SPOTEL GSM1SIP User manual

Spotel™ By Nicherons Canada
User Manual
GSM1SIP
One Channel GSM VoIP Gateway

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One Chunnel GSM VoIP Gateway
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Contents
1 Product Introduction ................................................... 3
1.1 General Information ............................................................................................................ 3
1.2 Protocol ............................................................................................................................... 3
1.3 Hardware Specification ....................................................................................................... 4
1.4 Software Specification ........................................................................................................ 4
1.5 List of the Package .............................................................................................................. 4
1.6 Appearance .......................................................................................................................... 5
2 Installation .................................................................... 6
2.1 Installation Steps ................................................................................................................. 6
2.3 LED Indicators .................................................................................................................... 7
2.4 SMS Commands.................................................................................................................. 7
3 Configuration Guide .................................................... 7
3.1 Web Configuration Menu .................................................................................................... 8
3.2 Status ................................................................................................................................... 9
3.2.1 Phone Information ................................................................................................ 9
3.2.2 Network Information ........................................................................................... 10
3.2.3 GSM Module Information .................................................................................. 10
3.3 Configurations ................................................................................................................... 10
3.3.1 Language ............................................................................................................. 11
3.3.2 Time Zone and Time Server.............................................................................. 12
3.3.3 DTMF Min Detect Time Gap ............................................................................. 12
3.3.4 Auto-Provision ..................................................................................................... 12
3.3.5 Network Tone....................................................................................................... 13
3.3.6 GSM Group Mode .............................................................................................. 14
3.3.7 Auto Reboot ......................................................................................................... 15
3.4 Call Settings ...................................................................................................................... 15
3.4.1 H.323 Phone........................................................................................................ 15
3.4.1.1 Direct Mode................................................................................................. 15
3.4.1.2 Gatekeeper Mode ........................................................................................ 16
3.4.1.3 Advance Settings ......................................................................................... 17
3.4.2 SIP Phone ............................................................................................................ 18
3.4.2.1 Advanced Settings ....................................................................................... 20
3.4.3 Media Setting ...................................................................................................... 22
3.4.4 Codec Preference............................................................................................... 24
3.4.5 NAT Traversal...................................................................................................... 24
3.4.5.1 Signaling NAT Traversal ............................................................................. 24
3.4.5.2 Media NAT Traversal .................................................................................. 25
3.5 Call Divert ......................................................................................................................... 26
3.5.1 Call Forward (From VoIP To PSTN)................................................................. 26

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3.5.2 Auto Forward Call To PSTN .............................................................................. 27
3.5.3 Call Forward (From PSTN To VoIP)................................................................. 27
3.6 SMS Disposal .................................................................................................................... 28
3.6.1 SMS Call Out....................................................................................................... 28
3.6.2 SMS Relay ........................................................................................................... 28
3.6.2.1 SMS Relay To VoIP System ........................................................................ 28
3.6.2.2 SMS Relay To GSM Network ..................................................................... 28
3.7 PSTN Caller ID Transparent ............................................................................................. 28
3.8 Dail Plan ............................................................................................................................ 28
3.9 Gain Settings… ................................................................................................................. 36
3.10 Network Configuration.................................................................................................... 37
3.10.1 LAN Port............................................................................................................. 37
3.10.2 PC Port Configurations.................................................................................... 38
3.11 Save Configuration.......................................................................................................... 40
3.12 Discard Changes.............................................................................................................. 40
3.13 Tools Menu...................................................................................................................... 40
3.13.1 Online Upgrade................................................................................................. 41
3.13.2 Change Password ............................................................................................ 41
3.13.3 Reset Configuration ......................................................................................... 42
3.13.4 Reboot the Device............................................................................................ 42
4 Hardware Specifications ........................................... 42
5 Manufactory Parameters ........................................... 42

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1 Product Introduction
1.1 General Information
A VoIP GSM Gateway enables direct routing between IP and GSM network without the
use of a FXO port or the PSTN network. With this device, the usage of VoIP is greatly
enhanced with significant savings on long distance and roaming charges.
1.2 Protocol
TCP/IP V4 (IP V6 auto adapt)
ITU-T H.323 V4 Standard
H.2250 V4 Standard
H.245 V7 Standard
H.235 Standard（MD5，HMAC-SHA1）
ITU-T G.711 alaw/ulaw, G.729A, G.729AB, and G.723.1 Voice Codec
RFC1889 Real Time Data Transmission
Proprietary Firewall-Pass-Through Technology
SIP V2.0 Standard
Simple Traversal of UDP over NAT (STUN)

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Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client
Hyper Text Transfer Protocol (HTTP)
Dynamic Host Configuration Protocol (DHCP)
Domain Name System (DNS)
User account authentication using MD5
Out-band DTMF Relay: RFC 2833 and SIP Info
1.3 Hardware Specification
ARM9E 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
1.4 Software Specification
LINUX OS
Built-in HTTP Web Server
PPPoE Dial-up
NAT Broadband Router Functions
DHCP Client
DHCP Server
Firmware On-line upgrade
PSTN Caller ID transmit
Multiple Language Support
Supported call divert
Supported PSTN auto call out to PSTN
Supported Multi_devices Cooperate Mode(Group Mode)
Supported SMS call out
1.5 List of the Package
a) One GSM Gateway main unit
b) One DC4.5V/2000mA power adaptor
c) One Ethernet cable (3M)

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1.6 Appearance
1）LAN
Connect this port to an Ethernet Switch/Router, the Ethernet of a DSL modem, or
other network access equipment.
2）PC

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Connect a computer or other network device to this port.
3）POWER (DC4.5V/2000mA)
Connect the 4.5V/2000mA Adapter provided to this power jack.
4）Reset
Press this button to reset the Gateway to factory defaults.
2 Installation
2.1 Installation Steps
Please follow the connection diagram above to install the Gateway.
a) Insert a GSM SIM card in the SIM card compartment located at the bottom of the
Gateway’s.
b) Connect an Ethernet cable the LAN port of the Gateway and the other end to your
existing network equipment.
c) Connect an Ethernet cable to the PC Port of the Gateway and the other end to a PC
or other network device (Optional).
d) Connect the power adapter provided to the power jack of the Gateway.

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2.3 LED Indicators
The following table defines the status of the LEDS located on the top case and on the
RJ-45 connectors.
LED
DESCRIPTION
RUN
1. When the is booting，this LED will flash
100ms ON and 100ms OFF.
2. When the is login your softswitch, this
LED will flash 1s ON and 1s OFF.
GSM
When the ’s GSM login the ISP’s system,
this LED will flash 1s ON and 1s OFF.
2.4 SMS Commands
supported commands come from SMS.
FUNCTION
SMS CONTENT
REMARK
Obtain LAN Port Info
INFO
Not distinguish majuscule and lowercase
Reset device
RESET Password
Not distinguish majuscule and lowercase
Reboot device
REBOOT Password
Not distinguish majuscule and lowercase
Note：In command Reset and Reboot, the Password is the device’s admin password.
The command keywords can use majuscule and lowercase, but the password must
distinguish majuscule or lowercase.
3 Configuration Guide
To configure the Gateway, you must login to its Web server via the LAN or PC port.
The LAN port is factory preset to obtain an IP from the local DHCP host and the PC port is
set to the fixed IP 192.168.8.1.
If a local DHCP host is available, the LAN will obtain an IP address automatically. To listen
to the IP address assigned, just dial a call via the Gateway’s SIM card phone number.
When the call is connected, you will hear a dial tone. Just dial “*01#” for English voice
prompt on the LAN IP and “*00#” for Chinese voice prompt on the LAN IP. The LAN IP
Address can also be obtained by sending a SMS message to the GSM phone number.
The will then reply with a SMS message containing the LAN IP address.
If you want obtained LAN port IP by sending a SMS message, please send” INFO “or”

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info”.
If a local DHCP host is not available, you can then access the Gateway via the PC port.
You will need to change the PC LAN configuration via the Network Connections under the
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 address.
Once either the IP address of the LAN or PC port is known, you are now ready
to access the Web server of Gateway.
3.1 Web Configuration Menu
If your PC is connected to the Gateway via the LAN port network segment, you need to
type the LAN IP address of the Gateway in your Web Browser to access the Web server
of the 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 prompt you to enter the “User name” and
“Password: as shown below.

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Enter the User name and Password and the press OK to access the Gateway Web
Server. The default for both user name and password is “admin”.
3.2 Status
The Status page shown below is the default / home page of the Web server.
3.2.1 Phone Information
A. Serial Number
Each Gateway has a unique serial number assigned by the factory such as 08030031.
This number is important for centralized configuration, technical support, and warranty.

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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 such as GHS-3.01-5.
C. Hardware Mode
This field shows terminal’s hardware type.
D. Phone Status
This field shows the status of Line’s connection status. If the connection is successful,
this field displays LOGIN; otherwise, it displays LOGOUT.
3.2.2 Network 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 module type.
B. GSM Signal
This field displays the GSM signal status.
C. GSM Status
This field shows the status of GSM connection status. If the connection is successful,
this field displays LOGIN; otherwise, 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 Changes.

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Click on “Preference” in the left menu of the configuration web, and the screen will be
displayed as below:
3.3.1 Language
Currently supports English, Simplified Chinese and Traditional Chinese. Select the
language desired and the Web page will be shown in the language selected accordingly.
The language can also be selected at the top of the web page. Once selected, the
webpage language is refreshed immediately. However, the language selection is not
saved until the Save Changes icon is clicked.

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3.3.2 Time Zone and Time Server
The Gateway supports Network Time Protocol (NTP) to obtain the date and time
information from an NTP server (Time Server). The time zone is specified as in GMT ±
offset. For example, the Pacific Standard Time is GMT-8, and the Pacific Daylight Time
is GMT-7.
Note: The Gateway supports CDR and Billing Information, it is important to set up these
two parameters properly.
3.3.3 DTMF Min Detect Time Gap
This parameter use to limit two same DTMF digit’s minimum time gap, the range is 60ms
to 120ms, default is 80ms.
If you encounter double digit problem, gain it to more, if you encounter lose digit, then gain
it to less.
3.3.4 Auto-Provision
The Gateway supports Auto Provisioning which enables configuration parameters to 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 further information on this.

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3.3.5 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.
You can configure the Network Tones as Customized option:
Each tone listed above is defined in the following format:
nc, rpt, c1on, c1off, c2on, c2off, 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 milliseconds)
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 off (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=3dB, -1dB increments)
p2 is the attenuation index for f2, 0~31(0=3dB, -1dB increments)
p3 is the attenuation index for f3, 0~31(0=3dB, -1dB increments)
p4 is the attenuation index for f4, 0~31(0=3dB, -1dB increments)
For example, the tone definition for 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

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3.3.6 GSM Group Mode
supported multi_devices cooperate with one group; it can work like a multi_chunnels
GSM gateway. Any one are can work as Group Server Mode or Client Mode.
Server Mode:
When running in Server Mode;
It can auto forward the GSM’s incoming call to any free client devices.
At this moment, you only offer this ’s GSM number to your user.
Client Mode:
When running at Client Mode;
It will auto send itself GSM number and state to Server device and waiting the
incoming call forward from Server .
You must enter itself GSM number and Server device’s IP address into follow
option.

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Disable:
When run as alone mode, please set it to Disable.
3.3.7 Auto Reboot
This option enable auto reboot each day, will auto reboot itself at Reboot Time. If meet
a current called, the action will auto delay after the call finish.
3.4 Call Settings
Click on the “Call Settings” to configure the VoIP call settings. The first thing to set is the
Endpoint Type: H.323 or SIP.
3.4.1 H.323 Phone
For H.323 protocol, 2 Endpoint Modes are supported: Direct Mode and Gatekeeper
Mode.
3.4.1.1 Direct Mode
In Direct Mode, running at H.323 P to P type.

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A. H.323 Phone Number
H.323 phone number: fill the login number (E164) here.
B. Display Name
Display name is the name to be displayed on the called VoIP party.
C. H.323 ID
If the system requires an H.323 ID as a method of Authentication, enter the H.323 ID
provided.
D. Default Voice Gateway
This field assigns the IP address or the domain name of the gatekeeper or other VoIP
gateway. The port number can be added with the colon “:” symbol. For example:
192.168.1.70:8080.
will send out all VoIP calls to this address.
3.4.1.2 Gatekeeper Mode
The “Gatekeeper Mode” mode allows a user to setup the Gateway by registering to the
gatekeeper with one H.323 account.
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 prefix number (do not fill the Phone number).
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 Authentication, enter the H.323 ID
provided.
E. Gatekeeper Address
This field assigns the IP address or the domain name of the gatekeeper. The port
number can be added with the colon “:” symbol. For example: 192.168.1.70:8080.
F. Enable Auth

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If H.235 Authentication is required, enable this field and fill in the values as provided.
3.4.1.3 Advance Settings
Click “Advance Settings” to access additional H.323 parameters as shown below.
A）RAS Port
RAS Port is an unreliable channel which is used to convey the registration,
admissions, bandwidth change, and status messages between two H.323 entities.
B）Q.931 Port （Call Signaling Port ）
Call Signaling Port is a reliable channel which is used to 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.

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D）Fast Start
Enable or disable the Fast Start in H.225.0. Most H.323 terminals or Gateways
support the Fast Start feature.
E）Register Mode
Register Multiple Numbers: The Gateway sends registration request in one signaling
packet to the gatekeeper. In the mode, one signaling
packet includes two VoIP line’s registration information.
Register Multiple Times: In this mode, the Gateway will register like two terminals.
F）DTMF Signaling
1）DTMF TYPE
DTMF signals can be sent over to the called party once a call is established.
Gateway supports both Inband and Outband DTMF signal types.
For Inband DTMF type, DTMF signals are generated locally at the calling phone and
then send to the called party as part of the voice signals. This 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 signal commands are sent to the called party and the
actual DTMF signals are actually generated by the called party. This method allows
more reliable DTMF signaling. However, it requires the called party to support this
feature in order for this to work properly. Gateway supports RFC2833 Outband
DTMF protocols.
2）DTMF Payload Type
DTMF Payload Type is by RFC2833 protocol to carry the tone definitions for various
applications. The default DTMF Payload Type is 96. Please consult your VoIP service
provider for the proper setting if required.
.
3.4.2 SIP Phone
Set the “Endpoint Type” to SIP Phone for connections to SIP Servers.
Gateway’s SIP configure page as follow:

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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.nicherons.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, then add “:” and the port number.
An example is sip.nicherons.com:8080.
D）Home Domain
SIP Networks sometimes use the Home Domain name as an identifier. Enter this field
as required.
E）Authentication ID
Enter the Authentication ID as provided.
F）Password
Enter the authentication password as provided.
G）Display Name
Enter this field for the name to be displayed on the called VoIP party.
H）Backup Server