2.2.1 For Windows 2000/NT.......................................................................................................................11
2.3NETWORK CONFIGURATIONS IN YOUR GATEWAY .....................................................................................13
2.3.1 Static ip address .................................................................................................................................15
2.4MAKING A V OIPCALL................................................................................................................................19
2.4.1 Configure the gateway into the Peer-to-Peer mode....................................................................19
2.4.2 Configure the gateway into the Proxy mode.................................................................................22
3.7PHONE P ATTERN...........................................................................................................................................34
4.2COMMAND LIST ...........................................................................................................................................61
Setting the configurations of dialing, including the
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Steps in configuration
STEP 1
Start Up
STEP 2
How to
Setup and
basically
STEP 3
Advanced
STEP 4
Command
List
To check out the peripheral equipments and
understand the feature of this gateway. Please
read this step very carefully before starting the
configuring.
Connecting the gatew ay and computer to start
configuring by WEB GUI.
Setting the ip address for this gateway to make
sure that it could connect with the internet.
Peer-To -Peer, GK mode and how to set these
tables to make calls by this gateway easily.
The other configurations of make call will be
discussed in this step.
Advanced configurations and special functions of
this gateway. Using the WEB GUI to show how to
set this table and explain the meaning of these
tables.
To explain the meaning of the command in the
command line interface and example the usage of
the command.
To get more usages or configuration in this step
and study about the command line configuration.
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1. Start Up
1.1 Introduction
SIPFXSO 1S1O/2S2O is a one-port/two-port FXS+FXO gateway. It
supports an innovative intelligent call rouging function that transparently routes
calls to destination either through PSTN or Internet.
SIPFXSO 1S1O/2S2O provides voice over IP and FAX over IP services
for ITSP/ISP Internet Telephony Services Provider and Office/SOHO IP -PBX
application.
Application Architecture
PBX
PBXPBX
Extension port
Extension port
FXO
FXO
FXO
IP
IP
IP
Network
Network
Network
FXS
FXS
FXS
l FXO ports can connect with PSTN Line or Extension Line of PBX
FXS
Trunk side
Trunk side
PBX
PBXPBX
Extension
Extension
FXO
FXS
FXS
FXS
FXS
l FXS ports can connect with Phone Set or Trunk Line of PBX
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1.2 Features and specification
Features
- IETF RFC 3261
- Automatically Dial Path Selection (IP or PSTN)
- PSTN Line switch to telephone set when power is failure
- PPPoE support
- Behind NAT router or IP sharing device
- DNS server inquiry
- Provide Peer-to-Peer Mode (Non SIP Proxy needed) selection
- Provide In-band or RFC2833 DTMF generation/detection
- Provide call progress tone
Management Feature
- TELNET/Console port and Web Browser configuration
Certification
- UL, CE, FCC
FXS Features
- 2-wire loop start
- Programmable On-Hook voltage, Ring voltage/Cadence/Frequency, Loop
current
- Line polarity reversal generation
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FXO Features
- 2-wire loop start
- Support auto-attendant (Tone or voice greeting)
- PSTN polarity reversal detection
- Disconnect tone detection
- Asking ping function with the incoming calls from PSTN side
- Record and analyze the Tone from PSTN side
Environmental
- Operation temp:0°C to 40°C
- Humidity: 10% to 90% (Non-condensing)
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1.3 Accessories and equipment
u The voice gateway in 2 FXS and 2 FXO ports or 1 FXS port and 1 FXO
port models and two RJ-45 connector (WAN and LAN).
u The AC adapter.
u The CD of user manual.
u The connection cabl e in RS-232 interface.
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1.4 Appearance
Front panel: The LED lights provide related system messages of the gateway.
SIPFXSO 1S1O
SIPFXSO 2S2O
Power: Light on means Gateway is power on, and vice versa.
TEL: Light on means the line is in use (off-hook), and vice versa.
LINE: Light on means the line is in use (off-hook), and vice versa.
Status:
1. LED light on means Gateway has successfully registered to Proxy when it
is in the Proxy Mode.
2. LED flash means Gateway is not registered to the Proxy when it is in the
Proxy Mode.
3. Or when Gateway is in downloading mode, LED should be flash as well.
4. LED light off means Gateway is in Peer-to-Peer Mode.
Ready:
1. Light on and slow flash means Gateway is in normal mode.
2. Light on and fast flash means Gateway is in downloading mode.
WAN: Connected to Public Ethernet
1. Line- LED light on means Gateway is physically connected to the Ethernet
correctly.
2. ACT- LED light on and flash when Ethernet data is being transmitted /
received.
LAN: Switch to another device, such as PC
1. Line- LED light on means Gateway is physically connected to the Ethernet
correctly.
2. ACT- LED light on and flash when Ethernet data is being transmitted
received.
Back panel:
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SIPFXSO 1S1O
SIPFXSO 2S2O
1. Ethernet Port
LAN/WAN: 10/100 Base-T; RJ-45 socket, complied with ETHERNET
10/100base-T.
The pin-out is as following:
PIN 1, 2: Transmit
PIN 3, 6: Receive
2. COM:
RS232 console port (DB-9pin male connector)
Note: use straightforward cable to connect to your computer.
PINOUTS
Pin Name Dir Description
2 RXD
Receive Data
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3 TXD
5 GND
Transmit Data
System Ground
3. TEL:
RJ-11 connector, FXS interface is for connecting the analog phone sets or
trunk line of PABX.
4. LINE:
RJ-11 connector, FXO interface is for connecting the extension line of
PABX or PSTN Line.
5. 12V DC:
Input AC 100V~120V;output DC12V.
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2. How to Setup and Connect basically
2.1 System Requirement
1. One PC (a) Pentium 100 or above, 64 RAM, Windows 98 or above.
(b) Ethernet card or COM port
2. One standard straightforward RS -232 cable (female connector to Gateway
side).
3. Analog telephone sets or the PBX trunk Lines.
4. PBX extension Lines or PSTN Lines.
5. Software tools – Hyper Terminal, TELNET, Web Browser.
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2.2 IP Environment Set ting
User must prepare a valid IP address, complied with IP Network, for Gateway’s
proper operation.
For testing the validation of chosen IP address, using the same IP
configuration in other PC or Notebook, and then try to connect to Public
Internet (go to well-known website, receive Internet mail, or ping a specific
public IP address). If it works, use the same IP address and network
configuration for Gateway.
Please follow up the step for the configuration of your computer or notebook.
2.2.1 For Window s 2000/NT
Please make sure that the network interface of your computer is working fine
and the cross over line (RJ-45) is connecting with the computer correctly or
you could use a hub to connect with your computer and this gateway. Turn on
your computer and configure the network parameter as follow:
1Go to the start menu and enter the setting area. Click control panel .
2Enter the network configuration.
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Figure 2.1: Network Configuration
3Select the Property of the LAN card.
4 Setup the ip address, subnet mask and default gateway as below:
Figure 2.2: Configure the network
5Click OK after you finished the network setup.
The default ip address, netmask and default gateway address of the gateway
is 10.1.1.3, 255.0.0.0, 10.1.1.254.
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2.3 Net work configurations in your gateway
1 Key in the ip address of the gateway (http://10.1.1.3) with the browser. (see
figure 2.3)
Figure 2.3: WEB Browser
2 After key in the ip address, you have to enter the user name and password
to enter the WEB configuration. (Username: root ; No password) (see
figure 2.4)
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Figure 2.4: Login the username and password
3 You will enter the main page of the configuration after key in the login name
and password correctly: (see figure 2.5)
Figure 2.5: The main WEB configuration
4 Press the Network Interface to configure the networking of your gateway.
(see figure 2.6)
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Figure 2.6: The Network Interface
2.3.1 Static ip address
1 Please get the correct ip address, netmask and default gateway address
from your ISP firs t. Press the OK button if you finished. (see figure 2.7)
Figure 2.7: Configure the static ip address
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2Press the commit if you finish the configuration. (see figure 2.8)
Figure 2.8: Commit the data
3Press the reboot if you want the configuration executed. (see figure 2.9)
Figure 2.9: Reboot the system
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2.3.2 DHCP mode
1 Enable the DHCP if you are using the cable modem or DHCP server. (see
figure 2.10)
Figure 2.10: Enable the DHCP function
2 Please commit the data and reboot the machine after you enable the DHCP
function.
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2.3.3 PPPoE mode
1Switch to the PPPoE mode and press the “OK” button. Press the Network
Interface button after the “OK” button. (see figure 2.11)
Figure 2.11: Switch to the PPPoE mode
2 Enter the Login account and password. Press the “OK” button if the
configuration is finished. (see figure 2.12)
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Figure 2.12: Enter the Account and password
2 Please commit the data and reboot the machine after you finished the
configuration about the PPPoE function.
2.4 M aking a VoIP Call
There are two modes that you could configure the gateway for making VoIP
calls. One is the Peer-to-Peer mode, another is Proxy mode. The
configurations and functions are different. Please make sure about the mode
you want and follow up the step to configure your gateway.
2.4.1 Configure the gateway into the Peer-to-Peer mode
1 Enter the SIP Configuration table and change the mode to Peer-to-Peer.
Define the port numbers whatever you like. Press the “OK” button if the
configuration is all finished. (see figure 2.13)
Figure 2.13: Configure the Peer-to-Peer mode
2 Enter the Phone Book configuration table and configure the name, ip
address and phone number of the destination. (see figure 2.14)
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Figure 2.14: Phone Book
【Example】
Figure 2.15: The example of Phone Book configuration
The name of the destination: test
The E164 number (phone number) of the destination: 123
The ip address of the destination: 10.1.1.100
Drop prefix: Enable – The e164 number you define will be deleted
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Disable – The e164 number you define will be kept
Insert prefix: To add a number you define in this table
Press the “Add Data” button when you finished, and the new table will display
on the first index if you press the Phone Book configuration button .
4 Please Commit it and Reboot the system if the configuration is finished. (see
figure 2.16)
Phone Book is only for the Peer-to- Peer mode and could
support forty records.
Figure 2.16: To show the Phone Book record
【The application in the drop and insert function】
Input (E164) Drop Insert Output
100 Disable X 100
200 Disable 0 0200
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300 Enable X X
400 Enable 500 500
※ X – Do not enter any numbers
2.4.2 Configure the gateway into the Proxy mode
1 Enter the SIP Config table and change the mode from Peer-to-Peer to Proxy.
To change the Proxy information from your service provider (Ex: The Proxy
IP, Domain and Line numbers). (see figure 2.17)
Figure 2.17: Configure the Proxy info
2 Press the OK button that is on the bottom of this page to save the
configuration.
3 Switch to the Security Config page and put the user account and password
in the correct table. Please get this info from your ITSP. Press the OK
button if the configuration is finished. (see figure 2.20)
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Figure 2.20: Configure the Security info
4 Press the Commit Data and Reboot System buttons when you finished the
configuration.
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3. Advanced
There are too many advanced commands for the advanced users. The
following chapters are based on the application layer. Please get the info what
you need. If you need the command, please watching the chapter of Command
Line Interface.
3.1 Network Interface
Users have to configure the Network configurations in this page. This gateway
will be work while it is connecting with the internet network. Please get more
info from the following descriptions. (see figure 3.1)
Figure 3.1: Network Interface
u IP Address – Define the ip address for your networking if it is the fixed ip.
Please get this info from your ISP.
u Subnet Mask – Define the mask address for your networking. Please get
this info from your ISP.
u Default Routing Gateway – Define the default gateway for your networking.
Please get this info from your ISP.
u IP Mode – To configure the fixed or dynamic ip address for this unit.
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Please configure to PPPoE if the ADSL is using the PPPoE type.
u HTTP Port – To configure the HTTP port for access this unit from the
remote side.
u DNS primary – To configure the first ip address for the DNS server.
u DNS secondary – To configure the second ip address for the DNS server.
u SNTP – Enable the SNTP server registering function if user wants to get
the correct time from the Command Line Interface.
u SNTP Server Address – Enter the correct ip address of the SNTP server
or get the incorrect time from the Command Line Interface.
u GMT – Configuring the time area for the time display in the Command
Line Interface.
u IP Sharing – Enable this function if the gateway is behind the IP sharing
device.
u IP Sharing Server Address – Enter the WAN IP address of the IP sharing
device if it is the fixed ip.
u PPPoE User Name – To configure the user name for the PPPoE
connection.
u PPPoE Password – To configure the password for the PPPoE connection.
u PPPoE IP Address – In the PPPoE mode, this table will show the ip
address that this unit gets from the ISP.
u PPPoE Destination – In the PPPoE mode, this table will show the default
gateway address that this unit gets from the ISP.
u PPPoE DNS primary – In the PPPoE mode, this table will show the DNS
ip address that this unit gets from the ISP.
u After Remote Host Disconnection – This unit will reboot and re-connect to
the ISP
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3.2 SIP Config
This WEB page will help user to configure the information about the dial mode,
SIP information and the ports for the communication. Please get more info
about this configuration from the below detail descriptions. (see figure 3.2)
Figure 3.2: SIP Configuration
u Mode – Pick up the calling mode for this gateway.
n Peer-2-Peer : It only supports the peer-to-peer mode and users have
to define the phone book for this mode.
nProxy : Users have to register on the Proxy if users picked up this
option.
u Proxy IP Address – Enter the proxy ip if users pick up the proxy mode.
u Outbound Proxy – This version could support the outbound proxy, users
could define the ip address or domain name in this table.
u Proxy port – Users could change the destination port of the proxy. Before
this port changed, please make sure about that the Proxy port could be
changed in your proxy. The default value is 5060.
u Prefix String – Users could define this if the registration name was a
phonetic alphabet not the numbers.
u Line 1 Number – The phone number of the Tel 1.
u Line 2 Number – The phone number of the Line 1.
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u Line 3 Number – The phone number of the Tel 2.
u Line 4 Number – The phone number of the Line 2.
u SIP port – Users could change the sip port of this unit for the registration.
u RTP port – Users could change the beginning RTP ports in this table.
u Expire – Users could change the expire time for the register message
sending.
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3.3 Security Config
Some proxy will include the security policy. The endpoint may need the user
account and password for the registration. If these are necessary, users could
put the correct account and password in the correct table. (see figure 3.3)
Figure 3.3: Security Configuration
u Line 1 Accou nt – The user name for the line 1 account.
u Line 1 Password – The password for the line 1 account.
u Line 2 Account – The user name for the line 2 account.
u Line 2 Password – The password for the line 2 account.
u Line 3 Account – The user name for the line 3 account.
u Line 3 Password – The password for the line 3 account.
u Line 4 Account – The user name for the line 4 account.
u Line 4 Password – The password for the line 4 account.
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3.4 Line Configuration
The Line configuration will show the status of the registrations and the ports. It
includes the hunt group, hotline, and no answer forward configuration. Press
the Line configuration button to enter configuration table (see figure 3.4)
Figure 3.4: Line Configuration
u Type – Show the type of this port. There are only two types of this gateway.
One is FXS type another is FXO type. It couldn’t be changed.
u Hunting Group – Define the group number of this port. When the port is
busy, the call could be transferred to another port in the same group.
u Hotline – Enab le or Disable the hotline mode. The hotline mode will be
enabled if you enter the hotline number. The default setting is disabled.
u FWD Type – This version could support the call forward with the Proxy.
Users could define the forward type in this table. The call forward could
only support the FXS port as so far.
n Disable – This selection will make the forward function disabled.
n Un conditional – This selection will enable the un-conditional forward
function. Every call will be routed to another destination whatever the
port is busy or not.
nBusy – The selection will enable the busy forward function. Every call
will be routed to another destination if this port is busy.
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The range of the forward time
, the No Answer forward number is
seconds. The call will be forwarded to the
The no answer forward function could support the IP or PSTN side. If users
he PSTN side, please define the routing table for
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The busy forward couldn’t be work if users define the hunt group function.
This will be work without the hunt group. Please make sure about that the
group number is different before you enable the busy forward function.
nNo Answer Forward – This selection will enable the no answer
forward function. Every call will be routed to another destination if the
call didn’t answer by someone.
The no answer forward function should be used with the forwardtime
command (in the System configuration table).
is 5 to 65535. If the function is enabling
123, and the forward time is 5
destination with 123 numbers when the origin port didn ’t answer the call.
u Fwd Number – Please put the number for the call forwarding in this table.
want the calls routed to t
some special number.
u Registration – To show the gateway registered on the GK or not.
u Status – To show the port is busy or ready.
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3.5 System Configuration
There are some parameters in the system configurations, please get more
detail as following. (see figure 3. 5)
Figure 3.5: System Configuration
u Keypad type – There are tow types for the Keypad. On is the In-Band type,
another is the RFC2833 type. User could define the keypad type for the
dialing.
u Inter Digit Time – It’s the time for the time out during the dialing numbers.
u Forward Time – It’s the time for the no-answer-forward. Users have to
configure it with the no-answer-forward function.
u Ring Time – FXO will detect the ring tone according this time.
u End of Dial – It will transfer the digit “#” if this function is disabling.
u Hardware Type – It’s for the hardware issue.
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There are two firmware versions for the 2S2O. One is for G.723 only, another is
the codec in G.729. User could check out the firmware version first and
at the same
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3.6 Voice Setting
User could define some parameters about the voice in this voice-setting page.
(see figure 3.6)
Figure 3.6: Voice Setting
u Codec Priority :It’s for the codec setting. User could use the codec, which
they want by the setting.
for G.729 only. It with no meanings if users use the G.723 version and configure
configure the codec second. 1S1O could support all the codec
times.
u Frame Size:It’s the packet size for all codec. It will take more bandwidth if
users configure the packet size in the minimum value.
u G723 Silence Suppression:For the VAD and CNG function support.
u Volume:To adjust the gain of the output, input and dtmf.
u Echo Canceller :To enable the echo cancellation function.
u Jitter Buffer :To adjust the Jitter Buffer size to avoid the packets losing.
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A large jitter buffer causes increase in the delay and decreases the packet loss.
A small jitter buffer decreases the delay but increases the packet loss. The size
ork, which varies with
time. Typically the packet loss should be less than 10% for a good quality of
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of the jitter buffer depends on the condition of the netw
speech.
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the call transfer function. Users could define a range for
the low and high flash time. Before you change this range, please make sure
set could generate the flash signal in this range.
The FXS will know that is the flash signal if the flash time will in this range.
. For example, if the
ms, the FXO will generate the
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3.7 Phone Pattern
The FXSO could generate some tones, such like the busy tone, dial tone, ring
back tone and second dial tone… etc. Users could adjust these tones or get the
detail info from this page. (see figure 3.7)
Figure 3.7: Phone Pattern
u Ring Cadence – Adjust the pattern for the cadence of the Ring tone.
Including the Frequency, On time, Off time and the gain level.
u Ring Back Tone – Adjust the pattern for the Ring Back Tone. Including the
High, Low frequency, High, Low Level, and the On, Off time.
u Busy Tone – Adjust the Busy Tone.
u Dial Tone – Adjust the parameters for the dial tone playing.
u 2nd Dial Tone – Adjust the second dial tone.
u Flash Frequency – Adjust the High, Low frequency for the Flash. (unit :
ms)
The Flash will affect
about that the analog phone
For the FXO, it will generate the flash between this range
low flash time is 400ms and the high is 800
600ms for the flash time.
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3.8 Tone Setting
The Tone Setting is for the Tone detecting. The call will be dropped if the
pattern of the tone from PSTN side is as same as the pattern in the disconnect
tone table. The same result for the Ring Back Tone. User could define the
pattern of the disconnect tone if the disconnect tone from PSTN side is not the
standard tone. (see figure 3. 8)
Figure 3.8: Tone Setting
u Disconnect To ne – Users could put the correct pattern of the disconnect
tone in this table. The call will be dropped if the tone from PSTN side is
match with these patterns. Users could have four tables for the disconnect
tone.
u Remote Ring Back Tone – User could adjust this table to help the FXSO
gateway to detect the Remote Ring Back Tone. There could be four tables
for the configuration.
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It will be the drop function if user enable the Drop Prefix function and
It will be the insert function if user disable the Drop Prefix function and
It will be the replace function if user enable the Drop Prefix function
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3.9 Phone Book
The Phone Book configuration is only support the gateway in Peer-to-Peer
mode. Please refer the chapter 2 about the Peer-to-Peer mode. (see figure
3.9)
Figure 3.9: Phone Book
u Index – The list number of the Phone Book.
u Name – The name for this contact number.
u E164 – The dialing number for the calling side.
u IP Address – The destination IP address for this phone number.
u Port – The call signal port of the destination.
u Drop – Support the drop function. Enable is for enable this drop function;
Disable is for disable this drop function. The Drop Prefix will drop the E164
number, which you had configured in the E164 table.
u Insert – Support the insert digits function.
1.
put nothing into the Insert Prefix table.
2.
put the digits into the Insert Prefix table.
3.
and put the digits into the Insert Prefix table.
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u Add Data – Press this button if users fill the entire information table above.
u Delete Date – If users want to delete the record from the table, enter the
index number first and press this button. The record will be deleted.
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3.10 Prefix Configuration
The Prefix functions are using the drop and insert function (see figure 3.6).
Figure 3.10: Prefix Configuration
u Index – The list number of the Phone Book.
u Prefix – The prefix number of the whole numbers that could be into this
gateway
u Drop – The drop function. Enable this function by the Enable button;
Disable this function by the Disable button.
u Insert – The insert function. Users could enter the digits that you want to
insert in this number.
u Add Data – Press this button if users fill the entire information table above.
u Delete Date – If users want to delete the record from the table, enter the
index number first and press this button. The record will be deleted.
This function is just like the Phone Book configuration. But it will make the drop
and insert function in the GK routed mode. All the numbers into this gateway
will check out the prefix table first and find out the destination in the Routing
Table.
There is an example about the configuration, please follow up these steps.
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1 Press the Prefix Configuration button to enter the configuration table (see
figure 3.10)
2 Enter the index number. Put the prefix numbers you will dial in the prefix
table, enable (disable) the drop function and enter the numbers you want
to insert (see figure 3.11)
Figure 3.11: Configure the Prefix Table
The usage is as same as the drop, insert function of the Phone Book.
Input (Prefix) Drop Insert Output
100 Disable X 100
200 Disable 0 0200
300 Enable X X
400 Enable 500 500
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3 Press the Prefix Configuration button to reload the configuration table (see
figure 3.12)
Figure 3.12: Show the added table
4Please Commit it and Reboot the system if the configuration is finished.
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tion will be the FXS port
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3.11 Routing Table
Routing Table is a rule to define the destination of the calls you make. You
could define the rules by the number you dial or by the ports. The Routing
Table button will show you the configuration table (see figure 3.13).
In fact, there are three directions of the incoming calls (from IP, FXS and FXO
side). The explanation of the default routing is as below:
The location with
the incoming calls
The location with
the destination
IP (Default) Fxs
Fxs (Default) IP
Fxo (Default) IP
The explanation
The destina
when the calls from the IP side
without any define rules.
The destination will be the IP side
when the calls from the FXS port
without any define rules.
The destination will be the IP side
when the calls from the FXO port
without any define rules.
u Index – The list number of the Route Table.
Figure 3.13: Routing Table Configuration
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u Default – For change the default setting. Users have to pick the direction
for the default setting changed.
u Prefix – The prefix number for the dialed digits. The call will be followed
this route table if the prefix number was matched.
u Destination – To decide the destination for this route table.
u E.164 – The E.164 number of the destination.
u Mini Digits – The mini digits requirement for this route table.
u Max Digits – The max digits requirement for this route table.
u Hunt Method – Enable the Hunt Group function and pick up the hunt type.
n NONE : Disable the Hunt Group function.
n GROUP : The Hunt Group function will working for the same group.
User could configure the group in the Line Configuration table.
nALL : The Hunt Group will working for the same type.
None – Disable this function
Group – The call will search other ports to be the destination with the same
group if the origin destination is busy.
All – The call will search other ports to be the destination with the same type
the origin destination is busy.
About the Group setting, Please get more info from the Line Configuration.
u Add Data – Add a new record for the route table.
u Delete Data – Delete a record for the route table.
u Change Default – Change default route table.
3.11.1 Change the default routing
Please follow up the steps if you want to change the default routing:
1 Pick up the side for the incoming calls and define the destination of this side.
Press the Change Default to save the data. (see figure 3.14)
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Figure 3.14: Change the Default Setting
2 The default setting is changed after you press the Change Default button.
Please press the Routing Table button again to show the new setting. (see
figure 3.15)
Figure 3.15: The Default Setting Changed
3Please Commit it and Reboot the system if the configuration is finished.
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3.11.2 Add a new Routing Table
1 The default setting is changed after you press the Change Default button.
Please press the Routing Table button again to show the new setting. (see
figure 3.16)
FIGURE 3.16:EDIT AND ADD A NEW ROUTING TABLE
2 Press Add Data button to save the configuration and press the Routing
Table button again to reload the configuration. (see figure 3.13)
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Figure 3.17: New Special Routing
The explanation of figure 3.17 is as below:
When the user dial 8 with the first digit of the numbers from FXS or IP side.
And the numbers you dial is between 1 and 8 digits. If this number matches the
rule, it will be transferred to the FXO port whose E164 number is 1003.
3Please Commit it and Reboot the system if the configuration is finished.
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s not ready for the IP
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3.12 FXO Password
You will get the IVR if you make calls from PSTN side. The IVR will ask you the
password you set, and you co uld make other calls to IP side if the password
you type is correct. Please press the FXO Password button to configure the
password (see figure 3.18)
Figure 3.18: FXO Password
u Index – The number of this table.
u Password – The password you set.
This function is only for the calls from the PSTN side. It’
side as so far.
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3.13 IP Packet ToS
The Type of Service should be worked with the network router. The router will
check all the packets if it supports the TOS function. There is a field in the
packet for the TOS value. This WEB is for users to configure these values to
make the packets with the correct values for the TOS service from the gateway.
(see figure 3.19)
Figure 3.19: ToS
According to the RFC 1349 document, the ToS value as following:
These values are the Binary format. Please change to the Decimal and put
these values in to the correct table.
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Please remember the password you configure for the account. It will be more
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3.14 Password
There are two-login accounts in this unit. One is the account root another is
administrator. The default setting for these two accounts are empty. Users
could define the passwords for these two accounts. Please get more info from
the following description. (see figure 3.20)
Figure 3.20: Password
u root – The password for the root account.
u administrator – The password for the administrator account. This account
couldn’t upgrade the 2M and boot rom file.
u Current Password – Enter the original password for the account.
u New Password – Enter the new password for the account.
u Confirm New Password – Enter the new password again.
u Change – This button will make the configurations saved and next time
login will need the new password.
u Abort – Abort the configuration of the password changing.
difficult to access it if you forgot the password.
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3.15 RTP Pay Load Type Configuration
There are more types for the RTP Payload. This web page could support users
define the Payload Type for some special payload. (see figure 3.21)
Figure 3.21: Pay Load Type Configuration
u RFC2833 Payload Type – To define the payload type for RFC2833 type.
u DTMF Payload Type – To define the payload for the DTMF type.
u FAX Payload Type – To define the payload for the FAX type.
u FAXByPass Payload type – To define the payload for the FAX by Pass
type.
u MODEMByPass Payload Type –To define the payload for the Modem by
Pass type. (This is no use for the hardware as so far.)
u Redundancy Payload Type – To define the payload for the Redundancy
type.
u MODEMRelay Payload Type – To define the payload for the FAX by Pass
type. (This is no use for the hardware as so far.)
The modem function couldn’t be work in this kind of VoIP unit.
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3.16 Version and Information
Users could get more detail about the software version for all the parts in this
web page. (see figure 3.22)
Figure 3.22 Version and Information
u Boot Rom – The version of the Boot Rom layer.
u Application Rom – The version of the Application Rom layer.
u DSP Application – The version of the DSP Application Rom layer.
u DSP Kernel – The version of the DSP Kernel layer.
u DSP Test Code – The version of the DSP Test Code layer.
u Greeting – The version of the Greeting file.
u ASK Pin – The version of the ASK Pin file.
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3.17 ROM Upgrade
User could update the firmware just by the web configuration interface. There
are two type for the upgrading procedure. One is using the TFTP server,
another is using the FTP server. Please follow the step to update the gateway
firmware version. (see figure 3.23)
Figure 3.23 ROM Upgrade
u TFTP/FTP server IP Address – Put the ip address of the TFTP or FTP
server in this table.
u Target File name – Put the target file name in this table.
u Method – There are two upgrade methods for the upgrade procedure.
One is TFTP and another is FTP. Please change to correct method for the
upgrading.
u FTP Login – Please enter the login name and password for the FTP
upgrade method. This is necessary if user change the method to the FTP.
u Target File Type – Please pick up the correct file type for upgrading. If the
file name and the file type is unconformable, the upgrade procedure will
be failed.
u OK – Press the OK button if all the info above are correct. The unit will
start to download the firmware file from the TFTP or FTP and write to the
flash after the downloading.
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【Updating the firmware by the FTP server】
1 Pick up the “Rom Upgrade” button to enter the upgrading web page and
switch to the FTP method. (see figure 3.24)
Figure 3.24: ROM Upgrade for FTP
2 Key in the IP address, the login name, password of your FTP server and the
correct file name, file type. (see figure 3.25)
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red blank. The Target File Type
has to be matched with the Target File name. Please put the correct info
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Figure 3.25: FTP information
Please pay more attentions about the
about the Target file in this table.
3Press the OK button to execute the upgrade procedure.
4Please press the “Reboot System” button to make it reboot.
【Updating the firmware by the TFTP server】
1Downloading the TFTP program from our web site and install it first.
Executing the TFTP program before you want to use the TFTP upgrade
method.
2 Pick up the “Rom Upgrade” button to enter the upgrading web page and
switch to the TFTP method. (see figure 3.26)
Figure 3.26: ROM Upgrade for TFTP
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s only for the application layer firmware. In fact, the latest version
Please use the flash clean web to make all the configuration back to the
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3 Key in the IP address of your TFTP server, pick up the file type for your
upgrade file and the correct file name for upgrading. (see figure 3.27)
Figure 3.27: TFTP information
4 Press the OK button to execute the upgrade procedure.
Please pay more attentions on the file name you used. The file name
with the prefix “2m” is the complete firmware, it will take more times for
the downloading and upgrading. About th e file name without the prefix
“2m”, it ’
firmware is only changed in the application layer.
default setting if th e upgrade procedure was finished.
5Please press the “Flash Clean” button when the procedure is finished.
6After pressing the “Flash Clean” button, please press the “Reboot System”
button to make it reboot.
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3.18 Flash Clean
Users could make all the configurations back to the default setting by this
button. The password of the account and the networking configuration couldn ’t
be back to the default setting by this command. (see figure 3.28)
Figure 3.28: Flash Command
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3.19 Commit
This web page could save the configurations if users change some
configurations. This is necessary for users change the configurations. (see
figure 3.29)
Figure 3.29: Commit the data
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3.20 Reboot System
This web page will restart the whole system. This is the necessary step for the
changing the configurations and makes it executed. (see figure 3.30)
Figure 3.30: Reboot System
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4. Command List
4.1 Hyper Terminal Setting
A terminal emulator is needed when using RS -232 port to configure Gateway.
There are kinds of terminal emulator software. Here, we use Microsoft
HyperTerminal to depict how to set up terminal emulator:
1. Execute the Hyper Terminal program, and then the following windows will
pop-up on the screen. (START – Program files – Accessories –
Communication – Hyper Terminal)
Figure 4.1: Hyper Terminal
2. Define a name such as ‘wg37’ for this new connection.
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Figure 4.2: Edit the name of the connection
3. After pressing OK button, the next window appear, and then choose
COM1/2 Port, which you are going to use.
Figure 4.3: Pick up the right interface to use
4. Configure the COM Port Properties as following:
u Bits per second: 9600
u Flow control: None
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Figure 4.4: Configure the right Bps and control
5. Press ‘OK’ button, and then start to configure Gateway.
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4.2 Command List
4.2.1 [help]
Type help or man or ? to list all the available command.
usr/config$ help
help help/man/? [command]
quit quit/exit/close
debug show debug message
reboot reboot local machine
flash clean configuration from flash rom
commit commit flash rom data
ifaddr Internet address manipulation
time show current time
ping test that a remote host is reachable
sysconf System information manipulation
sip SIP information manipulation
security Security information manipulation
line Line information manipulation
route Routing information manipulation
prefix Prefix drop/insert information manipulation
pbook Phone book information manipulation
voice Voice information manipulation
phone Setup of call progress tones and ringing
tone Setup of disconnect tone
fxopwd Setup of FXO password
record Record voice for greeting and ask pin code
tos IP Packet ToS (Type of Service)values
pt DSP payload type configuration and information
rom ROM file update
passwd Password setting information and configuration
usage: help [command]
4.2.2 [quit]
Type quit will quit the Gateway configuration mode and turn back to login
prompt (in console mode) or disconnect (in TELNET mode).
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usr/config$ quit
Disconnecting...
login:
Note: It is re commended that type the “quit” command before you leave the
console. If so, Gateway will ask password again when next user connects to
console port.
4.2.3 [debug]
Open debug message will show up specific information while Gateway is
in operation. After executing the debug command, it should execute
command debug -open as well. One example is demonstrated below.
-print print out current [ifaddr] settings and status
-ip assign IP address for Gateway
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-mask assign internet subnet mask
-gate assign IP default gateway
-mode Switch the network type (0 = Static IP; 1 = DHCP mode 2 =
PPPoE mode)
-sntp Simple Network Time Protocol (1 = ON; 0 = OFF) When
SNTP function is activated, users have to specify a SNTP
server as network time source. An example is demonstrated
below:
usr/config$ time
Current time is THU JAN 01 05:29:23 1970
4.2.9 [ping]
Use ping to test whether a specific IP is reachable or not.
For example: if 192.168.1.2 is not existing while 192.168.1.254 exists.
Users will have the following results:
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usr/config$ ping 192.168.1.2
no answer from 192.168.1.2
usr/config$ ping 192.168.1.254
PING 192.168.1.254: 56 data bytes
64 bytes from 192.168.1.254: icmp_seq=0. time=5. ms
64 bytes from 192.168.1.254: icmp_seq=1. time=0. ms
64 bytes from 192.168.1.254: icmp_seq=2. time=0. ms
64 bytes from 192.168.1.254: icmp_seq=3. time=0. ms
System information and configuration
Usage:
sysconf [-idtime digit][-keypad dtmf][-prefixsw digit]
[-prefixdisab digit][-usrdefprefix digits]
[-codec digit][-localrbt digit][-forwardtime digit]
[-hwtype digit][-gwprefix digit][-ring on_time off_time]
[-fxotype digit]
sysconf -print
-print Display system overall information and configuration.
-idtime Inter-Digits time.(1~10 sec)
-forwardtime Forward time for FXS line if no answer.(5~65535 sec)
-keypad Select DTMF type: 0=In-band,
1=RFC2833.
-hwtype Hardware type.(Auto:0 / 1FXS+1FXO:1 /
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2FXS+2FXO:2)
-callerid CallerId Type .(Caller type, 0: none, 1: FSK)
-ring The ring time for ring detection.(Uint:ms)
-rba The number of ring times before answer.(1~5)
-eod End of dial.(Enable: 1 / Disable : 0)
-connect Auto connect time.(0=Disable, for 1~65535 sec)
Example: sysconf -ring 500
usr/config$
Parameters Usage:
-print print out all current settings
-idtime set the duration(in second) of two pressed digits in dial mode
as timed out. If after the duration user hasn’t pressed next
number, it will dial out all number pressed. (1-10 seconds)
-forwardtime set forward time(5-65535 seconds)for FXS Line. If called
party hasn’t answered the call in this time, call will be
forward to assigned number in [line] command. (please
refer to [line] command for forward setting)
-keypad DTMF replay type. When value is “0”, Gateway will transfer
DTMF signal via In -Band type, “1” via RFC2833 type. Users
can adjust the value according to various applications.
1. number (instead of Line number of FXO Line)+ PSTN
number to make a call to PSTN side connected with FXO
Line.
2. After gateway-prefix-drop function is enabled, user must
remember to re-configure line number of FXS Line,
because line number of FXS Line must remove prefix
number. For example, origin line number of FXS line is
1001, prefix is 100, since prefix number will be drop, once
gateway has incoming call 1001, after drop gateway prefix
100, it will search line number “1”. So line number must be
set as “1”.
-hwtype application rom file of 37 series are the same no matter how
many ports is the module, so after user downloads the
application rom file, user can select which hardware type
is . ”0” means gateway will automatically detect the hardware
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type, “1” means the hardware type is 1FXS+1FXO, “2” means
the hardware type is 2FXS+2FXO.
Note:
The default value is to auto detect hardware type. Usually it is
not necessary to change this setting. Please make sure about
your Hardware Type, Gateway may be not functional if set
wrong hardware type.
-callerid Caller ID function disables or enable. The caller ID will be
disable if user out the value 0 in this configuration or users
could put the value 1 to enable the FSK caller ID function.
Please check out the phone set could support the FSK caller
ID display. (0 for disable ; 1 for FSK type.)
-ring ring time for ring detection(in ms). When Gateway has
incoming call from PSTN side to FXO port, Gateway will
determine it is a ring but not noise only if it is longer than this
ring time.
Note:
In Taiwan the ring time of PSTN usually is 1000ms, so if user
set ring time longer that 1000ms, FXO port may not be able to
pick up the call from PSTN side.
-rba When the calls from the PSTN side, FXO port will off hook if
the ring time is matched with this number.
-eod It will transfer the DTMF in “#” if users disable the end of dial
function. Users have to press the key pad in “#” if the end of
dial function is enable.
-connect The unit will send the connect message to the IP side
automatically. This function just supports the
one-stage-dialing function.
4.2.11 [sip]
This command is for sip configuration related parameters.
-px to assign the ip address of the proxy when Gateway is in
proxy mode.
-pxport define the proxy port for the registrati on or call.
-prefix this will be prefix the alphabets before the sip line number.
-line1 assign FXS TEL1 number.
-line2 assign FXO Line1 number.
-line3 assign FXS TEL2 number.
-line4 assign FXO Line2 number.
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Note:
User can also set “x” in line number to disable the port. If the port is
disabled, it can only receive calls but not calling out.
Note:
1. This is for 3702, for 3701 model, there are only line1 and line2
command.
2. No matter in Proxy or P2P mode, user only needs to dial line number
to reach local port. For example, in P2P mode, user wants to dial from
FXS TEL1 to FXO Line1, only need to dial number of line2.
-expire It just like the TTL function in H323, the gateway will make
sure the registration is success or not for a period times.
-port define the local sip port for this gateway.
-rtp to allocate RTP port range— NOT RECOMMENDED. This
may be used when RTP port range conflicts with Firewall
policy. (each port of Gateway use 2 RTP ports)
Gateway line information and configuration
Usage:
line -config number [hunt number][hotline number][forward number]
line -print Gateway line information.
hunt Hunting group.
hotline Hot line configuration.
fwdtype forward type for FXS line.
To IP:
0:disable
1:un condition
2:busy
3:no answer:
To Local:
0:disable
3:no answer
forward forward for FXS line.
Example:
line -config 1 hunt 1 hotline 1003 forward 1002
usr/config$
Parameter Usages:
-print print out all current settings of line
-config determine which line to configure
-hunt set hunting group flag of each line. User can assign different
hunt group number represent different hunt group. For
example, if user assigns FXS TEL1 as hunt group 1, and FXS
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TEL2 as hunt group 2, they will be determined as 2 different
groups. On the other hand, if user assigns FXS TEL1 as hunt
group 1, and FXS TEL2 as hunt group 1 too, when having
incoming call to FXS TEL1, which is busy, this call will be
route to FXS Line2.
Note: FXO Lines and FXS TELs are treated as 2 different
groups, so even they are in the same hunt group, call will only
be routed to the same FXS or FXO Lines.
-hotline set hotline table. The Hotline Mode is applied in limited two
channels. User just picks up the phone set of one FXS TEL or
calls in one FXO line, and gateway will automatically dial out
a phone number. In the other hand, user will hear ring back
tone or dial tone immediately depended on configurations of
destination device.
Note: This function can both work in Proxy or P2P mode.
(1) Call out from FXS Line
Proxy Mode Usage:
Set gateway under proxy mode.
Create a Hotline table with “line ” command.
------------------------------------------------------------------------------In this example means: if user picks up phone set of FXS
Line1, gateway will automatically dial out “1001”.
P2P Mode Usage:
Set gateway under P2P mode.
Create phone book table with “pbook ” command.
Create a Hotline table with “line ” command.
usr/config$ pbook –add name sipfxso ip 10.1.1.1 e164 1001
usr/config$ line –config 1(3) hotline 1001
------------------------------------------------------------------------------In this example means: if user picks up phone set of FXS
Line1, gateway will automatically dial out IP address of
“1001”.
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(2) Call out from FXO Line
Proxy Mode Usage:
Set gateway under proxy mode.
Create a Hotline table with “line ” command.
------------------------------------------------------------------------------In this example means: if user calls in FXO Line1, gateway
will automatically dial out “1001”.
P2P Mode Usage:
Set gateway under P2P mode.
Create phone book table with “pbook ” command.
Create a Hotline table with “line ” command.
usr/config$ pbook –add name sipfxso ip 10.1.1.1 e164 1001
usr/config$ line –config 2(4) hotline 1001
------------------------------------------------------------------------------In this example means: if user calls in FXO Line1, gateway
will automatically dial out IP address of “1001”.
-fwdtype This version could support all the forward function, including
the unconditional, busy and no answer forward function. All
the forward function will make all the incoming calls routed to
other number which users define in the forward table. The
busy won’t be worked if the hunt group function had been
enabled. The no answer forward function could support the IP
and local forward at the same time. Please define the special
route table for the local no answer forward function. The
forward could support the FXS port only.
-forward Users could define the forward number for the forward
function.
port type” e164 “SIP number of port” min “minimum
digits needed” max “maximum digits can ’t be
exceeded”)
-delete delete a routing rule in routing table ( route –delete “index of
routing rule”)
-modify modify a routing rule in routing table. (route –modify “index
of routing rule” prefix “prefix number” dst “destination
port type” e164 “SIP number of port” min “minimum
digits needed” max “maximum digits can ’t be
exceeded”)
-ip create routing table for incoming call from IP side. (route –ip
dst “destination port type” e164 “SIP number of port ”)
-fxs create routing table for incoming call from FXS TELs.
(route –fxs dst “destination port type” e164 “SIP number of port ”)
-fxo create routing table for incoming call from FXO Lines.
(route –fxo dst “destination port type” e164 “SIPnumber
of port ”)
prefix prefix of dialed number
dst destination port, 0 means FXS TELs, 1 means FXO Lines, 2
means IP side, x means no determinate number.
e164 destination SIP number. This only need to be set when routed
port is FXS TELs or FXO Lines to determine which port will
this call be routed to.
min minimum digits needed.
max maximum digits needed.
hunt set hunt method for busy forward. 0 means no hunting, 1
means hunting method follows the rule of [line], 2 means
hunting method is to hunt between all ports in the same type,
for example, destination port is FXS TEL will hunt in all FXS
TELs, destination port is FXO Lines will hunt in all FXO Lines.
Usage Example:
1. route –add prefix 100 dst 0 e164 1001 min 1 max 3 hunt 1
This command means if gateway has incoming call’s prefix number is
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100, and total digits is between 1 to 3, this call will be routed to FXS
TEL 1001, and if TEL 1001 is busy, call will be routed to another FXS
TEL.
2. route –ip dst 1 e164 1002
This command means incoming call from IP side will be routed to FXO
Line of number 1002.
3. route –fxs dst 1 e164 1002
This command means incoming call from FXS TELs will be routed to
FXO Line of number 1002.
4. route –fxo dst 2
This command means incoming call from FXO Lines will be routed to
IP side.
Note:
(1) When destination is IP side, SIP number doesn’t need to
determine. (Ex. route –fxs dst 2)
(2) If user doesn’t want to determine a specific port to route, SIP
number must set as “x”. (Ex. route –ip dst 1 e164 x)
(3) Default value: Incoming call from FXS and FXO ports will be
forward to IP side directly.
4.2.15 [prefix]
This command is for make rules f or drop or inserts prefix digits.
Prefix drop/insert information and configuration
Usage:
prefix -add [prefix number][drop number][insert digits]
prefix -delete index
prefix -modify index [prefix number][drop number][insert number]
prefix -print Prefix drop/insert information.
prefix The prefix of dialed number.
drop Drop prefix(Enable:1/Disable:0).
insert Insert digits.
Example:
prefix -add prefix 100 drop 1 insert 2000
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prefix -add prefix 100 drop 1
prefix -add prefix 100 drop 0 insert 200
prefix -delete 1
prefix -modify 1 prefix 100 drop 0 insert 300
usr/config$
Parameter Usages:
-add add a rule to drop or insert prefix digits of incoming call.(prefix –add prefix “prefix number” drop 0/1 insert “insert number”)
-delete delete a rule to drop or insert prefix digits of incoming call.
(prefix –delete prefix “prefix number”)
-modify modify a rule to drop or insert prefix digits of incoming call.
(prefix –modify prefix “prefix number” drop 0/1 insert “insert number”)
prefix set which prefix number to implement prefix rule.
drop enable or disable drop function.If this function is enabled, Gateway
will drop prefix number on incoming call.
insert set which digit to insert on incoming call.
4.2.16 [pbook]
Phone Book function allows users to define their own numbers, which
mapping to real IP address. It is effective only in peer-to-peer mode.
When adding a record to Phone Book, users also have to reboot the
machine, and the record will be effective immediately.
Phone book information and configuration
Usage:
pbook [-add [name string][e164 number][ip address]
[port number][drop digit][insert number]]
[-modify number [name string][e164 number][ip address]
[port number][drop digit][insert number]]
[-delete number]
pbook -print
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-print Display phone book information and configuration.
-add Add new phone book record)
-delete Delete phone book record
-modify Modify phone book record.
name : 1 ~ 10 characters.
e164 : 1 ~ 10 digits.
ip : IP adress.
port : 1024 ~ 65535.
drop : 0:Disable/1:Enable.
insert : 1 ~ 10 digits.
Example:
pbook -add name test e164 1234 ip 192.168.1.10 drop 1 insert 5678
pbook -delete 1
pbook -modify 1 name test e164 5678 ip 192.168.1.10 drop 0
usr/config$
Parameter Usages:
-print print out current contents of Phone Book. (pbook -print)
Users can also add index number, from 1 to 100, to the
parameter to show specific phone number. (Ex. pbook –print
1)
Note: <index number> means the sequence number in phone
book. If users do request a specific index number in phone
book, Gateway will give each record a automatic
sequence number as index.
-add add a new record to phone book. When adding a record,
users have to specify name, ip, and e164 number to
complete the command.
name name to represent callee.
e164 The SIP number for mapping with IP address of called
ip ip address of called
port user could define the sip port for the peer to peer mode
calling mode
drop drop e.164 number when dial out. 0 means to keep e.164
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number, 1 means to drop e.164 number when dialing out.
insert insert digits.(1~10 digits)
-delete delete a specific record. “pbook –delete 3” means delete
index 3 record.
-modify modify an existing record. When using this command, users
have to specify the record’s index number, and then make the
change.
PhoneBook Rules:
The SIP number defined in phone book will fully carry to destination. It is
not just a representative number for destination ’s IP Address. In other
words, user dial this number to reach the destination, destination will
receive the number and find out if it is matched to it self, including Line
number in some particular device.
4.2.17 [voice]
The voice command is associated with the audio setting information.
There are four voice codecs supported by Gateway.
-print print current voice information and configurations.
-send define packet size for each codec. 20/40/60ms m eans to
send a voice packet per 20/40/60 milliseconds. The smaller
the packet size, the shorter the delay time. If network is in
good condition, smaller sending packet size is recommended.
In this parameter, 20/40/60ms is applicable to G.711u/a law,
and G.729/G.729A/G.729B/G.729AB codec, while 30/60ms is
applicable to G.723.1 codec.
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-priority codec priority while negotiating with other h323 device. This
parameter determines the listed sequence in h.245 TCS
message. The codec listed in left side has the highest priority
when both parties determining final codec. User can also
select the particular codec without others.
usr/config$ voice –priority g723 (only select this codec)
usr/config$ voice –priority g723 g729 g711u g711a (select four codecs,
and g723 is the first choice)
Note:
(1) For 2S2O there are 2 versions of Application rom, please check
out the version of Application rom (rom –print). If the version is
2sipfxso729.102, 2S2O doesn’t have the codec G.723.1. If the
version is 2sipfxso723.102, 2S2O doesn’t have the codec G.729
series.
(2) For 1S1O, the Application rom has the only one version which is
named sipfxso.102 provide all codec.
-volume There are three adjustable value. voice volume stands for
volume, which can be heard from Gateway side; input gain
stands for volume, which the opposite party hears; dtmf
volume stands for DTMF volume/level, which sends to its own
Line.
Note:level of volume is too high or too low may be result in bad
performance while connecting to each other.
-nscng silence suppression and comfort noise generation setting (1 =
ON; 0 = OFF). It is applicable to G.723 codec only. An
example is demonstrated below:
Note: be sure to know well the application before you change voice
parameters because this might cause incompatibility.
4.2.18 [phone]
Gateway’s progress tone is configurable. Default tone value is set
according to U.S. tone specification. Users may adjust the values
according to their own country’s tone specification or users-defined tone
specification.
Phone ring back tone paramter
Ringback Tone frequency high : 480
Ringback Tone frequency low : 440
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Ringback Tone frequency high level : 13
Ringback Tone frequency low level : 13
Ringback Tone tone1 on : 100
Ringback Tone tone1 off : 200
Ringback Tone tone2 on : 1023
Ringback Tone tone2 off : 1023
usr/config$
Note:
For tone simulation, Gateway adopts dual frequencies as traditional
telephone does. If users want to have their own call progress tone, they
can change the value of tones. High and Low frequency/level/cadence
can be configured respectively.
-ring to set RING Tone value.
The played tone type, when Gateway is receiving a call.
-rbt to set Ring Back Tone value
The pl ayed tone type, when Gateway receives a Q.931 Alerting
message. In condition that Gateway is the originate side.
-bt to set Busy Tone value.
The played tone type, when destination is busy.
-dt to set Dial Tone value.
The played tone type, when off hook a phone set of workable
Gateway.
-2dt to set the 2nd dial tone value.
The played tone type, when the unit accept the call hold function,
it will generate the 2nd dial tone.
-flash set the detective flash range in ms, for example, 300-500 ms.
Disconnect tone and remote ring back tone configuration
Usage:
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tone [num][freqHi ][freqLo ][freqHiLev][freqLoLev]
[Tone1ON][Tone1OFF][Tone2ON ][Tone2OFF ]]
tone -print Display disconnect tone configuration.
[num] Tone index(1~4:Disconnect tone / 5~8:Remote ring back
tone).
Example:
tone -print
tone 1 620 480 8 8 50 50 1023 1023
usr/config$
Parameter Usages:
-print show all tone configuration
[num] tone index. 1~4 is disconnect tone, 5~8 is remote ring back
tone.
For FXO ports Gateway must detect disconnect tone to
determine when to disconnect the call, so user must set
disconnect tone of PBX or PSTN network connected to FXO
ports.
When making a call from FXO ports, there are 2 ways to
detect callee has already picked up the call, one is to detect
reverse signal, the other is to detect the termination of ring
back tone, so user must set ring back tone of PBX or PSTN
network.
(If user doesn’t know about the frequency of disconnect tone
or ring back tone, please refer to [record] command to detect
frequency.)
For each tone may has 1 set or 2 sets (high and low) of
frequencies. If user wants to set 0 in on/off time, please set
“1023” represent “0”. (ex. tone 1 620 480 8 8 50 50 1023
1023)
(tone “index of tone” “frequency of high” “frequency of
low” “level of high” “level of low” “on time of high” “off
time of high” “on time of low” “off time of low” )
4.2.20 [fxopwd]
This command is for FXO ports.
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Analizing!! Please wait a moment...
(You coule hang up the call from PSTN if you get this message)
Frequency 1 : 481
Frequency 2 (2623) is more than 1000, please ignore it.
Frequency 3 : 621
tone 4 481 621 8 8 25 25 1023 1023
(Put this value in to the tone table)
tone –print
Disconnect tone 1 paramter
Frequency high : 620
frequency low : 480
frequency high level : 8
frequency low level : 8
Tone1 on : 25
Tone1 off : 25
Tone2 on : 1023
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Tone2 off : 1023
Disconnect tone 2 paramter
Frequency high : 450
frequency low : 0
frequency high level : 8
frequency low level : 0
Tone1 on : 35
Tone1 off : 35
Tone2 on : 1023
Tone2 off : 1023
Disconnect tone 3 paramter
Frequency high : 620
frequency low : 480
frequency high level : 8
frequency low level : 8
Tone1 on : 50
Tone1 off : 50
Tone2 on : 1023
Tone2 off : 1023
Disconnect tone 4 paramter
Frequency high : 621
frequency low : 481
frequency high level : 8
frequency low level : 8
Tone1 on : 25
Tone1 off : 25
Tone2 on : 50
Tone2 off : 50
(Confirm the values is correct or not)
(Key in the commit and reboot command if you finish the procedures as above)
【Example-2】
(Make a call into FXO port)
usr/config$ record -tone
Press (R) to start record...
(Please make sure that you are already finish the steps 2 ~ 7)
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Analizing!! Please wait a moment...
(You coule hang up the call from PSTN if you get this message)
Frequency 1 : 473
Frequency 2 (2623) is more than 1000, please ignore it.
Frequency 3 (1856) is more than 1000, please ignore it.
tone 4 473 473 8 8 25 25 1023 1023
(Please configure the high and low frequency as the same value if you just
get a singal frequency)
tone –print
Disconnect tone 1 paramter
Frequency high : 620
frequency low : 480
frequency high level : 8
frequency low level : 8
Tone1 on : 25
Tone1 off : 25
Tone2 on : 1023
Tone2 off : 1023
Disconnect tone 2 paramter
Frequency high : 450
frequency low : 0
frequency high level : 8
frequency low level : 0
Tone1 on : 35
Tone1 off : 35
Tone2 on : 1023
Tone2 off : 1023
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Disconnect tone 3 paramter
Frequency high : 620
frequency low : 480
frequency high level : 8
frequency low level : 8
Tone1 on : 50
Tone1 off : 50
Tone2 on : 1023
Tone2 off : 1023
Disconnect tone 4 paramter
Frequency high : 621
frequency low : 481
frequency high level : 8
frequency low level : 8
Tone1 on : 25
Tone1 off : 25
Tone2 on : 50
Tone2 off : 50
(Confirm the values is corr ect or not)
(Key in the commit and reboot command if you finish the procedures as above)
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4.2.22 [tos]
IP Packet ToS(type of Service)/Differentiated Service configuration.
IP Packet ToS(type of Service)/Differentiated Service configuration
Usage:
tos [-rtptype dscp]
tos [-sigtype dscp]
tos -print
[-rtpreliab mode]
tos -print
Example:
tos -rtptype 7 -sigtype 0
Parameter Usages:
-rtptype the packages of voice
-sigtype the package of call signal
Note:
The value of rtptype and sigtype is from 0 to 63.
It’s working if it supported by your network.
ROM files updating commands
Usage:
rom [-print][-app][-boot][-dsptest][-dspcore][-dspapp][-greet][-askpin]
-s TFTP/FTP server ip -f filename
rom -print
-print show versions of rom files. (opti onal)
-app update main application code(optional)
-boot update main boot code(optional)
-boot2m update 2M code(optional)
-dsptest update DSP testing code(optional)
-dspcore update DSP kernel code(optional)
-dspapp update DSP application code(optional)
-greeting update greeting voice file(optional)
-askpin update ask pin code voice file(optional)
-s IP address of TFTP/FTP server (mandatory)
-f file name(mandatory)
-method download via TFTP/FTP (TFTP: mode=0, FTP:
mode=1)
-ftp specify username and password for FTP
Note:
This command can run select one option in 'app', 'boot',
, 'dsptest', 'dspcore', and 'dspapp'.
Example:
rom -method 1
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rom -ftp vwusr vwusr
rom -app -s 192.168.4.101 -f app.bin
usr/config$
Parameter Usages:
-print show versions of all rom files
-app, boot, boot2m, dsptest, dspcore, dspapp, greeting, askpin to
update main Application
program code, Boot code, DSP
testing code, DSP kernel code,
DSP application code, greeting
file, askpin file.
-s to specify TFTP server’s IP address when upgrading ROM
files.
-f to specify the target file name, which will replace the old one.
-method to decide using TFTP or FTP as file transfer server. “0” stands
for TFTP, while “1” stands for FTP.
-ftp if users choose FTP in above item, it is necessary to specify
pre-defined username and password when upgrading files.
4.2.25 [passwd]
For security concern, users have to input the password before entering
configuration mode. “passwd” command is for password setting purpose.
Password setting information and configuration
Usage:
passwd -set Loginname Password
passwd -clean
Note:
1. Loginname can be only 'root' or 'administrator'
2. passwd -clean will clear all passwd stored in flash,
please use it with care.
Example:
passwd -set root Your_Passwd_Setting
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usr/config$
Parameter Usages:
-set
(passwd –set “login name” “password”) Note : “login name” can be “root” or “administrator” only. “root” and
“administrator” have the same authorization, except some
commands that can be executed by “root” only – “passwd –clean”,
“rom –boot”, ”rom –bot2m” and “flash –clean”.
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