 82 00 e3 95,
the User encoding should be : “00 0222 82 00 333 3 95”
Find the MAC Address of the device. The MAC address of the device is located at the
: Encode
the “MAC address to decimal” digits. Please use the following mapping:
Step 3
prompt “RESET”
Step 4:
correct encoded MAC address is entered, the device will reboot automatically and restore the
factory default settings.
: Access the voice menu by pressing *** or the LED button, then dial “99” and get the voice
Key in the encoded MAC address decimal digits after hearing the IVR prompt. Once the
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3.4 VoIP Master
VoIPMaster
3.4.1 What is the VoIP Master and how it works
This device connects GSM cellular telephones to the internet, by way of VoIP (Voice over Internet Protocol). A GSM module,
including a SIM card, is installed inside the VoIP device. A SIM card is a smart card that is received with a subscription to a
cellular telephone network. This following is the communication solution architecture enabled by the VoIPMaster:
One Location in the
World
GSM
Cellphones
VoIP
Master
GSM
Cellphones
IP Network
VoIP
Near 0 Cost!
Another Location in the
World
GSM
Cellphones
GSM
Base Stations
VoIP
Master
Management
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1.
From your cellular phone, you can dial to the VoIP device
2.
The GSM module in the VoIP device provides you with
.
a dial tone of Voice over Internet Protocol.
3
. You can now dial and make a telephone connection by way of the internet which has near 0 cost.
Main usage features:
•
Up to
32 cellular phon es
can use the VoIP device for connection to the internet in parallel.
• A local desktop telephone can be connected to the VoIP device. The desktop phone can send and receive calls via
the internet, as well as via the GSM network (according to telephone prefixes).
• A “
follow me
” function can be activated to serve the
desktop phone
.
• If two systems install in remote offices a call from a mobile in one location – let say N.Y can call a remote
cellular user let say in Japan in the cost of a local enterprise Cellphone cost only!
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3.4.2 Set-up and Installation
Insert SIM and connect the cables as described below.
1. Insert
a. Using the tip of the antenna (or a similar object), press on the small yellow button on
the left side of the gateway, as pictured below. A SIM drawer pops out.
b. Insert the SIM into the drawer as pictured below. Ensure that the angled notch of the
SIM is in the matching corner of the SIM drawer (upper left corner). Ensure that the
SIM is flat in the drawer.
c. Return the SIM drawer to the SIM slot on the left side of a Free Gate.
2. Connect cables as follows:
a.
b. Insert the communication cable from the PC
the left side of the
c. Insert the network cable into a socket on the right side of the gateway and
connect it to the computer with the internet connection.
SIM card into the VoIP Master as follows:
Insert the antenna to a connector on the right side of the VoIP Master.
VoIPMaster
COMport
.
to the serial COM port socket on
PC
COM
Port
RJ-45
RJ-11
To Internet
d. Insert the telephone jack, from your land line telephone, into a socket on the left side of the gateway.
e. Plug the transformer into a wall socket and insert the power cable into its socket on the left side of the gateway.
After connecting the cables, install the Manager and configure the settings for the VoIP Master gateway as described in the
following chapters.
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3.4.3 Installing the Manager Application
Before operation, configuration settings must be made in the VoIP Master gateway. Configuration is done by a manger
application in the computer. Install the manager application on the software cd, then define the comport connection, as
described in this chapter.
1. Insert the VoIP Master CD into the computer drive.
2. In Windows Explorer, navigate to Icon (in the software disk).
3. Double click the Icon, wait till the installation window will open.
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4. Click Next. The Setup Type window opens.
5. Select “Complete” and click “Next”. The “Begin Installation” window opens.
VoIP Master
6. Click Install. Wait till the VoIP Master Manager application will install itself.
3.4.4 Define the Com port Connection
After installing the manager application, launch it and define the Comport to which the
VoIP Master is connected.
1. Launch the PRI Manager by pressing on your computer desktop, or by
pressing
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Start > Programs > EuroTech Communications > VoIP Master Manager.
The VoIP Master Manager window opens.
VoIP Master
2. In the toolbar, press . The Select Connection window opens.
3. Select the Com Port in the computer to which the VoIP Master is connected.
The connection indicators in the lower right corner of the window blink green
:
After installing the Manager and defining the port connection, define port and SIM
Settings as described in the following chapter.
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3.4.5 Port and SIM Settings
This chapter details port to SIM association as well as required and optional settings.
3.4.5.1 Dial Settings for the GSM Port
After defining the Comport, press
GSM Port
in the left pane. The Port Setting window is displayed.
Define dial settings in this window as follows:
1. In the Dial pause box, set the
time interval
, whereupon a
dialed number is dispatched
after the designated
delay time. Each unit is 0.1 second. For example, if you want the number to be dispatched 3 seconds after you
finish dialing, enter 30 in this box.
2. Upon completion of a call, if you want to remain connected to the
VoIP
to On
.
3. Set
Receiving (Rx)
and transmitting volumes in the Rx (receiving) and
GSM Network
Transmitting (Tx)
, set
Repea t Access
volume
settings.
4. Press
Write Settings.
to
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3.4.5.2 SIM Settings
After making the port settings, press a
VoIP Master
SIM
icon in the left pane. The
SIM
Setting window opens
.
1
. In the PIN Code box, enter the PIN number of the SIM.
2. In the Network box, enter the GSM network number of the SIM.
3. In boxes 1 through 8, set enter telephone number prefixes to which this SIM can dial.
4. Press
Write Settings
.
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3.4.5.3 Follow Me Settings
If there is no answer on the local phone connected to your VoIP gateway you can use a “
Follow me
feature. The “Follow me” feature connects the incoming call to your cellular phone. To activate, after
making SIM settings, press Follow me in the left pane. The Follow Me Setting window opens.
VoIP Master
”
1.
Set the
Mode box
toON.
2. In the Rings Number box, enter the number of times the local phone will ring before being diverted to the "
function.
3. In the
Called Number box
, enter the telephone number that you want dialed when the "
follow me"
activated.
4. Press
Write Settings
.
Follow me
function is
"
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3.4.5.4 Call Back Settings
A call back feature is available with the VoIP master gateway. If person A is having a conversion via the VoIP master
gateway, and person B attempts to make a phone call via the same VoIP:
1. Person B will receive a busy signal.
2. If the telephone number of
call of
person A
is
completed
person B
is listed in the
, the VoIPMaster gateway will
Call Back settings
previously busy by person A.
To enable this feature perform the following:
1. On the right side of the window, set the box to ON.
2. Enter desired telephone numbers in the center of the window.
3. Next to each telephone number, set the box to ON.
4. Press
Write Settings
VoIP Master
.
of the VoIPMaster manager, when the phone
call person B
and
provide a telephone line that was
After making these settings, your VoIP gateway is ready for operation.
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