Grandstream Networks Network Card GSM gateway User Manual

VoIPMaster™

Version 4.x

VoIP to GSM gateway

Connecting Cellular Phones directly to

‘Voice over IP’

worldwide networks

User Manual

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Download from Www.Somanuals.com. All Manuals Search And Download.

TABLE OF CONTENTS

Getting Started................................................................................................................................................. 5

Check your package Items............................................................................................................................... 7

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 Attended Transfer................................................................................................................................ 12

3.1.5 Call Features.......................................................................................................................................... 13

3.1.6 Fax Support ........................................................................................................................................... 13

3.1.7 LED Light Pattern Indication.................................................................................................................... 14

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

Februaray-2006

Page 4 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

1 Getting Started

Eurotech Communication team is glad you have chosen to use the Eurotech’s VoIPMaster GSM to VoIP gateway

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 pleasant tasks as possible. We wish you a smooth operation while greatly saving

your office mobile phone calls.

This chapter is your “Map for installation, configuration and monitoring tasks” and includes a short explanation

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 over 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 Check your package Items at Chapter 2 “Check your package Items”.

Later proceed with the Gradstream HanyTone 286 VoIP Client ATA (Analog to Telephone Adaptor) which

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 VoIPMaster gateway adds new capabilities of GSM to VoIP calls origination and termination to the client ATA.

The following topics of the VoIP Client ATA are reviewed in Chapter 3:

Client ATA Product Overview, Key Features, Hardware Specification and Basic Operations as follows



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

Now you shall start configure the VoIP Client ATA following the Configuration Guide at chapter 3.2.

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 VoIP Client ATA you can start configuring the VoIP Master, starting with learning

the VoIPMaster concept rule in the network and the way it works at “What is the VoIP Master and how it works

chapter.

”

Februaray-2006

Page 5 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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

Download from Www.Somanuals.com. All Manuals Search And Download.

2 Check your package Items

Please verify your package contains the following components (some were ordered specific) before installation:



Main Hardware Device – The VoIP Master Gateway

110/220V Electric Power converter to 24V with cables supplied

VoIP master software Installation CD - Installation kit for MS-Windows Management Application, this User

Manual file and additional auxiliary utilities.



GSM Antenna – To be installed to the VoIP Master Gateway

RS-232 Serial PC COMport connection cable we’ll be referred as Comport cable in this manual

Februaray-2006

Page 7 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Februaray-2006

Page 8 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.



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

Button

1xRJ45 10Base-T

LED

GREEN & RED color

Universal

Power Adaptor

Input: 100-240VAC

Output: +5VDC, 1200mA

UL certified

65mm (W

93mm (D)

27mm (H)

Dimension

Weight

Operating

Temperature

32 - 104oF

0 - 40oC

10% - 95%

(non-condensing)

Humidity

Compliance

FCC/CE/C-Tick

Februaray-2006

Page 9 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Main Menu

Voice Prompt

“Enter a Menu Option”

User’s Options

Enter ‘*’ to next option and “#” back to main menu, or

Dial 01 – 06, 47, 86 or 99 Menu option

“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.

“IP Address” + IP address

The current WAN IP address is announced. Enter 12-digit

new IP address if in Static IP Mode.

“Subnet” + IP address

Same as Menu option 02

TFTP server is used to update the firmware of the device.

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”.)

If there are voice messages, user can dial ‘9’ and dial pre-

configured phone number to retrieve voice message.

“Gateway “ + IP address

“DNS Server” + IP address

“TFTP Server “ + IP address

“Direct IP Calling”

“No Voice Messages”; or

“Voice Messages Pending”

“RESET”

Dial ‘9’ to confirm the RESET; or

Enter MAC address to restore factory

default setting (For detail, see section 8)

Automatically return to Main Menu

“Invalid Entry”

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

Februaray-2006

P_Doa_wg_ne_loa1_d0_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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 calls

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

. (dot character)

: (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.

Februaray-2006

P_Doa_wg_ne_loa1_d1_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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”.

Februaray-2006

P_Doa_wg_ne_loa1_d2_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

3.1.5 Call Features

The Following table shows the call features of VoIP Client ATA.

Key

*30

*31

*67

*82

*50

*51

*70

*71

Call Features

Block Caller ID (for all subsequent calls)

Send Caller ID (for all subsequent calls)

Block Caller ID (per call)

Send Caller ID (per call)

Disable Call Waiting (for all subsequent calls)

Enable Call Waiting (for all subsequent calls)

Disable Call Waiting. (Per Call)

Enable Call Waiting (Per Call)

Unconditional Call Forward.

*72

*73

*90

To use this feature, dial “*72” and get the dial tone. Then dial the forward number and “#” for a dial tone, then

hang up.

Cancel Unconditional Call Forward

To cancel “Unconditional Call Forward”, dial “*73” and get the dial tone, then hang up.

Busy Call Forward

To use this feature, dial “*90” and get the dial tone. Then dial the forward number and “#” for a dial tone, then

hang up.

Cancel Busy Call Forward

To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then hang up

Delayed Call Forward

*91

*92

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.

Februaray-2006

P_Doa_wg_ne_loa1_d3_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

3.1.7 LED Light Pattern Indication

Following are the LED light pattern indications

RED LED indicates abnormal status

flash every 2 seconds (if DHCP is configured)

flash every 2 seconds (if SIP is configured)

DHCP Failed or WAN No Cable

VOIP Client-486 fails to register

GREEN LED indicates normal working status

Message Waiting Indication

RINGING

Button flashes every 2 seconds

Button flashes at 1/10 second

Button flashes every second

RINGING INTERVAL

Februaray-2006

Page 14 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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 is the IP address of the phone.

http://Phone-IP-Address where the Phone-IP-

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

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.

Februaray-2006

Page 15 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Februaray-2006

Page 16 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

Here are the status details shown

Februaray-2006

Page 17 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

MAC Address

The device ID, in HEX format. This is very important ID for ISP troubleshooting.

This field shows WAN port IP address.

WAN IP Address

Product Model

This field contains the product model info.

Program: This is the main software release. This number is always used for

firmware upgrade.

Bootloader: This is normally not changed.

HTML: This is the user interface, normally not changed.

VOC: This is the codec program, normally not changed.

Software Version

System Uptime

Registered

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

PPPoE Link Up

This shows what kind NAT the VoIP Client ATA is connected to via its WAN port. It is

based on STUN protocol.

NAT

NAT Mapped IP

WAN side public IP if connected to LAN of a SOHO router.

NAT Mapped Port

External port detected by STUN

Statistical Status

Self explainable. Please refer to the page displayed.

Februaray-2006

Page 18 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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:

Februaray-2006

Page 19 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

The following window if for advanced configuration regarding IP, SIP, QoS, NAT, IP Telephony modes setting:

Februaray-2006

Page 20 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Admin Password

SIP Server

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 be left blank. If not blank, all outgoing requests will be sent to this outbound

proxy.

Outbound Proxy

Februaray-2006

Page 21 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

Februaray-2006

Page 22 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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

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 User ID

SIP service subscriber’s ID used for authentication. Can be

Authenticate ID

identical to or, different from SIP User ID.

SIP service subscriber’s account password for GXP-2000 to register to (SIP)

servers of ITSP.

Authenticate

Password

SIP service subscriber’s name which will be used for Caller ID display.

Name

This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is

chosen.

G723 Rate:

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.

iLBC frame size

iLBC payload type

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.

Preferred Vocoder

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”.

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.

Silence Suppression

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 3 QoS

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.

This parameter controls whether the IP phone supports the DNS SRV route

function.

Use DNS SRV

Februaray-2006

Page 23 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

Voice Frames per

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

T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec

PCMU/PCMA)

User ID is phone

number

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.

SIP Registration

This parameter controls whether the IP phone needs to send REGISTER

messages to the proxy server. The default setting is “Yes”.

Unregister On

Reboot

Default is No. If set to Yes, the SIP user’s registration information will be

cleared on reboot.

Registration

Expiration

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).

Early Dial

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.

Februaray-2006

Page 24 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Dial Plan Prefix

No Key Entry

Timeout

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

Use # as

Send Key

be included as part of the dial string to be sent out.

Local SIP port

Local RTP port

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.

Use Random Port

keep-alive interval

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.

Use NAT IP

NAT IP address used in SIP/SDP message. Default is blank.

Proxy-Require

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.

NAT Traversal

Firmware Upgrade

This radio button will enable VoIP Client ATA to download firmware or configuration file

through either TFTP or HTTP.

Februaray-2006

Page 25 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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

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.

Automatic HTTP

Upgrade

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.

SUBSCRIBE for

MWI

Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting

Indication will be sent periodically.

Offhook

Auto-Dial

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.

Enable Call Feature

Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported

(locally).

Disable Call

Waiting

Default is No.

Send DTMF

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.

DTMF Payload

Type

This parameter sets the payload type for DTMF using RFC2833

Februaray-2006

Page 26 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Send Flash Event

FXS Impedance

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)

Caller ID Scheme

• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)

• ETSI-DTMF (Finland, Sweden)

• DTMF (Denmark)

Onhook Voltage

Polarity Reversal

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.

NTP server

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.

Send Anonymous

Lock keypad

update

If this parameter is set to “Yes”, the configuration update via keypad is

disabled.

The IP address or URL of the System log server. This feature is especially useful

for ITSP (Internet Telephone Service Provider)

Syslog Server

Februaray-2006

Page 27 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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:

Syslog Level

• 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

Februaray-2006

Page 28 / 42

MasterVoIP

VoIP to GSM Gateway

Download from Www.Somanuals.com. All Manuals Search And Download.

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.

Download from Www.Somanuals.com. All Manuals Search And Download.

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: Find the MAC Address of the device. The MAC address of the device is located at the

bottom of the device. It is a 12 digits hex’ number.

Step 2: Encode the “MAC address to decimal” digits. Please use the following mapping:

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”



Step 3: Access the voice menu by pressing *** or the LED button, then dial “99” and get the voice

prompt “RESET”

Step 4: Key in the encoded MAC address decimal digits after hearing the IVR prompt. Once the

correct encoded MAC address is entered, the device will reboot automatically and restore the

factory default settings.

Februaray-2006

P_Doa_wg_ne_loa3_d0_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

3.4 VoIP Master

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:

Another Location in the

World

One Location in the

World

GSM

Cellphones

GSM

Base Stations

GSM

Cellphones

GSM

Cellphones

IP Network

VoIP

Master

VoIP

Master

Near 0 Cost!

VoIPMaster

Management

Februaray-2006

P_Doa_wg_ne_loa3_d1_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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.

. 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 phones 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!

Februaray-2006

P_Doa_wg_ne_loa3_d2_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

3.4.2 Set-up and Installation

Insert SIM and connect the cables as described below.

1. Insert SIM card into the VoIP Master as follows:

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. Insert the antenna to a connector on the right side of the VoIP Master.

b. Insert the communication cable from the PC COMport to the serial COM port socket on

the left side of the VoIPMaster

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.

COM

Port

anagement PC

RJ-45

RJ-11

To Internet

Landline Phone

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.

Februaray-2006

P_Doa_wg_ne_loa3_d3_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

Februaray-2006

P_Doa_wg_ne_loa3_d4_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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.

Februaray-2006

P_Doa_wg_ne_loa3_d5_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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

VoIP to GSM Gateway

Februaray-2006 P_Doa_wg_ne_loa3_d6_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search And Download.}

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.

Februaray-2006

P_Doa_wg_ne_loa3_d7_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

Februaray-2006

P_Doa_wg_ne_loa3_d8_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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 GSM Network, set Repeat Access to

VoIP to On

3. Set Receiving (Rx) and transmitting volumes in the Rx (receiving) and Transmitting (Tx) volume

settings.

4. Press Write Settings.

Februaray-2006

P_Doa_wg_ne_loa3_d9_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

3.4.5.2 SIM Settings

After making the port settings, press a SIM icon in the left pane. The SIM Setting window opens

VoIP Master

. 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

Februaray-2006

P_Doa_wg_ne_loa4_d0_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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 to ON

2. In the Rings Number box, enter the number of times the local phone will ring before being diverted to the "Follow me

function.

3. In the Called Number box, enter the telephone number that you want dialed when the "follow me" function is

activated.

4. Press Write Settings

Februaray-2006

P_Doa_wg_ne_loa4_d1_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

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 person B is listed in the Call Back settings of the VoIPMaster manager, when the phone

call of person A is completed, the VoIPMaster gateway will call person B and provide a telephone line that was

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

After making these settings, your VoIP gateway is ready for operation.

Februaray-2006

P_Doa_wg_ne_loa4_d2_fro/_m42_Www.SomM_anua_as_lst_.e_cor_mV_.o_AI_llP_{Manuals Search}V_Ao_nI_dP_Dt_oo_wnG_loS_adM_.Gateway

Gravely Lawn Mower 990100 User Manual
Haier Air Conditioner AS072AZNBA User Manual
Haier Freezer HUF138PA User Manual
Harbor Freight Tools Air Compressor 92403 User Manual
Harbor Freight Tools Automobile Parts 66632 User Manual
Hasbro Motorized Toy Car 22613 User Manual
Healthrider Treadmill HMTL796080 User Manual
Honeywell Weather Radio C7735A1000 User Manual
HP Hewlett Packard Server BL460C G7 User Manual
Image Home Gym 831159700 User Manual