Etross GoIP User Manual

User Manual

Single Channel GSM Gateway

Model: GoIP

Release 1.2

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Contents

1 Introduction .................................................................. 3

1.1 Overview ............................................................................................................................. 3

1.2 Protocol ............................................................................................................................... 4

1.3 Hardware Specification ....................................................................................................... 5

1.4 Software Specification ........................................................................................................ 5

1.5 List of the Package .............................................................................................................. 5

1.6 Appearance .......................................................................................................................... 6

2 Installation .................................................................... 8

2.1 Connection Diagram ........................................................................................................... 8

2.3 LED Indicators .................................................................................................................... 9

2.4 SMS Commands ................................................................................................................. 9

3 Configuration Guide .................................................. 10

3.1 Web Configuration Menu .................................................................................................. 11

3.2 Status ................................................................................................................................. 11

3.2.1 Phone Information .............................................................................................. 12

3.2.2 Network Information ........................................................................................... 13

3.2.3 GSM Module Informat ion .................................................................................. 13

3.3 Configurations ................................................................................................................... 13

3.3.1 Language ............................................................................................................. 14

3.3.2 Time Zone and Ti me Server ............................................................................. 15

3.3.3 Auto-Provision ..................................................................................................... 15

3.3.4 Network Tone ...................................................................................................... 15

3.3.5 GSM Group Mode .............................................................................................. 16

3.3.6 GSM Caller ID Anonymous ............................................................................... 18

3.3.8 Auto Reboot ......................................................................................................... 18

3.3.7 Remote Server .................................................................................................... 18

3.4 Call Settings ...................................................................................................................... 18

3.4.1 H.323 Phone ....................................................................................................... 19

3.4.1.1 Single Configuration ................................................................................... 19

3.4.1.2 Configuration by Group .............................................................................. 20

3.4.1.4 Advance Settings ......................................................................................... 20

3.4.1.5 H.323 Direct Mode ...................................................................................... 22

3.4.2 SIP Phone ............................................................................................................ 22

3.4.2.1 Advanced Settings ....................................................................................... 23

3.4.3 Media Setting ...................................................................................................... 25

3.4.4 Codec Preference ............................................................................................... 27

3.4.5 NAT T rav ersal ...................................................................................................... 27

3.4.5.1 Signaling NAT Traversal ............................................................................. 27

3.4.5.2 Media NAT Traversal .................................................................................. 28

3.5 Call Divert ......................................................................................................................... 29

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3.5.1 Dial Plan ................................................................................................................. 30

3.5.1.1 Basic Syntax ................................................................................................ 30

3.5.1.2 Advanced Syntax for Limiting Number Length .......................................... 31

3.6 SMS Mode ........................................................................................................................ 32

3.6.1 SMS Dial Mode in SIP ........................................................................................... 32

3.6.2 SMS Dial Mode in H,323 ............................................................................... 35

3.6.3 SMS Relay Mode (FOR SIP ONLY) ............................................................. 38

3.7 Relay Incoming Caller ID (GSM to VoIP Call) ................................................................ 40

3.8 Gain Settings… ................................................................................................................. 41

3.9 Network Configuration ..................................................................................................... 42

3.9.1 LAN Port .............................................................................................................. 42

3.9.2 PC port configurations ....................................................................................... 43

3.10 Save Configuration ......................................................................................................... 44

3.11 Discard Changes .............................................................................................................. 45

3.12 To ols Menu...................................................................................................................... 45

3.12.1 Online Upgrade ................................................................................................ 45

3.12.2 Change Password ............................................................................................ 46

3.12.3 Reset Configuration ......................................................................................... 46

3.12.4 Reboot the Device ............................................................................................ 46

4 Hardware Specifications ........................................... 47

5 Useful Factory Default Settings ................................ 48

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1 Introduction

1.1 Overview

A VoIP GSM gateway (GoIP) is an IP-based device that enables inbound and outbound
VoIP and GSM cellular calls. It is an alternative to a VoIP FXO gateway, especial ly in
area where GSM service i s readily av ailable and c heaper for V oI P ca ll term ination. Many
applications can be evolved from this technology using GSM termination, for examples,
distributed call centers, VoIP termination, and cell phone roaming. A VoIP GSM gat eway
is not only a great way to prov ide f ast deployment but also provides significant savings in
usage, infrastructure and ma int enance cost compared to conventional FXO gateway s.

The diagram below shows a t ypical topology for a VoIP GSM gateway.

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1.2 Protocol

 TCP/IP V4 (IP V6 auto adapt)
 ITU-T H.323 V4 St andar d
 H.2250 V4 Sta ndard
 H.245 V7 Stan dard
 H.235 Sta ndard （MD5，HMAC-SHA1）
 ITU-T G.711 Alaw/ULaw , G.729A, G.729AB, and G.723.1 Voice Codec
 RFC1889 Real T ime Dat a Transmission
 Proprietary Firewall-Pass-Through T ech nology
 SIP V2.0 St an dard
 Simple Traversal of UDP over NAT (STUN)
 Web-base Management
 PPP over Ethernet (PPPoE)
 PPP Authentication Protocol (PAP)
 Internet Control Message Pr ot ocol (ICMP)
 TFTP Client
 Hyper Text T rans fer Protocol (HTTP)
 Dynamic Host Configurat i on Protocol (DHCP)
 Domain Name System (DNS)
 User account authentication using MD5
 Out-band DTMF Relay: RFC 2833 and SI P Info

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1.3 Hardware Specification

 Embedded Processor
 DSP for voice codec and voice processing
 Two 10/100 BaseT Ethernet ports with full compliant with IEEE 802.3
 LEDs for Ethernet port status
 One GSM Connection:

o Huawei GSM Module - GSM 850 MHz/GSM 900 MHz
o Simcom GSM Module – GSM 850 MHZ, 900MHz, DCS 1800 MHz, PCS 1900

MHz

1.4 Software Specification

 Embedded LINUX OS
 Built-in HTTP Web Server
 PPPoE Dial-up
 NAT Broadband Router Functions
 DHCP Client
 DHCP Server
 Firmware On-line upgrade
 Caller ID
 Multiple Language Suppor t
 Support Multi devices Cooperate Mode

1.5 List of the Package

a) One GoIP gateway ma in unit
b) One DC4.5V/2000mA power adapt or
c) One Ethernet cable (3M)

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1.6 Appearance

VoIP GSM Gateway (GoIP) – Front View

VoIP GSM Ga teway (GoIP) – Rear View

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1

） LAN

Connect this port to an Ether net switch/router, the Ethernet of a DSL modem, or
other network access equipm ent .

2

） PC

Connect a computer or ot her net work device to this port.

3

） POWER (DC4.5V/2000mA)

Connect the 4.5V/2000mA adapter provided to this p ow er jack.

4

） Reset

Press this button to reset the GoIP gateway to factory defaults.

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2 Installation

2.1 Connection Diagram

Please follow the connecti on di agr am above to install the GoIP gateway.

a) Insert a GSM SIM card in the SIM card compartment located at the bottom of the

GoIP gateway.

b) Connect an Ethernet cable the LAN port of the GoIP gateway and the other end to

your existing network equipm ent.

c) Connect an Ethernet cable to the PC port of the GoIP gateway and t he other end to a

PC or other network device (optional).

d) Connect the power adapter pr ovided to the power jack of the GoI P gateway.

The diagram below shows a typical installation of the device.

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2.3 LED Indicators

The following table defines the status of the LEDS located on t he t op case and on the
RJ-45 connectors.

LED DESCRIPTION

RUN

1. When the GoIP is booting，this LED will
flash 100ms ON and 100ms OFF.

2. When the GoIP is properly registered to
your softswitch, this LED flashes at a
rate of 1s ON and 1s OFF.

GSM When the GSM channel is ready to sue, this

LED flashes at a rate of 1s ON and 1s OFF.

2.4 SMS Commands

GoIP supports the following management co m man ds via GSM SMS messages.

Function Command (SMS

Content)

Remarks

Obtain LAN Port
IP Address

INFO or info Case Non-sensitive

Reset GoIP
Configuration

RESET<Password> Key word “RESET” not case sensitive

Reboot GoIP REBOOT <Password> Key word “REBOOT” not case sensit iv e

1》 Obtain LAN Port IP Address

Once the GSM SMS with message content “info” or “INFO” is received, the GoIP
sends back a SMS message to the sender with the message content containing the
LAN Port IP address.

2》 Reset GoIP Configuration

Upon receiving the SMS message “RESET <password>” or “reset <password>”, the
GoIP reset its con figurations to factory defaults.

3》 Reboot GoIP

Upon receiving the SMS message “REBOOT <password>” or “reboot <password>”,
the GoIP reboots itself automatically.

Note: <password> is t he webpage login password as described in Section 3.1.

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3 Built-in Web Server

To configure the GoIP gateway, you must login to its Web server via the LAN or PC port.
The LAN port is factory preset t o obtain an IP address from the local DHCP host and the
PC port is set to the fixed IP address 192.168.8.1.

If a local DHCP host is av ailab le, the LA N will o bt ain an I P address a uto matic ally. T o liste n
to the IP address a ss igne d, just dial a call via the GoIP gateway’s SIM card phone number .
When the call is connected, you will hear a dial t one. Just dial “*01#” for English voice
prompt for the LAN IP address and “*00#” for Chinese voice prompt for the LAN IP
address. The LAN IP address can also be obt ain ed by sending a SMS message to t he
GSM phone number. The GoIP will then reply with a SMS mess age co nt a ining the LAN IP
address.

If you want obtained LAN port IP by sending a SM S message, please send” INFO “or”
info”.

If a local DHCP host is not available, you can access the GoIP gateway via the PC port.
You will need to change t he PC LAN configuration vi a t he Network Connections und er t he
Control Panel.

Windows: Control Panel--Network Connections--Local
Connectionism’s Property--TCP/IP Protocol’s Property

Set an unused IP address that is in the same segment as the PC port addr ess.

Once the IP address of the LAN or PC port is known, you are now ready to
access the Web server of GoIP gateway.

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3.1 Web Configuration Menu

If your PC is connected to the GoIP gateway via the LAN port network segmen t , you need
to type the LAN IP address of the GoIP gateway in your Web browser to access the Web
server of the GoIP gateway. If not, you should type the PC IP address (192.168.8.1) in
the Web browser.

If the connection is correct, the Web browser will pro mpt you to enter the “User name” and
“Password: as shown below.

Enter the User name and Password and the press OK to access the GoIP Gateway Web
Server. The default for both user name and password is “admin”. The default built-in
webpage is then shown below.

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3.2 Status

The St at us page shown below is the de fau lt / hom e page of the GoIP Web server.

It consists of 3 columns status information of the GoIP and they are:

a. Phone Information
b. Network Information
c. GSM Module Information

3.2.1 Phone Information

A. Serial Number

Each gateway has a unique se r i al number assigned by the factory, for example,

HT304O12VTEST01. This number is important for centralized configuratio n, t echnical

support, and warranty. This number is printed on the bottom of the gateway and is
associated with your software license.

B. Firmware Version

Firmware version identifies the firmware version of the gateway, for example,

GHS-3.01.

C. Hardware Mode

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This field shows terminal’s hardware type.

D. Phone Status

This field shows the statu s of l ine’s connection status. If the connection is successful,
this field displays LOGIN; otherwise, it displays LOGOUT.

3.2.2 Netwo rk Information

A. LAN Port Configuration

This field displays the status of the LAN port.

B. PC Port Configuration

This field displays the status of the LAN port.

C. PPPoE

If PPPoE is enabled, it displays its status.

D. Default Route

This field displays the IP address of the default routing gateway.

E. DNS Server

This field displays the IP address of the domain name server.

3.2.3 GSM Module Information

A. GSM Module

This field displays the GSM mo dul e type.

B. GSM Signal

This field displays the GSM signal type.

C. GSM Status

This field shows the statu s of GSM c onnection status. If the connection is successful,
this field displays LOGIN; ot her w ise, it displays LOGOUT.

3.3 Configurations

Click on the “Configurations” tab on the left hand column to access the device
configuration menu: Preference, Network, Call Settings, Call Divert, Save Changes,
and Discard Cha nges.

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Click on “Preference” in the lef t menu of the configuration web, and t he screen will be
displayed as below:

3.3.1 Language

Currently GoIP suppor ts English, simplified Chinese and traditional Chinese. Select the
language desired and the Web page will be shown in t he lan guage selected accordingly.

The language can also be selected at the top of the web page. Once selected, the

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webpage language is refreshed immediately. However, the language selecti on is not
saved until the Save Changes icon is clicked.

3.3.2 Time Zone and Time Server

The GoIP gateway supports Network Time Protocol (NTP) to obtain the date and time
information from an NTP server (Time Server). The time zone is specif ied as in GMT ±
offset. For example, the Pacific S t an dard Time is GMT -8, and the Pacific Daylight Time
is GMT-7.

Note: The GoIP gateway su pports CDR and billing information, it is important to set up
these two parameters pr operly.

3.3.3 Auto-Provision

The GoIP Gateway suppo rts Auto Provis ioning w hic h enab les con figur ation p a ram eters t o
be set automatically without human intervention. The Auto Provisioning supports both
HTTP and TFTP protocols. For higher security, encrypted configuration file is also
supported. This feature requires external Auto Provisioning Server. Please contact
your service provider for f urther information on this.

3.3.4 Network Tone

Network tones are a set of tones used for VoIP calls. Select one of the countries defined
or customized to define your own Network Tones.

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You can configure the Net w or k Tones as Custom ized option:

Each tone listed above is def ined in the following format:

nc, rpt, c1on, c1off, c2on, c2of f, c3on, c3off, f1, f2, f3, f4, p1, p2, p3, p4

Where:

nc is the number of cadences
rpt is the repeat counter(0 - infinite, 1~n - repeat 1~n times)
c1on is the cadence one on (in milliseconds)
c1off is the cadence one off (in millise conds)
c2on is the cadence two on (in milliseconds)
c2off is the cadence two off (in milliseconds)
c3on is the cadence three on (in milliseconds)
c3off is the cadence three of f (in milliseconds)
f1 is the tone #1 frequency (300Hz-3000Hz)
f2 is the tone #2, frequency (300Hz-3000Hz)
f3 is the tone #3 frequency (300Hz-3000Hz)
f4 is the tone #4 (300Hz-3000Hz)
p1 is the attenuation index for f1, 0~31(0=3 dB, -1dB increments)
p2 is the attenuation index for f2, 0~31(0=3dB, -1dB incre ments)
p3 is the attenuation index for f3, 0~31(0=3dB, -1dB increments)
p4 is the attenuation index for f4, 0~31(0=3 dB, -1dB increments)

For example, the tone definitio n f or a tone of 450Hz with a cadence of 700 ms on and

1000 ms off is 1,0,700,1000,0,0,0,0,450,0,0,0,20,0,0,0

3.3.5 GSM Group Mode

The GSM Group mode enables multiple GoIP devices to simulate a multi-channel GSM
gateway.

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In this mode, only one GoIP acts as a Server and the others act as clients of the server
and reports its GSM number and status to the serv er. The number of clients is not
restricted. When the server receives a GSM call, it finds an i dl e client (not engaged in a
GSM call) and then forwar d t he call to this client. This enables a scalable multi-channel
VoIP GSM gatew ay. A t ypical application is to implement a call center that is accessed
via a single phone number (GSM).

When Client Mode is selected, the Server IP address and t he Client GSM number are
required to be filled in as show n below.

Note: Each GoIP still n eeds to register to V oI P server or proxy separately.

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3.3.6 GSM Caller ID Anonymous

When this parameter is enabl ed, the G SM Caller I D is not sent; the Caller I D shown at the
callee is anonymous.

3.3.8 Auto Reboot

When the Auto Reboot box is chec ked, the GoIP reboots itself automatically at the time
specified at the Reboot Time.

3.3.7 Remote S erver

When this parameter is enabl ed, the G SM Caller I D is not sent; the Caller I D shown at the
callee is anonymous.

3.4 Call Settings

Click on the “Call Settings” to configure the VoI P call settings. The first thing to set is the
Endpoint Type: H.323 or SI P.

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3.4.1 H.323 Phone

For H.323 protocol phone, 2 configuration modes are supp or t ed: Si ngle Configuration
and Configuration by Group.

3.4.1.1 Single Configuration

The Single Configuration supports only one V oI P number to a single H.323 Gatekeeper .

A. H.323 Phone Number

H.323 phone number: fill the login number (E164) here.

B. Gateway Prefix

If login with a Prefix method fill the pr efix number (do not fill the Phone num ber)..

C. Display Name

Display name is the name to be displayed on the called VoIP party.

D. H.323 ID

If the system requires an H.323 ID as a method of authenticatio n, ent er t he H.323 ID
provided.

E. Gatekeeper Address

This field assigns the IP addr ess or the domain name of the gatekeeper. The port
number can be added with the col on “ : ” symbol. For example: 192.168.1.70:8080.

F. Enable Auth

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If H.235 authentication is required, enabl e t his field and fill in the values as prov ided.

3.4.1.2 Configuration by Group

The “Config by G r oup” mode allows a user to setup the GoIP gateway to have 4
identities by registering to t he same gatekeeper with dif ferent phone numbers or to
different gatekeepers with different phone numbers or the same phone number, or a
combination of both. The GoIP gateway can be assigned to each group i ndivid ually. This
allows the V oI P channel to be shared by different group.

3.4.1.4 Advance Settings

Click “Adv ance Sett i ngs” to access additional H.323 parameters as shown below.

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A） RAS Port

RAS Port is an unreliable channel which is used to convey the regist ration,
admissions, bandwidt h change, and status messag es bet w een t wo H.323 entities.

B

） Q.931 Port （ Call Signaling Port ）

Call Signaling Por t is a reliable channel which is used t o convey the call setup and
release messages between two H.323 endpoints.

C

） H.245 Port （ Media Control Ports ）

Media control port is the port or port range used by the H.245 media control protocol

.

D） Fast Start

Enable or disable the Fast S t art in H.225.0. Most H.323 terminals or gateways support
the Fast Start feature.

E

） Register Mode

Register Mu ltiple Numbers: The GoIP gateway sends registration request in one

signaling packet to the gatekeeper. In the mode, one
signaling packet inclu des two VoIP line’s registration
information.

Register Multip le Times: In this mode, the GoIP gateway will register like two

terminals.

F

） DTMF Signaling

1

） DTMF TYPE

DTMF signals can be sent over to the called party once a ca ll is establish ed. GoIP
gateway supports bot h Inband and Outband DTMF signal types.

For Inband DTMF type, DTMF signals are generat ed locally at the calling phone and
then send to the called pa r t y as part of the voice signals. Th is method is not reliable
since the quality of the DTMF signals is subject to the codec used and t he quality of
the network traffics.
For Outband DTMF type, DTMF signal co mmands a re sent t o the called p arty and the
actual DTMF signals are a ct ually generated by the called par t y. This method allows
more reliable DTMF signalin g. However, it r equires the called party to support this
feature in order for this to work properly. GoIP gateway supports RFC2833 Outband
DTMF protocols.

2

）DTMF Payload Type

DTMF Payload Type is by RFC2833 pr ot ocol to carry the tone definitions for various
applications. The de fa ult DTMF payload type is 96. Pl ease consult your VoIP service

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provider for the proper sett ing if required.
.

3.4.1.5 H.323 Direct Mode

The Direct Mode allows peer-to-peer calls without registering to a gatekeeper.

3.4.2 SIP Phone

Set the “Endpoint Type” to SIP phone for connections to SIP servers.
GoIP gateway’s SIP configure page as follow:

A）Phone Number

Enter a SIP phone number.

B

）SIP Proxy

Enter the SIP proxy IP address or domain name. If the registration port isn’t 5060,
then add “:” and the port number. An example is sip.yourdomain.com:8080.

C

）SIP Registrar Server

If the Registrar Server is different from the SIP Proxy, enter its IP address or domain
name in this field. If the registration port isn’t 5060, t hen add “ :” and th e port num ber.
An example is sip.yourdomain.com:8080.

D

）Home Domain

SIP networks sometim es use the Home Domain name a s an i dentifier. Enter this field

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as required.

E

）Authentication I D

Enter the authentication ID as pr ovided.

F

） Password

Enter the authentication password as provided.

G

） Display Name

Enter this field for the name t o be displayed on the called VoIP party.

H

） Backup Server

The GoIP gateway supports one backup server in case of a main server failure. Once
registration to the main ser ver fails, the GoIP gateway will try to register to the backup
server.

I

） Outbound Proxy

OutBound proxies are devices that will forward SIP signal ing ( and frequently
RTP media traffic too). OutBound proxies are used for a number of reasons,
including, firewall traversal – both in parallel with a fi r ewall and situated in the
Internet as a Session Border Cont r oller, and also for hiding customer IP
addresses – calls are all routed through one point so that a public ITSP
address can be used for accessing the customers, rather than t he customer’s
own IP address.

If required, enter this field with the outbound proxy IP address or domain name
as provided.

3.4.2.1 Advanced Settings

Click on “Advance Settings” tab on the top right cor ner of th e Cal l Set ting p ag e to dis play
all the parameters available, as shown below, for programming. These parameter s al low
more advanced control over the SIP signaling and media preference.

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A

） Signaling Port (SIP Local port)

The default SIP port is 5060. Change this as required.

B

） NAT Keep-alive

The NA T Keep-alive feature sends a null packet to the SIP proxy periodically in order
to keep the NAT open for incoming data traffics.

C

） Advanced Timing Settings

Some SIP proxies may have special timing requirem ents. Change these para meter s
as required.

D

） Signaling Qos

Signaling QoS improves t he performance of SIP signaling. If local network device
supports Qos, select this field accordingly. Please consult your net w or k administrator

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for further information.

E

） DTMF Signaling

1

） DTMF TYPE

DTMF signals can be sent over to the called party once a call is established. GoIP
gateway supports bot h Inband and Outband DTMF signal types.

For Inband DTMF type, DTMF signals ar e generated locally at the calling phone and
then send to the called pa r t y as part of the voice signals. Th is method is not reliable
since the quality of the DTMF signals is subject to the codec used and the quality of
the network traffics.
For Outband DTMF type, DTMF si gnal commands are sent to the call ed par ty and the
actual DTMF signals are a ct ually generated by the called par t y. This method allows
more reliable DTMF signalin g. However, it requires the call ed party to support this
feature in order for this to work properly. GoIP gateway supports both RFC2833 and
SIP INFO Outband DTMF protocols.

2

）DTMF Payload Type

DTMF Payload Type is by RFC2 833 pr ot ocol to carry the tone definitions for various
applications. The default DTMF pay load t ype is 96. Please consult your VoIP serv ice
provider for the proper sett ing if required.

3.4.3 Media Setting

Click on “Media Settings” in the “Call Set tin g” menu to access the parameters available
for media settings.

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A

） RTP Port Range

This parameter speci fies t he r ange of the RTP (Real Time Protoco l) ports used by the
GoIP gateway. If your netw or k li mits the usable port range, this parameter may need
to be modified. Please consult your netw or k administrator for more information.

B

） Packet Length

This parameter defines t he voice packet length. The default setting is 20ms. The
range is from 5ms to 40ms at an increme nt of 5ms. Please n ote that so me codes hav e
a minimum packet length of more than 5 ms.

C

） Jitter Buffer Mode

Since data pac kets may arrives at different orders, the jitter buffer is used to hold the
data packet s r eceived for re-arrangement accor di ng to the packet sequence num ber.
Three jitter buffer modes are suppor t ed: Adaptiv e, Sequential, and Fixed. The
default is set to Adaptive mode with a mini m um jitter of 60 ms and a maximum jitt er of
120ms. Please consult your network administrator for more infor mati on on t he
network environment in order to determine the optimal settings.

D

） Media QoS

Similar to the Signaling QoS, the Media Qos is intended to impr ove the voice
performance or quality if QoS is supported by your local netw ork.

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3.4.4 Codec Preference

Codec Preference allows a user to select t he codes to be used and its priorit y to be

selected for a voice call.

Click on the check box to enable a codec. Select a codec and then press the UP or
DOWN button to move the position of the codec on the codec list wit h a pr iority in
descending order.

3.4.5 NAT Traversal

3.4.5.1 Signaling NAT Traversal

Signaling NAT traversal may be required if the GoIP gateway is put behind a NAT (or
multiple NATs). Depending on your network environment and SIP serv er cap abilit ies,
this feature may or may not be t ur n on.

A

） None

Select None to turn off this featur e.

B

） STUN (RFC 3489)

STUN (Simple Trav er sal of U DP (User Datagram Protocol) through NATs
(Network Address Translators)) is a net wor k proto col a llowing a c lient behin d a NAT

(or multiple NATs) to find out its public address, the type of NAT it is behind and the
internet-side port associated by the NAT with a parti cu lar local port.
Select STUN (RFC 3489) to use a STUN serv er for Signa ling NAT T rav ersal. Enter the
IP address or the domain name of the STUN server to be used.

C

） Relay Proxy

Relay proxy is a proprietary NAT traversal technology. Please consult your service

provider for more information.

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3.4.5.2 Media NAT Traversal

Similar to Signaling NAT Traversal, this featur e allows media packets (RTP) to be
routed properly in various net w ork environments.

A

） None

Select None to disable this feature.

B

） STUN （RFC 3489 ）

STUN (Simple Trav er sal of U DP (User Datagram Protocol) through NATs
(Network Address Translators)) is a net wor k proto col a llowing a c lient behin d a NAT

(or multiple NATs) to find out its public address, the type of NAT it is behind and the
internet-side port associated by the NAT with a parti cu lar local port.
Select STUN(RFC 3489) to use a STUN server for Signaling NAT Traversal. Enter t he
IP address or the domain name of the STUN server to be used.

C

）Port forwardin g Support

Port forwarding (sometimes referred to as tunn el ing) is the act of forwarding a

network port from one netw or k n ode t o another. This technique allows an e xternal
user to reach a port on a private IP address (inside a LAN) from the outside via a
NAT-enabled router.
In order for this feature to work, the local network gateway must support this feature
and be set up properly. Please consult your netw ork administrator for help to enable
this Port forwarding feature.

D

） Relay Proxy

Relay proxy is a proprietary NA T traversal technology. Please consult your service

provider for more information.
Currently, the following 3 kinds of packaging mechanism are s upported:

 Mode 1: The media uses UDP packets and (or) e ncr ypt with multiple UDP

port;

 Mode 2: The media uses UD P packets and (or) encrypt with single UDP port;
 Mode 3: The media uses TCP packets and (or) en cr ypt (UDP over TCP).

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3.5 Call Divert

The Call Divert feature controls the routing of calls betw een VoIP and GSM.

Call Forward (From VoIP to PSTN)
Forward Number

Enter this field to forward all inco ming VoIP calls to t his nu mber (PSTN or Mobile). Using
“,” to add a 500ms delay to the dia ling sequence. If this field is blank, calls will not be
forwarded. The GoIP gateway answers an incoming VoIP call and generates a dial tone.
The caller can then dial the number (PSTN or M obile) desired.

Forward Passwo rd

This field sets the password protection for using the GSM connection. If a password is
entered, the GoIP gateway will generate an indication tone an d wait for the call to dial the

Dial Plan to PSTN

This field sets the password protection for using the GSM connection. If a password is
entered, the GoIP gateway w ill gener at e an indication tone and wait for the call to dial the

Forward all incoming calls from the GSM connection to the VoIP number specified in this
field. Forward Password is not required once this field is set. If this field is blank, the
GoIP answers an incoming GSM calls and then generates the VoI P dial tone. Please
see below if the Forward Passw or d is s et. The caller can then dial a V oI P number

Call Forward (From GSM to VoIP)
Forward Number

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manually. At the end, a pound (#) can be dialed to activ at e t he dialing of the VoI P
number immediately. If not, the VoIP number i s di aled after a preset timeout.

Forward Passwo rd

This field sets the p asswo rd pr otectio n for incoming GSM calls. If a password is entered,
the GoIP gateway will generate an indication tone after answering an incoming call. The
caller is then ready to dial the password. Once the password is cor rectly entered, the
GoIP gateway generates a VoIP dial tone and waits for the caller to dial a VoIP number.

Dial Plan to VoIP

This field sets the password protection for using the GSM connection. If a password is
entered, the GoIP gateway w ill gener at e an indication tone and wait for the call to dial the

3.5.1 Dial Plan

In Call Divert mode, dial plan is supported in order to pre-program the various call routes
based on the number dialed. Dial plan can be defined individually for VoIP to
PSTN/GSM calls and GS M / PSTN to VoIP calls.

3.5.1.1 Basic Syntax

1. Multiple rules are supp o rt ed; the “ |” charact er is us ed as a se p arator bet wee n tw o rules.
For example: "00:-00|0:-0+86|:+86755"

2. The rules are examined and executed from left to right. Whenever a match is found,
the rule examination terminat es and the rule matched is executed i m med iat ely.

3. T he dial plan syntax is “A:-a+b" where A, a, and b are single or multiple digits, for
example “0:-0+86". The “A" before the colon (“:”) is the matching condition and the
“-a+b” after the “:” is the action to be executed. If the condition “A” is matched, the
“a” portion of the number dialed (match ed only from the first digit) is taken out and the
“b” portion is added to the beginning of the number dialed. If a match is not found,
the dial plan matching terminates and no action is performed on the number dialed.
If “A” is not present, the rule is executed immediately after the number is dialed All
subsequent rules, if define d, ar e ignored.

Note: “a” .must be a subset of “A” in order for this t o work.

4．Range definition is supported. Use “[A-B]” to specify the digit range de sired. For

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example, A = [2-8] means that the number dialed with the leading digit being in the
range from 2 to 8 is a match and the cor r esponding action is executed.

Examples:

1. Dial Plan: "0:|:+0755".
a. Input: "02083185711" -> Output: “02083185711";
b. Input "83185700" -> Output: "075583185700".

2. Dial Plan："00:-00|0"-0+86|:+86755".
a. Input: "008522343318" -> Output: "8522343318";
b. Input: "02083185711" -> Output: "862083185711";
c. Input: "83185700" -> Output: "8675583185700".

3. Dial Plan："00:|0:-0+0086|:+0086755".
a. Input: "008522343318" -> Output: "008522343318";
b. Input: "02083185711" -> Output: "00862083185711";
c. Input: "83185700" -> Output: "008675583185700".

4. Dial Plan："0:|1[3-9]:+0|[2-8]:+0755|:+0755".
a. Input: "076322343318" -> Output: "076322343318";
b. Input "13044557766" -> Output: "013044557766";
or Input: "13644557766" -> Output: "013644557766"
c. Input: "23185700" -> Output: "075523185700".
or Input: "73185700",-> Output: "075573185700"

3.5.1.2 Advanced Syntax for Limiting Number Length

To limited the lengt h of t he number input, the syntax is in the fo ll ow ing format:

"DDXXXXXX:-a+b"

The "DDXXXXXX" before the “:” is the matching condition for a number with a length of 8
digits. The first two digits, “DD” are predefined and the rest can be any digits. Please
note that only the first 8 digits of the number entered is used and any extra digits are
ignored.

For examples:

1. Dial Plan: "00:|0:-0+0086|[1-8]xxxxxxx:+0086755".

Input Number: 21234567
Output Number: 008675521234567

2. Dial Plan: "0:|13[0-9]xxxxxxxx:+0|[1-8]xxxxxxx:+0755"

Input Number: 138123456 78
Output Number: 013812345678
Note: A “0” is added to the beginning of all China Cellphone number (11-digit)
starting with the leadi ng t w o digits as 13

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Input Number: 83185922
Output Number: 075583185922
Note: “0755” is added to the beginning of all loc al n um ber ( 8-digit)

3.6 SMS Mode

GoIP supports SMS Dial mode and SMS Relay mode. The Dial mode uses SMS for
Call Back service and the Relay mode bridges the SMS messages between GSM and
VoIP. Please note that Dial mode only supports incoming SMS message s and out going
SMS from VoIP to GSM is disabled.

3.6.1 SMS Dial Mode in SIP

SMS Dial mode in SIP is commonly used for Call Back Service to save on GSM / Long Distance
call charges. The caller initiates a call by sending a SMS message to the service provider via the
GoIP. Once the GoIP receives the SMS, it will initiates a call to the SIP Server (Service Provider)
based on the SMS contents. This is not an actual voice call that will be answered. Once the
Service P rovider receives the call information via the SIP call, it then connects the GSM Caller
and the Callee (the number specified in the SMS content) together. Please note that this feature
is intended for a Service Provider to integrate the GoIP in their system for Call Back Service.

Select the SMS Dial Mode and then select the correct mode for SMS message formats as
described below:

1. Mode 1

GoIP uses the Caller ID of the GSM/SMS caller to call to the number specified in the
SMS message content.

For example:

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GSM Caller ([phone] sends a SMS to the GoIP with the message
“8675588228822”. The GoIP uses the GSM Caller ID (8613800000000) as its
SIP ID to call the number 8675588228822 and the SIP Sever is located at

192.168.2.1 using signaling port 5060. The INVITE message is:

INVITE sip:8675588228822@192.168.2.1:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK36396 9813
From: <sip:8613800000000@192.168.2.1:5060>;user=phone;tag=65248630
To: <sip:8675588228822@192.168.2.1>
Call-ID: 117025903@192.168.2.237
CSeq: 2 INVITE
Contact: <sip: 8613800000000@192.168.2.237:5060>
Max-Forwards: 30
User-Agent: DBL
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 226

2. Mode 2

GoIP uses its SIP number as the caller ID to call the number specified in the message
content.

For example:

GSM Caller ([phone] sends a SMS to the GoIP with the message
“8675588228822”. The GoIP has a SIP number 20001 registered to the SIP
Sever located at 192.168.2.1 (signaling port 5060). In mode 2, the GoIP uses its
own SIP ID to call the number 8675588228822. The INVITE message is:

INVITE sip:8675588228822@192.168.2.1:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:50 60;branch=z9hG4bK36396 9813
From: <sip:20001@192.168.2.1:5060>;user=phone;tag=65248630
To: <sip:8675588228822@192.168.2.1>
Call-ID: 117025903@192.168.2.237
CSeq: 2 INVITE
Contact: <sip:20001@192.168.2.237:5060>
Max-Forwards: 30
User-Agent: DBL
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 226

3. Mode 3

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GoIP uses the GSM Caller ID and the SMS message content to form a new SIP number.
It will then call this number. This SIP number is in the format shown below.

SIP number = SMS Message content + “*” + GSM Caller ID

For example:

GSM Caller ([phone] sends a SMS to the GoIP with the message
“8675588228822”. The GoIP has a SIP number 20001 registered to the SIP
Sever located at 192.168.2.1 (signaling port 5060). In mode 2, the GoIP uses its
own SIP ID to call the SIP number 8675588228822*8613800000000. The
INVITE message is:

INVITE sip:8675588228822*8613800000000@192.168.2.1:5060;transport=udp
SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK36396 9813
From: <sip:20001@192.168.2.1:5060>;user=phone;tag=65248630
To: <sip:8675588228822*8613902994477@192.168.2.1>
Call-ID: 117025903@192.168.2.237
CSeq: 2 INVITE
Contact: <sip:20001@192.168.2.237:5060>
Max-Forwards: 30
User-Agent:DBL
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 226

SMS Dial Mode Prefix is used to adds a prefix to the SIP number to be called in the SMS

Dial Mode.

For example:

Setting the SMS Dial Prefix to 999 in Mode 1 changes the INVITE message to:

INVITE sip:9998675588228822@192.168.2.1:5060;tr ansport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK36396 9813

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From: <sip:8613800000000@192.168.2.1:5060>;user=phone;tag=65248630
To: <sip:9998675588228822@192.168.2.1>
Call-ID: 117025903@192.168.2.237
CSeq: 2 INVITE
Contact: <sip: 8613800000000@192.168.2.237:5060>
Max-Forwards: 30
User-Agent: DBL
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 226

3.6.2 SMS Dial Mode in H,323

Similarly, SMD Dial Mode also works in H.323 Protocol in order to support Call Back Service.
The user still sends a SMS message to the GoIP to initiate the call back. The H.323 protocol for
initiating a call after receiving a Call Back SMS message Call Back has 3 different formats as
described below.

Select the SMS Dial Mode and then select the correct mode for SMS message formats as
described below:

1. Mode 1 (Currently not supported)

GoIP uses the Caller ID of the GSM/SMS caller to call to the number specified in the
SMS message content.

2. Mode 2 (Currently not supported)

GoIP uses its SIP number as the caller ID to call the number specified in the message
content.

For example:

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GSM Caller ([phone] sends a SMS to the GoIP with the message

“8675588228822”. The GoIP uses its own H.323 ID to call the number

8675588228822. The H.323 protocol command involved is:

Send RAS Message: admissionReq uest
admissionRequest {
requestSeqNum = 241
callType = pointToPoint NULL
endpointIdenti f i er = "3705_endp"
destinationInfo = 1 el em ents {
[0] = dialedDigits "8675588228822"
}
srcInfo = 2 elements {
[0] = dialedDigits "20001"
[1] = h323-ID "20001"
}
srcCallSignalA ddress = ipAddress {
ip = 4 octets {

c0 a8 02 ed ....

}
port = 2049
}
bandWid th = 2048
callReferenceValue = 7502
conferenceID = 16 octets {
7f f3 78 77 49 3f 4c c1 9a dc 6a 84 12 d8 30 8f ..xwI?L...j...0.
}
activeMC = FALSE
answerCall = FALSE
canMapAlias = FALSE
callIdentifier = {

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guid = 16 octets {
cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55

5b .@....`.k_....U[

}
}
gatekeeperIdentifier = "GnuGk"
willSupplyUUIEs = FALSE
}

3. Mode 3 (Currently not supported)

GoIP uses the GSM Caller ID and the SMS message content to form a new SIP number.
It will then call this number. This SIP number is in the format shown below.

SIP number = SMS Message content + “*” + GSM Caller ID

SMS Dial Mode Prefix is used to adds a prefix to the SIP number to be called in the SMS

Dial Mode.

For example:

Setting the SMS Dial Prefix to 999 in Mode 1 changes the INVITE message to:

Send RAS Message: admissionReq uest
admissionRequest {
requestSeqNum = 241
callType = pointToPoint NULL
endpointIdentifier = "3705_endp"
destinationInfo = 1 el em ents {
[0] = dialedDigits "9998675588228822"
}
srcInfo = 2 elements {
[0] = dialedDigits "20001"
[1] = h323-ID "20001"
}
srcCallSignalA ddress = ipAddress {
ip = 4 octets {

c0 a8 02 ed ....

}
port = 2049

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}
bandWid th = 2048
callReferenceValue = 7502
conferenceID = 16 octets {
7f f3 78 77 49 3f 4c c1 9a dc 6a 84 12 d8 30 8f ..xwI?L...j...0.
}
activeMC = FALSE
answerCall = FALSE
canMapAlias = FALSE
callIdentifier = {
guid = 16 octets {
cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55

5b .@....`.k_....U[

}
}
gatekeeperIdentifier = "GnuGk"
willSupplyUUIEs = FALSE
}

3.6.3 SMS Relay Mode (FOR SIP ONLY)

The SMS Relay mode bridges SMS messages between GSM and VoIP. This means that SMS
messages can send from GSM phone to SIP phone and SIP phone to GSM.phone. This feature is
only available for SIP protocol.

Select the Relay mode as shown above and enter the SMS Forward Number. Please make sure
that this extension phone must support SMS since all GSM SMS received will be forwarded to
this phone. Please also note that the SIP server must support this SMS feature as well.

1. Relay a GSM SMS to a SIP Phone

Here is an example of relaying a received GSM SMS to a SIP Phone:

a) GSM SMS is sent from 8613682626865.
b) GSM SMS content is 075583185700.
c) SMS Forward Number is set to 3999
d) GoIP creates a new message containing the GSM Caller ID and the GSM SMS

received

8613682626865
075583185700

e) The SI P Message sent from GoIP to the SIP Server is:

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MESSAGE sip:3999@192.168.2.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.162:5060;branch=z9hG4bK19676 85528
From: <sip:20001@192.168.2.1>;tag=667435795
To: <sip:3999@192.168. 2. 1>
Call-ID: 2094144847@192.168.2. 162
CSeq: 4 MESSAGE
Contact: <sip:20001@192.168.2.162:5060>
Max-Forwards: 30
User-Agent: DBL
Content-Type: text/plain
Content-Length: 28

8613682626865
075583185700

2. Relay a SIP SMS to a GSM Phone

Here is an example of the SIP SMS sent to the GoIP

a) SIP number or extension of send ing S M S i s 3999
b) GoI P SIP number is 20001
c) Designated GSM number for the SMS is 13682626800
d) SMS M essage content entered to 3999 is

13682626800
Hello world

e) SMS M essage content to GSM phone is Hello world
f) The SIP Message sent from GoIP to the SI P Server is:

MESSAGE sip:20001@192.168.2.162:5060 SIP/2.0
From: <sip:3999@192.168.2.89>;tag=5 031
To: <sip:20001@192.168.2.1>
Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008@192.168.2.89
CSeq: 3 MESSAGE
Contact: <sip:3999@192.168.2.89>
max-forwards: 16
date: Tue, 18 Nov 2008 06:36:37 GMT
user-agent: SIPPER for 3 CX Pho ne
p-hint: usrloc applied
Content-Type: text/plain
Content-Length: 26

13682626800
Hello world

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3.7 Relay Incoming Caller ID (GSM to VoIP Call)

For SIP mode, the GoIP al lows the incoming caller ID from a GSM Call to be transferred to
a VoIP terminal. This parameter is called CI D For war d Mode an d it can be accessed
under the Call Divert Page as shown below.

Three selections are avail abl e:

1. Disable – This mode disa bles the inco ming caller I D from a GSM call to be for warded

to a VoIP terminal.

2. Use Remote Party ID – T his set s the SIP method u sed to re lay the inco ming caller ID

to a VoIP terminal. The method is called Remote Par t y ID. For example, if the
incoming caller ID is 13800000000, the content of the SIP INVITE message is show
below . Please note the section is marked in red.

3. Use CID as SIP Caller ID – This sets the SI P method used to relay the inco m ing

caller ID to a V oI P terminal. The method is similar to t he Re m ot e Par t y ID except
that the SIP ID is replaced with t he incom ing cal ler ID.. For example, if the incomin g
caller ID is 13800000000, the co nt ent of the SIP INVITE message is show below.
Please note the section is marked i n r ed.

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For H.323 mode, this feature is currently not supported and it will be supported in future
firmware releases.

3.8 Gain Settings…

A hidden w ebpage is provided to set the r eceiving and transmit gains of VoIP Channel.
The URL lin k is:

http://xxx.xxx.xxx.xxx/default/en_US/gain.html

THIS PAGE I S INTENDED FOR AN EXPERIENCED USER OR A N ADMINIST RATOR
ONLY. PLEASE SET THE GAINS WITH CAUTIONS.

Note: A t oo low or t oo high input gain MAY affect the sensitiv ity of DTMF detections

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3.9 Network Configuration

Click on “Network” tab in the left menu colu m n t o configure the LAN and PC ports.

3.9.1 LAN Port

Three LAN Port modes are supported: DHCP, Static I P, PPPoE.

1

） DHCP

Choose DHCP if a local DHCP host i s available. This allows the GoIP gateway to
obtain network information (IP Address, Subnet Mask, Default Route, Primary DNS,
Secondary DNS, and other DHCP options) from the DHC P host.

2

） Static IP

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Choose Static IP if your net w or k topology requires. Please fill in the IP Address,
Subnet Mask, Default Route, Primary DNS, and Secondary DNS (optional) as

provided by your network ad minist r at or.

3

） PPPoE

PPPoE is a common di al up m ethod for you network modem (C able / xDSLs).
Choose this if your network environment requires. Enter the User Name and
Password as provided by your I SP.

4

）802.1q VLAN

This QoS feature requires your QoS support of your network to improve voice data
traffics. Please consult your network administrator for proper settings.

5

） Advanced…

The Advanced settings al low t he user t o set the broadcast address and to clone a
MAC address instead of using the factory preset MAC address. Please consult your
network administrator for f ur t her in formation.

3.9.2 PC port configurations

The PC Port allows additio n networ k devices to be at tach ed behind th e GoIP Gateway. It
offers both Bridg e a nd Static IP modes to meet your net wor k topo logy. It is factory preset
to the Static IP mode with the IP address 192.168.8.1.

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1） Bridge Mode

Select Bridge mode if your network topology requires the net w ork devices (PC or
others) to be in the same network segment as the GoIP gateway. In this case, the
GoIP gateway functions as an Et her net switch.

2

） Static IP Mode (Default Setting)

Select Static IP mode for a new network segment for the network devices behind the
GoIP gateway. In this case, the GoIP gateway functions as an Ethernet router. Fill
in the IP Address field with a new segment address that is different from that for t he
LAN port. Please select the Subnet Mask a ccor dingly. A commonly used value is

255.255.255.0.

Enable the DHCP Server if you want the GoIP gateway functions as a local DHCP
host for the PC segment. This will enables the GoIP gateway to assign IP addresses
to network devices that are at tached to the PC port segment.

Specify t he S t arting A ddress . Endin g Addre ss, and Static DNS accordingly.

4

） Advanced…

The Advanced settings al low t he user t o set the broadcast address and to clone a
MAC address instead of u sing the factory preset MAC address. Please consult your
network administrator for f ur t her in formation.

3.10 Save Configuration

To confirm and com mit all changes made, cl ick on the Save Changes tab. Otherwise, all
changes will be discarde d. Once all changes are saved, the follow ing screen message is
displayed.

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3.11 Discard Changes

To discard all chan ges made, click on the Discard Changes tab.

3.12 Tools Menu

Select the Tools to access the following funct io ns: Online Upgrade, Change Password,
Reset Config, and Reboot.

3.12.1 Online Upgrade

To perform a firmware upg rade, select the O nli ne Upgra de tab to access the pag e below.

Enter the update link as prov ided by your service provider. A sample link is:

http://202.155.200.154/update/A34HS-3.07-18.pkg

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Click the Start button to start the firmware upgrade.

WARNING: POWER SHUTDOWN / FAILURE DURING FIRMWARE U PGRADE MAY

PERMINENTLY DAMAGE THE GOIP GATEWAY.

3.12.2 Change Password

Click on the Change Password tab to access the page below.

A

） User Password

This is the password for t he user name/ID “user”. The default password is “1234”.
This user name is limited to access the Network Configurati on menu.

B

）Administrator Password (default: admin)

This is the password for the user name/ID “admin”. The default password is
“admin”. This user name allows full access to all con figur ation settings available.

3.12.3 Reset Configuration

Click on the Reset Config tab to reset the GoIP gateway to its factory default sett ings.

3.12.4 Reboot the Device

Click on the Reboot tab to reboot the GoIP gateway. The web page is then not
accessible until the device com pletes the reboot process.

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4 Hardware Specifications

Item Description Remark

CPU ARM9E 133MHz
DSP VPDSP101 95MHz
RAM 8M
FLASH 4M
Power Supply DC4.5V/2000mA +-10% Input AC100V to AC240V

GSM Module

Huawei GSM Module - GSM

850 MHz/GSM 900 MHz

Simcom GSM Module: GSM

850 MHZ, 900MHz, DCS 1800
MHz, PCS 1900 MHz

Default: 900 / 1800 MHz

Power Consumption

The Maximum 3 W

LED

RUN, GSM, LAN, PC
Ethernet Port Two RJ-45 Jacks 100/10BASE-T
Weight 105 Grams Without AC/DC Adapter

Operating Temperature
Range

0－40℃
Operating Humudilty 40％－90％ Not Congealed

Color Blue
VoIP Channel 1
GSM Channel 1

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5 Useful Factory Default Settings

Parameter Defaults

Ethernet

LAN DHCP Client mode

PC Fixed IP：192.168.8.1

Login ID /
Password

Admin mode: admin/admin

User mode User/1234

Time Zone GMT＋8