MX Series
MX Series
User Manual
http://www.newrocktech.com
Tel: 021-61202700
Fax: 021-61202704
1 Overview
1.1 Product Introduction
MX Series intelligent VoIP Gateways (hereinafter called MX Gateways or simply Gateways) are
designed for bridging the traditional telecom terminal devics into IP networks through SIP or MGCP
protocols. The main applications include:
For carriers and value-added service providers to provide telephone, fax and voice-band data services
to subscribers using IP access methods such as FTTB, HFC, and ADSL;
Used to bridge the traditional telecom terminal equipments, such as PBXs, to the IP core networks of
carriers;
Connected with PBX of enterprises to provide IP-based voice private network solutions for
institutions, enterprises and schools;
Used as remote acces equipments for IP-PBXs in call center deployment.
MX Gateways are suitable for placement on office desktops or installation on walls in the corridor and
racks in the equipment room.
MX Series includes MX8, MX60, MX100 and MX120 subseries. Their features are similar with the main
differences as follows:
Table 1-1 Differences Between MX Gateway Series
application software running cross the full range of MX product series, and ensures the consisten
functions and stable performance in different MX product lines.
MX Gateways support SIP and MCGP protocols. They can provide
PBX functions such as hunting group, second stage dialing, internal communications, caller ID
(FSK/DTMF), call transfer, call waiting, call hold, call barring, caller ID restriction, hotline, corporate
CRBT, three-way calling, ring group, fax and etc;
FXO related functions such as PSTN failover, gain control, busy tone detection, voice prompt in
inbound calls, polarity reversal detection;
Media stream processing functions such as RTP redundancy, packet loss compensation,
G.711/G.729A/G.723.1/iLBC/GSM voice codec, echo cancellation, and etc.
MX Gateways support local and remote, distributed and centralized management modes, including Web
access management, command line configuration based on Linux OS, auto-provision for firmware
upgrade and configuration management based on TFTP/FTP/HTTP, SNMPv2, TR069 based ACS.
Connect analog telephone, PBX, facsimile machine and POS machine to the IP core network, or
PSTN;
Work with service platform to provide various telephone supplementary services;
Support protocols: SIP, MGCP;
Flexible configuration of Phone/Line interfaces;
Support static IP address configuration or dynamically obtain an IP address through DHCP and
PPPoE;
Support G.711, G.729A, G.723.1, GSM, iLBC;
Support echo cancellation;
Up to 500 routing rules can be stored in gateways;
Intercom;
Support digitmap;
Support call progress tones for various countries and regions;
Support Line second stage dialing or voice prompt;
Support PSTN failover through Line ports;
Support High Capacity SD Card (optional, only for MX60)
Security strategy: IP filter, encryption
Support routing table;
Support T.30/T.38 fax mode;
Support polarity inverse detection and busy tone detection;
Support three-way calling;
Compatible with unified communication solutions, such as CallManager, OCS and Asterisk;
Support multiple local and remote maintenance & management modes such as Web, Telnet,
auto-provision, and TR069/TR104/TR106 clinet.
1.3.1 MX8
MX8 is the product with smallest capacity in MX Gateway Series. Designed with small plastic
structure for desktop placement, MX8 can provide up to 8 analog line interfaces. MX8 supports the
following types of configuration:
Table 1-2 Common Configuration Combination of MX8
# Description
① Power indicator (PWR), Light-on indicates that it has been powered.
Steady on indicates valid Ethernet link, flashing indicates Ethernet activities (receiving and/or
②
transmitting)
Analog subscriber line (FXS) or analog trunk (FXO) interface indicator, Light-on indicates that
③
it is in use.
# Description
① Power interface, 5-9 VDC input
② 10/100 M Ethernet Interface, RJ45
③ Analog subscriber line (FXS) or analog trunk (FXO) interface
Table 1-5 Configuration Description of Analog Line Interfaces for All MX8 Models
1.3.2 MX60
Designed with a 1U high and 19’’ wide compact chassis, MX60 is suitable for installation in a standard
cabinet. The interface card of MX60 uses a RJ-45 socket and is connected to the distribution panel in
equipment room using CAT-5 cables supplied with the unit. It has a built-in 110-220V power module.
MX60 offers up to 48 interfaces of FXS/FXO. MX60 supports the following types of configuration.
# Description
①③⑤ Three interface slots; each can correspond with four RJ45 sockets; each RJ45socket can
correspond with four pairs of analog lines. Note: numbers of interface slots vary from different
configuration.
②④⑥ Matrix of 4 x 4 LED status indicator on interface cardl.
Each RJ45 socket has 8 pins leading out 4 pairs of analog telephone or trunk lines in agreement with the
pair specifications for Ethernet interfaces, whose corresponding relations can be seen in the table below.
CAT-5 cables are used to connect the interface card and distribution panel in equipment installation.
Standard RJ11 telephone lines can be used to plug in a RJ45 socket. The telephone/trunk lines are
connected to the 3rd pair of pins for simple call test.
Table 1-8 Pin Specifications for MX60 RJ45 Socket Port
# Description
① Ground Pole
② Indicator, see Table 1-17 for description
③ USB Interface, reserved for future use
④ Configuration interface (CON), Ethnet lines used for local management and debugging
⑤ Two Ethernet interfaces: one IP address
⑥ Cooling fan
⑦ AC power socket, 100-240 VAC voltage input
1.3.3 MX100-AG
Designed with a 1U high and 19” wide compact chassis and a swappable modular structure,
MX100-AG can offer flexible on-site configuration and replacement. The interface card of MX100-AG
uses a Champ50-type socket and is connected to the distribution panel in equipment room using a
25-pair cable supplied with the unit. MX100-AG supports the following types of configuration:
Table 1-11 Configuration Combination of MX100-AG
# Description
① Main control module. It offers one 10/100M Ethernet port and one configuration interface
(CON).
② and ③ Two interface slots and each can contain one desired type interface card.
DO NOT plug or remove the main control module and interface cards of MX100 when equipment is
powered on.
# Description
① Two cooling fans.
② Ground pole.
③ AC power socket, 100-240 VAC voltage input.
1.3.4 MX120 1U
Designed with 1U high and 19’’ wide compact chassis and a swappable modular structure, MX120 can
offer up to 48 analog lines. The interface card of MX120 uses a RJ45 socket and is connected to the
distribution panel in equipment room using CAT-5 cables supplied with the unit.
The device of MX120 1U can hold two interface cards which enable to flexibly configure FXS and
FXO ports. And each card equips up to 24 ports. It supports the following configurations:
Table 1-15 Configuration Combination of MX120
# Description
① and ② Two interface slots; each can contain one 24-port interface card.
③ Matrix of 6×4 LED status indicator on interface card
Do not plug and remove the interface cards of MX120 when equipment is powered on.
Numbering definition of system interface slots: On the left side of main chassis is #1 slot (marked with
No.1 to 24), on the right side of main chassis is #2 slot (marked with No.25 to 48).
Each RJ45 socket has 8 pins leading out 4 pairs of analog telephone or trunk lines in agreement with
the pair specifications for Ethernet interfaces, whose corresponding relations can be seen in the table
below. CAT-5 cables are used to connect the interface card and distribution panel in equipment
installation. Standard RJ11 telephone lines can be used to plug in a RJ45 socket. The telephone/trunk
lines are connected to the 3rd pair of pins for simple call test.
Table 1-17 Pin Specifications for MX120 RJ45 Socket Port
Table 1-18 Corresponding Relation Between MX120 RJ45 Socket and Line Number
There is a 6 × 4 LED indicator matrixes on the left side of interface board. Each row of LED indicator
matrixes matches four telephone lines on a RJ45. The first row on the left matches Line 1-4
respectively from top to bottom, the first row on the right matches Line 21-24 respectively from top to
bottom, and the middle rows in the same manner.
LED indicators are used for multiple purposes as follows
Line status indication: This is the most common mode during normal use of equipment. In this mode,
if a line is idle, the indicator corresponding to it goes off; if a line is in call or in use status (such as
ringing, offhook and caller ID transmission of FXS interface, ringing, offhook and caller ID detection
of FXO interface) the indicator corresponding to it goes on.
Line type indication: This is the mode for cable wiring check when installing the equipment. This
mode can be entered by disconnecting Ethernet cables (Both WAN and LAN ports must be
disconnected) at installation stage. After entering this mode, steady on LED indicates that the
corresponding line is equipped as analog subscriber line type, flashing LED indicates that the
corresponding line is equipped as analog foreigh exchange line type, off LED indicates that the
corresponding line is not equipped or not ready for use.
System operation status indication: This is the mode for displaying information on system operation of
equipment in specific conditions. Usually, this mode is entered when some prompts are required to
give operator during equipment startup, diagnosis or operation. In this mode, LED flashes to display
numbers, letters or other patterns in matrix. Please refer to the Appendix: Check List for Operation
Status Indication of MX120 System.
# Description
① Ground Pole
② Indicator, see Table 1-20 for description.
③ USB interface, reserved for future use.
④ Configuration interface (CON), used for local management and debugging.
⑤ Two Ethernet interfaces: ETH1 and ETH2, only ETH1 has been set when the equipment is
delivered from factory, default IP address: 192.168.2.240
⑥ Cooling fan
⑦ AC power socket, 100-240 VAC voltage input.
1.3.5 MX120 2U
The device of MX120 2U can hold four interface cards which enable to flexibly configure FXS and
FXO ports. And each card equips up to 24 prots. MX120 2U can provide up to 96 ports. It supports the
following configurations:
Table 1-21 Configuration Combination of MX120 2U
# Description
① Matrix of 6×4 LED status indicator on interface card
②③④⑤ Four interface slots; each can contain one 24-port interface card.
Do not plug and remove the interface cards of MX120 when equipment is powered on.
Numbering definition of system interface slots: On the low-left side of chassis is #1 slot (marked with
No.1 to 24), on the low-right side of chassis is #2 slot (marked with No.25 to 48), on the up-left side of
chassis is #3 slot (marked with No.49 to 72), and on the up-right side of chassis is #4 slot (marked with
No.73 to 96).
Each RJ45 socket has 8 pins leading out 4 pairs of analog telephone or trunk lines in agreement with the
pair specifications for Ethernet interfaces, whose corresponding relations can be seen in the table below.
CAT-5 cables are used to connect the interface card and distribution panel in equipment installation.
Standard RJ11 telephone lines can be used to plug in a RJ45 socket. The telephone/trunk lines are
connected to the 3rd pair of pins for simple call test.
Table 1-23 Pin Specifications for MX120 RJ45 Socket Port
Table 1-24 Corresponding Relation Between MX120 RJ45 Socket and Line Number
There is a 6 × 4 LED indicator matrixes on the left side of interface board. Each row of LED indicator
matrixes matches four telephone lines on a RJ45. The first row on the left matches Line 1-4
respectively from top to bottom, the first row on the right matches Line 21-24 respectively from top to
bottom, and the middle rows in the same manner.
# Description
① Indicator, see Table 1-28 for description.
② USB interface, reserved for future use.
③ Configuration interface (CON), used for local management and debugging.
④ Ground Pole
⑤ Two Ethernet interfaces: ETH1 and ETH2, only ETH1 has been set when the equipment is delivered from
factory, default IP address: 192.168.2.240
⑥ Cooling fan
⑦ AC power socket, 100-240 VAC voltage input.
2 Parameter Setting
2.1 Login
MX8 Gateways start DHCP service by default, and automatically obtain an IP address on the LAN;
users can use the factory-default gateway IP address if it is unable to be obtained (e.g. when connected
directly with a computer).
MX60, MX100 and MX120 Gateways use a static IP address by default.
Table 2-1 Default IP Address of Gateway
2.1.2 Logging On
Double-click the icon (or whatever browser you use) to open IE browser, and enter the gateway IP
address in the browser address bar (eg. 192.168.2.218); you can enter the login interface for gateway
configuration by entering a password on the login interface.
Figure 2-1 Login Interface for MX Gateway Configuration
Both Chinese and English Languages are provided for the Web interface.
The administrator can browse and modify all configuration parameters, and modify login passwords.
The operator can browse and modify a subset of the configuration parameters.
The gateways allow multiple users to log in:
If both an administrator and operator have logged in, the administrator may modify the configuration,
while the operator is limited to browsing;
When multiple users with the same level of permission log in, the first may modify, while the others
may only browse.
The system will confirm timeout if users do not conduct any operation within 10 minutes after login.
They are required to log in again for continuing operations.
Upon completion of configuration, click the Logout button to return to the login page, so as not to
affect the login permission of other users.
2.3.1 Network
After login, click Basic > Network tab to open the configuration interface.
Name Description
Host name This is the equipment name of the gateway. The default values of MX8, MX60, MX100 and
MX120 are MX8-VoIP-AG, MX60-VoIP-AG, MX100-VoIP-AG and MX120-VoIP-AG
respectively. Users can set a different name for each gateway when multiple gateways are on the
same subnet.
A host name can be a maximum of 48 characters, either letters (A-Z or a-z), numbers (0-9) and
minus sign (-). It may not include a null or space, and it must start with a letter.
Logical IP address This parameter only exists in MX100, used to display the actual gateway IP address in use.
ETH1
MAC address Display the MAC address of the gateway.
IP address Methods for obtaining an IP address
assignment
Fixed: Static IP address is used;
DHCP: Use the dynamic host configuration protocol (DHCP) to allocate IP addresses and other
network parameters;
PPPoE: PPPoE service is used.
User name Enter an authentication user name if PPPoE service is selected, and there is no default value.
Password Enter an authentication password if PPPoE service is selected, and there is no default value.
IP address If “Fixed” or “DHCP” is selected for the network type but an address fails to be obtained, the
gateways will use the IP address filled in here. If the gateways obtain an IP address through
DHCP, the system will display the current IP address automatically obtained from DHCP. This
parameter must be set due to no default value.
Netmask The subnet mask is used with an IP address. When the gateways uses a static IP address, this
parameter must be entered; when an IP address is automatically obtained through DHCP, the
system will display the subnet mask automatically obtained by DHCP. This parameter must be set
due to no default value.
Gateway IP The IP address of LAN gateway. When the gateways obtain an IP address through DHCP, the
address system will display the LAN gateway address automatically obtained through DHCP. This
parameter must be set due to no default value.
DNS
Enable Activate DNS service.
Primary Server If DNS service is activated, the network IP address of the preferred DNS server must be entered,
and there is no default value.
Name Description
Secondary Server If DNS service is activated, the network IP address of a standby DNS server can be entered here.
It is optional and there is no default value.
SNTP
Primary Server Enter the IP address of preferred time server here. This parameter must be set due to no default
value.
Secondary Server Enter the IP address of standby time server here. This parameter must be set due to no default
value.
Time Zone Select a time zone, and the parameter values include:
(GMT-11:00) Midway Island
(GMT-10:00) Honolulu. Hawaii
(GMT-09:00) Anchorage, Alaska
(GMT-08:00) Tijuana
(GMT-06:00) Denver
(GMT-06:00) Mexico City
(GMT-05:00) Indianapolis
(GMT-04:00) Glace_Bay
(GMT-04:00) South Georgia
(GMT-03:30) Newfoundland
(GMT-03:00) Buenos Aires
(GMT-02:00) Cape_Verde
(GMT) London
(GMT+01:00) Amsterdam
(GMT+02:00) Cairo
(GMT+03:00) Moscow
(GMT+03:30) Teheran
(GMT+04:00) Muscat
(GMT+04:30) Kabul
(GMT+05:30) Calcutta
(GMT+05:00) Karachi
(GMT+06:00) Almaty
(GMT+07:00) Bangkok
(GMT+08:00) Beijing
(GMT+09:00) Tokyo
(GMT+10:00) Canberra
(GMT+10:00) Adelaide
(GMT+11:00) Magadan
(GMT+12:00) Auckland
After login, click Basic > System tab to open the configuration interface.
Name Description
First digit timer If a subscriber hasn’t dialed any number within a specified time by this parameter after offhook,
the gateways will consider that the subscriber has given up the call and prompt to hang up in
busy tone. Unit: second; Default value: 12 seconds.
Inter-digit timer If a subscriber hasn’t dialed the next number key from the time of dialing the last number key to
the set time by this parameter, the gateways will consider that the subscriber has ended dial-up
and call out the dialed number. Unit: second; Default value: 12 seconds.
Critical digit timer This parameter is used with the "x.T" rule set in dialing rules. For example, there is "021.T" in
the dialing rules table. When a subscriber has dialed 021 and hasn’t dialed the next number
within a set time by this parameter (eg. 5 seconds), the gateways will consider that the
subscriber has ended dial-up and call out the dialed number 021.
Input intergers, not decimal fractions Unit: second; Default value: 5 seconds.
Codec Codecs supported by MX include G729A/20, G723/30, PCMU/20, PCMA/20, iLBC/30 and
GSM/20. This parameter must be set due to no default value.
Several encoding methods can be configured in this item at the same time, separated with “,” in
the middle; the gateways will negotiate with the platform in the order from front to back when
configuring the codec methods.
Hook-flash handle The gateways provide the following processing modes after detecting hook flash from
subscriber terminals:
Internal: the hook flash event will be handled internally;
Server(RFC 2833): transmitting the hook flash to platform with RFC 2833;
Server (SIP INFO): transmitting the flash-off to platform with SIP INFO.
DTMF method Transmission modes of DTMF signal supported by the gateways include RFC 2833, Audio and
SIP INFO. The default value is Audio.
RFC 2833: Separate DTMF signal from sessions and transmit it to the platform through RTP
data package in the format of RFC2833;
Audio: DTMF signal is transmitted to the platform with sessions;
SIP INFO: Separate DTMF signal from sessions and transmit it to the platform in the form of
SIP INFO messages.
2833 payload type Used with “RFC 2833” in the DTMF transmission modes. The default value of 2833 payload
type is 100. The effective range available: 96 ~ 127. This parameter should match the setting of
far-end device (eg. platform).
DTMF on-time This parameter sets the on time (in ms) of DTMF signal sent from Line port. The default value
is 100 ms. The duration time range is 80 ~ 150 ms.
Name Description
DTMF off-time This parameter sets the off time (ms) of DTMF signal sent from Line port. The default value is
100 ms. The interval time range is 80 ~ 150 ms.
DTMF detection Minimum duration time of effective DTMF signal. Its effective range is 32-96 ms. The default
threshold value is 48 ms .The greater the value is set, the more stringent the detection is.
DTMF detection Increase the value can prevent false detection of DTMF signal. The valid values are 16, 32, and
adjust 48 in million seconds.
Codec Supported by MX Bit Rate (Kbit/s) Time Intervals of RTP Package Sending (ms)
iLBC 13.3/15.2 20/30
GSM 13 20
G729A 8 10/20/30/40
G723 5.3/6.3 30/60
PCMU/PCMA 64 10/20/30/40
After login, click Basic > SIP tab to open the SIP configuration interface.
Figure 2-4 SIP Configuration Interace
Name Description
Signaling port Configure the UDP port for transmitting and receiving SIP messages, with its default value
5060.
Note: The signaling port number can be set in the range of 1-9999, but cannot conflict with
the other port numbers used by the equipment.
Auto SIP port selection If “n”(ranked from 1-10) is chosen, after the failure registration of signaling port’s original
configuration, the range of signaling port’s change varies from “original signaling port,
original signaling port +n”. Register with the new signaling port value (signaling port +1)
until it succeeds.
Register server Configure the address and port number of the SIP registration server.The address and port
number are separated by “:”. It has no default value.
The register server address can be an IP address or a domain name. When a domain name is
used, you must activate DNS service and configure DNS server parameters on the
network-configuration page.
Name Description
Proxy server Configure the IP address and port number of the SIP proxy server. The address and port
number are separated by “:”. There is no default value.
The proxy server address can be set to an IP address or a domain name. When a domain
name is used, you must activate DNS service and configure DNS server parameters on the
network-configuration page. For example: 168.33.134.50:5060 or www.sip.com: 5060.
Backup proxy server By specifying the corresponding IP addresses, the gateway can be configured to have
multiple soft switches as backup proxy servers. Make sure that the IP addresses are in their
full format.
E.g. 168.33.134.53:5060. The proxy and register severs must be identical.
Conditions for failing over to the backup proxy server (any):
Gateway register is timedout
No response to master server calls timedout
User agent domain This domain name will be used in INVITE messages. If it is not set here, the gateways will
name use the IP address or domain name of the proxy server as the user-agent domain name. It has
no default value.
It is recommended that subscribers not use LAN IP address to set the domain name
parameter.
Authentication mode The gateway supports three registration schemes: register per line, register per gateway and
Line Reg/GW Auth. The default value is register by line.
Register by line: authentication and register per line;
Register by gateway: authentication and register per gateway;
Line Reg/GW Auth: register per line, but authentication per gateway.
User name Configure the user name as part of the account for registration, and it has no default value.
Note: If “Register by gateway” or “Line Reg/GW Auth”, is selected, the user name must be
entered here. If “register by line” is selected the user name should be set on “Line > Feature”
page (Refer to “Feature”).
Password Password as part of account information is used for authentication by platform. It has no
default value. It is formed with either numbers or characters, and case sensitive.
Note: If “Register by gateway” or “Line Reg/GW Auth”, is selected, the password must be
entered here. If “register by line” is selected the password should be set on “Line > Feature”
page (Refer to “Feature”).
Registration period Valid time of SIP re-registration in seconds. Its default value 3600.
The gateways use SIP protocol by default. When the gateways need to interface with MGCP protocol
-based softswitch platform, set the relevant parameters here.
After login, click Basic > MGCP tab to open the configuration interface.
Name Description
Signaling port Configure the UDP port for transmitting and receiving MGCP messages, and the default value is
2427.
Note: The signaling port number can be set in the range of 1-9999, but cannot conflict with the
other port numbers used by the equipment.
Proxy server Configure the IP address and port number of MGCP proxy server, separated by “:”, and it has no
default value.
The address can be set to an IP address or a domain name according to the subscribers’
requirements. When a domain name is used, it is required to activate DNS service and configure
DNS server on the page of configuring network parameters. Examples of complete and effective
configuration: 46.33.136.50:2727 or www.proxy.com: 2727.
User agent domain The domain name associated with the call agent, and it has no default value.
name Example: www.gatewaymgcp.com.
Default event List all the types of default event packages supported by the HX4. Multiple package names are
package separated by“,”.
The default value is L, D, G
L: Line Package
D: DTMF Package
G: Generic Media Package
Persistent line event List the event types that the gateway can report, with multiple types separated by “,”. When
gateways process the events listed here, they will report to the call agent.
Note: This parameter must be set since there is no default value. The factory setting is L/HD,
L/HU:
L/HD: Offhook
L/HU: Onhook
Fxo event package Handset Package
Line Package
Wildcard Select whether a wildcard with prefix is allowed when a gateway registers to the proxy server.
The default value is not allowed.
Partially allowed: Gateways will use a wildcard with fixed prefix (e.g. aaln / *) when
registering. For example, when configuring telephone numbers, if line 1 is set to
aaln/1, line 2 is set to aaln/2 and line 3 is set to aaln/3, the gateways will register to
the call agent in aaln/* without the need of registering the lines individually.
Allowed: the gateways will use a wildcard in registering without prefix.
Name Description
Compatibility
Configuration
CR for End-of-Line Select whether CR is used as the end of line in the MGCP messages. Default not selected.
Quarantine default Select the Quarantine handle of gateways making a request to the outside, and default not
to loop selected.
Selected: Quarantine using loop mode, the gateways will continually notify all
events as requested after receiving a request.
Enable first digit Select the processing mode when there is no timeout parameter in the outside request received
timer by the gateways, and default not selected.
Selected: the gateways will report timeout in terms of its own timeout setting (the
time interval set in non-dial timeout of configuration system parameters) when
subscribers hasn’t dialed up in time after offhook.
Using configured Select whether to activate the digit map configured by local gateway, and default value is not
digit map selected.
Using notify instead Set whether the gateways report “offhook events” to replace 401 messages in NTFY or report
of 401/402 “onhook events” to replace 402 messages in NTFY when responding to messages sent by the
proxy server. Default: not selected.
Selected: The gateways will use NTFY message to replace 401 and 402 messages.
No name in default Select if a package name is included when the gateways reply to the default package, and default
package not selected.
Keep connection Select if the gateways actively cancel connection disconnect when subscribers hook on, and
when on-hook default not selected.
2.3.5 FoIP
After login, click the label of Basic > FoIP to open this interface.
Figure 2-6 Fax configuration interface
Name Description
Transparent and T.30 Transparent
T.30
T.38 T.38 with CED
T.38 with CNG
T.30 and T.38 Both T.30 and T.38
Jitter buffer Set the extent of T.38 jitter buffer, and the default is 250. The valid range is 0~1000 in
milliseconds.
Receiving port for FoIP Set whether to open a new port when the gateway is switching to T.38 mode, and by
default, original voice port will be usd.
Open a new port: use the new RTP port.
Use the port of the original voice call: use the original RTP port that created on call set.
ECM Determine whether to use corrective mode of fax. By default, it is not selected.
Receive gain Set the receiving gain of T.38 fax, with the default of 6dB.
Transmit gain Set the transmission gain of T.38 fax, with the default of 0dB.
Packet size Set the packet size of T.38. 30 miliseconds is the default value.
Redundancy Set the number of the redundant frames in T.38 date package, default is 4.
2.4 Routing
After login, click Routing > Digit Map tab to open the dialing rules interface.
Figure 2-7 Configuration Interface for Dailing
Dialing rules are used to effectively judge if the received number sequence is completed, for the
purpose of terminating receiving numbers and sending received numbers. The proper use of dialing
rules can help to reduce the connection time of telephone calls.
The maximum number of rules that can be stored in gateways is 250. Each rule can hold up to 32
numbers and 38 characters. The total length of dialing rules table (the total length of all dialing rules)
can be up to 2280 bytes.
The following provides descriptions of typical rules:
400xxxxxxx
x.T
x.#
#xx
*xx
##
After login, click Routing > Routing Table tab to open the configuration interface.
Figure 2-8 Configuration Interface for Routing Table
The routing table with 500 rules in capacity provides two functions including digit transformation and
call routing assignment. Here are the general rules applied by gateways when executing the routing
table.
Rules must be filled out without any blank at the beginning of each line; otherwise the data can’t be
validated even if the system prompts successful submittal.
The routing table is empty by default. The gateways will point a call to the SIP proxy server when
there is no matched rule for the call.
Name Description
Source There are three types of source: IP, FXS (Phone/fax) and FXO (Line).
Among them, IP source can be any IP address and is denoted by IP; IP [xxx.xxx.xxx.xxx] is used
to denote a specific IP address; IP [xxx.xxx.xxx.xxx: port] is used to denote specific IP address
with port number.
FXS(Phone) and FXO(Line) ports can be any port, represented with FXS or FXO; special lines
can be represented with FXS or FXO plus the port number, e.g. FXS1, FXO2 or FXS [1-2], etc.
Number It could be a calling party number with the form of CPN + number,such as CPN6034340633 or a
called party number with the form of number. The number may be denoted with digit
0-9,"*",".","#"," x ", etc., and uses the same regular expression as that of dialing rules. Here are
examples of the form of number:
Designate a specific number: eg.114,61202700
Designate a number matching a prefix: such as 61xxxxxx. Note: the matching effect
of 61xxxxxx is different from that of 61x or 61. Number matching follows the
principle of minimum priority matching
Specify a number scope. For example, 268[0-1,3-9] specifies any 4-digit number
starting with 268 and followed by a digit between 0-1or 3-9
Note: Number matching follows the principle of minimum matching. For example: x matches any
number with at least one digit; xx matches any number with at least two-digit; 12x matches any
number with at least 3-digit starting with 12.
Example 2:
IP 800[0-3] ROUTE FXS 1
Point this call to FXS (Phone) port 1.
Example 3:
IP 800[0-3] ROUTE FXS 1-2/R
Select a port in round robin order
Example 4:
IP 800[0-3] ROUTE FXS 1-2/G
Select all idle ports and provide ringing.
ROUTE FXO Route a call to FXO (Line) port.
Example 1:
IP x ROUTE FXO 1-2
Select a port in sequential order.
Example 2:
IP 800[0-1] ROUTE FXO 1-2/R
Select a port in round robin order.
ROUTE IP Route a call to the SIP proxy server
Example:
FXS 021 ROUTE IP 228.167.22.34:5060
228.167.22.34:5060 is the IP address of the platform.
Some typical functions that can be realized by the routing table are provided in this section (Take
MX8-4S/4 gateway as an example):
1) One Phone with Two Numbers
2) Hunt Group
3) Outbound Call Barring
4) FXO (Line) Port Hunting for Outbound Call
Hunt Group
A hunt group can be associated with a set of FXO (Line) ports, and an inbound call from IP or FXS
(Phone) ports can be routed to a hunt group.
Routing Setting:
Send an inbound call from the IP trunk or an FXO line in a sequential way to the phone set on the 1st or
2nd FXS (Phone) port.
FXO x ROUTE IP 127.0.0.1:5060
IP x ROUTE FXS 1-2
Description:
1) Send all calls from the FXO (Line) port to port 5060 of gateway’s local IP;
2) Send all inbound calls from any IP (inside and outside) to the 1st or 2nd FXS (Phone) port in
sequence. Namely, the first FXS (Phone) port is selected firstly when it is available, otherwise the
2nd port is selected.
2.4.4 IP Table
After login, click Routing > IP Table tab to open the configuration interface.
Figure 2-9 Configuration Interface for IP Table
This table is designed to ensure the safe use of gateways. Administrators can add the authorized IP
addresses to this table, and the gateways will only process the information from authorized IP addresses.
If the IP table is empty, the gateways will not perform IP address-based message filtering.
For example: The gateway will only process the messages from 202.96.209.133 after adding
202.96.209.133 to its IP table.
After login, click Line > FXS phone number tab to open the configuration interface.
Name Description
FXS 1st line No. This number is used for the batch setup of consecutive number of subscriber line. Click Batch after
filling in initial number, the number of Line 1 adopts initial number; that of Line 2 increases 1
progressively based on that of Line 1, and so on. You needn’t fill in if you do not use batch
configuration or the number is not consecutive.
ID n Fill in the telephone number associated with the subscriber line n (FXS port). This should be
manually performed if Batch mode is not used.
After login, click Line > FXO phone number tab to open the configuration interface.
Figure 2-11 Configuration Interface for FXO phone number
Name Description
FXO 1st line This number is used for the fast setup of consecutive number of trunk line. Click Batch after filling
No. in initial number, the number of Line 1 adopts initial number; that of Line 2 increases 1
progressively based on that of Line 1, and so on. You needn’t fill in if you do not use batch
configuration or the number is not consecutive.
ID n Fill in the telephone number associated with the trunk n (FXO port). This should be manually
performed if Batch mode is not used.
This page is only used for configuring gateways with subscriber lines (FXS port).
After login, click Line > Feature tab to open the configuration interface.
Name Description
Phone number Fill in the phone number associated with this port.
Display name Fill in the name associated with this port.
Registration Select if this line is required to register with a softswitch. This is selected as default.
Password If Registration is selected, users must enter the authentication password for registerion of this line
here.
Note: The following features are valid only in SIP protocol. When the gateways use MGCP protocol,
features are controled by the proxy server without the need for setting on the gateway.
Hot line Select if the gateway is required to automatically dial out the hotline number after offhook. By
default, hot line is disabled.
Disable hot line: Close this feature.
Hot line: Automatically dial out the hotline number after offhook.
Delay mode: Automatically dial out the hotline number when the offhook is timeout
with a time delay of 5 seconds.
Hot line number After the hotline function is activated on this line, the hotline number must be entered here.
CRBT(Color CRBT stands for Color Ring Back Tone. Set if CRBT is activated, that is, provide prepared audio
ring back tone) package as ring back tone. Note: it is required to set the CRBT file download platform. This is not
selected by default.
MX supports two CRBT storage modes: on-system (stored in the flash memory) and run-time
download (from FTP server). The length of tone in both modes are described as follows:
On sytem:
MX8: No more than 20 seconds in G.729 format (fring1.dat)
MX100: Up to 20 tone files, a maximum of 400 seconds in G.729 format or 50 seconds in G.711
format.
Run-time download:
MX8/MX100: Up to 20 tone files, a maximum of 10000 seconds in G.729 format or 1250
seconds in G.711 format.
Note:Tone files are stored in the flash memory and the gateway automatically download the tone
files from FTP server after power on.
Speed dials Select if the Speed dials is activated on this line. By default, this is not selected.
Name Description
Call forwarding Select if Call forwarding is activated on this line. By default, it is not selected.
Forking Select to activate Forking. Forking allows the gateway to initiate a call to another telephone terminal
while ringing on this line terminal. Either terminal may answer, terminating ringing on the other
terminal.
Release control Select if the call release is controlled by the caller. By default, this is not selected. Note: Also see
by caller Caller release on page Advanced > Line.
Selected: The gateway will immediately release the call upon caller hanging up; the
gateway will not release the call after the called party hanging up as long as the caller is
still off-hook until timeout (60 seconds by default);
Unselected: The gateway will immediately release the call upon either party hanging up
the call.
Call waiting Select if Call waiting is activated on this line. By default this is not selected.
Call hold Select if Call Hold is activated on this line. By default this is not selected.
Note: If this function is activated, the gateways will automatically activate Call Transfer (Either
party may transfer the current call to a third party).
Caller transfer Select if Caller Transfer is activated on this line. By default, this is not selected. When A calls B, B
picks up the call and transfers the call to C.
Note: The call hold must be activated before caller transfer.
Caller ID Set whether the number of this telephone is sent to the called party. This feature requires the support
display of softswtich. By default this is not selected.
Caller ID Set whether the number of this telephone is sent to the called party with support from platform. By
restriction default this is not selected.
Outgoing call Select if outgoing calls are barred on this line. By default, this is not selected.
barring
DND(Do not Select if Do Not Disturb is enabled on this line. By default, this is not selected.
disturb)
DDI (Direct Set whether DDI (Direct Dialing In) is activated, By default this is not selected. Different from FXS,
Dialing in) DDI is only used for incoming calls, and the gateways will not send dial tone after off-hook (calling
in) on user side.
Note: Reverse polarity signal must be activated on the gateways when DDI is used.
Polarity reversal Select if reverse polarity signal is activated on this line. By default, this is not selected.
Note: The gateways will provide reverse polarity signal when the phone is connected after this
feature is activated.
Maintenance Select if the line is set to maintenance status, in which the FXS port not longer supplies current to
the phone. By default, this is not selected.
Subscribe MWI Select if voice mail service is activated, and by default this is not selected. (Also see MWI
Re-subscription timer on page Advanced > SIP.)
3-way Select if 3-way service is activated, and by default this is not selected.
This page is only used for configuring gateways with trunks (FXO port).
After login, click Line > Trunk tab to open the configuration interface..
Name Description
Trunk ID Select a trunk line required to configure. “FXO-n” corresponds to the “Line > FXO phone
Number ID n”. Copy the configuration of “FXO-n” for selected line to “FXO-n+1”~“FXO- m”
by clicking “Batch”, where n indicates the current selected trunk number and m indicates the
total number of trunks.
Phone number Display phone number associated with the trunk set in Line > FXO phone Number
Registration Select if this trunk registers with the SIP registration server. By default, this is selected.
Password If Registration is selected, the authentication password for register of this line must be entered
here.
Note: The following features are valid only in SIP protocol. When the gateways use MGCP protocol, the
control of all call services is provided by the proxy server without the need of these setting.
Inbound handle The gateways provide three scenarios for handling incoming calls on the FXO turnk Line ports
(Line Port):
Binding: When a telephone call comes to the Line port, the gateways will route the
call to a Phone port according to the DID number bound with the port. Note: Setting
a number to be bound is required or this setting is invalid.
Second-stage dialing: When a telephone call comes to the Line port, the gateways
will provide the second dial tone and route the call according to the extension
number entered. Note: dialing tone or voice prompt file can be changed by user.
Direct:The gateways will route the incoming call on FXO port n to FXS port n.
Polarity reversal If a PSTN line supports reverse polarity, make the selection here. By default, this is not selected.
detection
Caller ID detection Select if the detection function of caller ID for this Line port is enabled. By default, this is
selected.
Outbound blocking Select if this Line port bars outgoing call service to the PSTN. By default, this is not selected.
Echo cancellation Select if echo cancellation is enabled for this FXO (Line).By default, this is selected.
Connect signal After making an outgoing call from a Line port, the gateway will send a 200 OK message to the
delay platform with a delay if this parameter is selected. If unselected, the system sends a 200 OK
message to the platform after off hook on the Line port. Also see Answer delay on page
Advanced > Trunk.
Click , the following interface is shown. Choose batch configured features and click “ok”.
Click to choose whether to activate this function to configurate this parameter. Seen in Subscriber
Line Features.
Click , the following interface is shown. Choose batch configured trunks and click OK.
Click to choose whether to activate this function to configurate this parameter. Seen in Trunk
Line Features.
After login, click the label of Advanced > System to open this interface.
Figure 2-16 Inferface of system advanced configuraiton
Title Explanation
NAT
NAT traversal Gateways support several mechanisms for NAT traversal. Usually, static NAT is used when a
fixed public IP address is available. It’s necessary to perform port mapping or DMZ function on
router when choosing dynamic or static NAT.
Refresh period The refresh time must be filled in here when choosing dynamic NAT or STUN traversal. Refresh
time interval shall be determined by giving consideration to the NAT refresh time of the LAN
router where the gateway is located. Gateway’s NAT holding function and STUN function will
carry out periodic operation according to this parameter. With seconds as its unit, default value
of 60 seconds.
SDP Address This parameter determines the IP address used in transmitted SDP.
NAT IP Address: Apply NAT address into the transmitted SDP;
Local IP Address: Apply the gateway’s IP address into the transmitted SDP.
Note: The parameter should come into effect only on condition that gateway successfully
obtained NAT address.
Title Explanation
Remote
management
Remote TRO69
management
Enable
The gateways support EMS which is a centralized gateway management server provided by New
Rock, and Auto-provision.
Server Gateways may download software upgrade packages and configuration files automatically
through auto-provision server. Once the auto provision is selected, you must enter the IP address
of ACS here.
After login, click the label of Advanced > Media Stream to open this interface.
Figure 2-17 Media stream configuration interface
Title Explanation
Min. RTP port The lowest port number of UDP ports for RTP transmission and receiving.The parameter must
be greater than or equal to 3000. This ia a required field.
Note: each phone call will occupy RTP and RTCP ports. If the gateway is equipped with 4
subscriber lines (or trunk line), then at least 8 UDP ports are needed.
Max. RTP port The highest port number of UDP ports for RTP’s transmission and receiving.
This is a required field. The value must be greater than or equal to “2× number of lines+min.
RPT port”.
iLBC payload type Set the RTP payload type of iLBC,and the default value is 97. Accepted value is 97 ~ 127. The
parameter shall be configured in conformity to that of platform.
G.723.1 rate Set G.723.1 coding rate,the default value is 6300(bit/s). The optional parameters are
followings:
5300(bit/s): the Bit rate is 5.3k per second;
6300(bit/s): the Bit rate is 6.3k per second
TOS bits This parameter specifies the quality assurance of services with different priorities. The default
value is 0x0C. E.g: TOS=0xB8 indicates level 5 that has no reliability requirement.
Min. Jitter buffer RTP Jitter Buffer is constructed to reduce the influence brought by network jitter. This default
value is 3.
Title Explanation
Max. Jitter buffer RTP Jitter Buffer helps to reduce the influence brought by network jitter. The default value is
50.
RTP drop SID Determine whether to discard received RTP SID voice packets. By default, SID voice packets
will not be dropped.
Note: RTP SID packets should be dropped only when they are in nonconformity to the
specifications. Nonstandard RTP SID data could generate noise for calls.
Enable VAD Only applicable to G.723, GSM, iLBC. In case of selecting this parameter, it will not send any
voice packet during mute period. By default, this is selected.
RTP destination This parameter determines where to obtain the IP address of the receiving side for RTP packets.
address By default, the IP address is obtained From SDP global connection.
From SDP global connection: Obtain the IP address from SDP global connection;
From SDP media connection: Obtain the IP address from SDP Media Description.
SIP messages consist of request message and response message. Both include a SIP message-header
field and SIP message-body field. The SIP message header mainly describes the message sender and
receiver; SIP message body mainly describes the specific implementation method of the dialog.
Message of request: The SIP message sent by a client to the server, for the purpose of activating the
given operation, including INVITE, ACK, BYE, CANCEL, OPTION and UPDATE etc.
Message of response: The SIP message sent by a server to the client as response to the request, including
1xx, 2xx, 3xx, 4xx, 5xx, and 6xx responses.
Message header: Call-ID.
Parameter line: Via, From, To, Contact, Csq, Content-length, Max-forward, Content-type, White Space,
and SDP etc.
MX gateways provide good flexibility in content setting in order to improve compatibility with the SIP
register server.
After login, click the label of Advanced > SIP to open this interface.
Title Explanation
SIP related
configuration
MWI The default is 86400 seconds. The gateway will send the platform a message to confirm that it
has subscribed to MWI service at intervals of the time period set here. This parameter should be
Re-subscription used in conjunction with voice mail subscription on the page of the subject subscriber line.
timer
PRACK Determine whether to activate Reliable Provisional Responses. (RFC 3262)
Session timer Choose to activate session refresh (RFC 4028). By default, session timer is not activated.
Session interval Set the session refresh interval, the gateway will enclose the value of Session-Expires into
INVITE or UPDATE messages. Default value is 1800 seconds.
Minimum timer Set the minimum value of session refresh interval.
Request/Response
configure(SIP
header)
Contact field in Choose the registration mode of gateway under LAN traversal circumstance, the default is NAT
REGISTER IP Address.
NAT IP address: Use the NAT information returned by registration server.
LAN IP address: Keep original content of Contact when register;
Domain name in The default is Domain name.
REGISTER
Domain name: Complete domain name used for registration (for example:
[email protected]);
Sub domain name: Only use the common part of the name of domain (for example:
[email protected]).
Via field Choose whether to use NAT IP address or LAN IP address for Via header field value, the
default is NAT IP address.
To field Choose whether to apply Sub domain name or Outbound proxy to To header field, the default is
Sub domain name.
Address in Call ID Choose whether to fill Call ID field with Host name or Local IP address, the default is Local IP
field address.
Title Explanation
Called party number Choose whether the gateway acquires the called number from Request Line header field or To
header field. The default is From Request line field.
Calling party number Under call forwarding, the calling party number sent can be chosen from Originating number or
in call transfer Forwarding number being set for sending, the default is Forwarding number.
For example: the subscriber line 2551111 on the gateway activates call forwarding feature and
set the destination to 3224422. When caller with 13055553333 calls 2551111, the call will be
forwarded to 3224422:
if Originating number is chosen,the number 13055553333 will be sent to 3224422
as calling party number;
if Forwarding number is chosen,the number 2551111 will be sent to 3224422 as
calling party number.
Do not validate Via Set whether to ignore Via field, By default, Via is ignored.
Register upon invite Set whether to activate registration when SIP message of INVITE is failed or time expired, and
timeout by default, re-registration is not selected.
Selecting the Use the receiving port of proxy or use the sending port of proxy.
receiving port for
response
After login, click the label of Advanced > Line to open this interface.
Figure 2-19 Subscriber-line characteristics configuration interface
Title Explanation
Gain to IP Set the voice volume gain toward the IP side, the default is 0. Taking decibel as the unit, setting
range is -3 ~ +3 decibels. -3 means declining of 3 decibels; +3 denotes the amplification of 3
decibels.
Title Explanation
Gain to terminal Set the voice volume gain toward Phone port side, the default is -3. Taking decibel as the unit,
setting range is -6 ~ +3 decibels. -3 means declining of 3 decibels; +3 denotes the amplification
of 3 decibels.
Impedance Select the parameter of FXS (Phone) port line impedance and the default value is 600 ohm. The
optional values as below:
Complex
600(ohm)
900(ohm)
Min.hookflash Used by the gateway to detect Hook Flash event, the default is 75 milliseconds. The
gateway will ignore any flash that fall short of the shortest flash time. Generally, this
value should not be less than 75 milliseconds.
Max.hookflash Used by gateway to detect hook flash, the default is 800 milliseconds.
The gateway will regard the flash duration between Min.hookflash and Max.hookflash as
effective flash. Any flash lasting over the longest time will be considered by gateway as hang
up. Generally, this value should not be less than 800 milliseconds.
Hook debouncing Used by gateway to avoid a glitch of the phone status, with default of 50 milliseconds.
When the duration from hang-up to off-hook falls short of this value, the gateway will
ignore the status variation, and consider that the phone remains in hang-up status. In
opposite case, the gateway will ignore the status variation, and consider the phone
remains in off-hook status. Effective range of setting is 10~1000 milliseconds.
Ring frequency Set the ringing frequency to be transmitted by gateway to the phone, ranging from 15 to 50 Hz,
with default of 20 Hz.
Caller release Set the delay release time of line as caller control method, with default of 60 seconds. Effective
range of setting is 15~180 seconds.
Outpulsing delay Used when gateways’ FXS (Phone) port is connected with the trunk interface of PBXs. For calls
from gateway to PBX, gateways will relay the extensions to PBX after the delay set here. Setting
of 0 means no extension number relay. The default is 0 milliseconds.
Polarity reversal Set the trigger for polarity reversal the default is Outgoing.
Outgoing: Transmit reverse polarity signal only when the outbound is connected;
Bi-direction: Transmit reverse polarity signal for the connection of both inbound
and out bound calls.
Polarity reversal The delay time from a call being answered to the transmission of reverse polarity signal. The
delay default value is 3 in seconds. Effective range of setting is 0 ~ 30 seconds.
Call ID transmit Select transmission mode of Caller ID signal from the FXS (Phone) port to the phone.
FSK or DTMF
SDMF or MDMF
Sending Caller ID data before or after ringing
Sending Caller ID data with or without parity
Music on hold Choose whether to play the background music while call waiting, and the default is not to play.
Call waiting with Choose whether to activate hunt group feature for call waiting, Default not selected.
hunt group
Message waiting Choose the lighting method of message waiting indicator of voice mail here: None, Polarity
light reversed, FSK. Message waiting indicator refers to the special LED on a phone,
working with voice mail function. When user receives a voice message. The gateway
will light this lamp upon receiving the notice from platform; the light goes off when
there’s no unheard mail. It’s essential to understand whether the phone supports the
indicators and lighting method when selecting the lighting method.
Distinctive
Alert/Ringing
Alert-Info 1 To match with User-Ring 1. Four patterns of user ring are offered. When the Alert-info value of
INVITE message matches with this parameter, User-Ring 1 is activated.
User-Ring 1 Configure user ring 1.
E.g 1: if the user ring is set 2, 500, 500, 1000, 3000, the ringing cadence is 0.5s on, 0.5s off; 1s
on, 3s off.
E.g 2: if the user ring is set 2000, 4000, the ringing cadence will be 2s on, 4s off.
Alert-Info 2 To match with User-Ring 2
User-Ring 2 Configure user ring 2
Alert-Info 3 To match with User-Ring3
User-Ring 3 Configure user ring 3
Alert-Info 4 To match with User-Ring 4
User-Ring 4 Configure user ring 4
After login, click the label of Advanced > Trunk to open this interface.
Figure 2-20 Trunk line characteristics configuraiton interface
Title Explanation
Gain to IP Set the voice volume gain toward IP side, the default is 0. Taking decibel as the unit, setting
range is -3 ~ +9 decibels. -3 means declining of 3 decibels; +3 denotes the amplification of 3
decibels.
Gain to PSTN Set the voice volume gain toward PSTN side, the default is -3. Taking decibel as the unit,
setting range is -6 ~ +9 decibels.
Title Explanation
Impedance Set the parameter of FXO (Line) impedance, with the default of 600 ohm. The optional settings
are below:
Complex
600(ohm)
900(ohm)
Outplusing delay Set the time interval between the FXO (Line) going off-hook and starting outpulsing of the first
digit to the PSTN. The default is 600 in milliseconds.
Ring relay Whether to relay the ring of inbound call to the FXS (Phone) port when applying to DID. The
default is Phone ring independently.
Busy line handle Either a voice prompt or hanging up can be applied to FXO (Line) port when an incoming call
goes to the FXS (Phone) port which is in busy. This only applies to DID feature.
PSTN failover Whether to route a call to the PSTN through an FXO (Line) port when the IP network faults or
no response to the call request. Default selected.
Caller ID detection Before ringing
mode.
After ringing
Inbound first digit Set the timeout of calling DTMF on FXO (Line) port for inbound calls, ranging from 10-60
timeout seconds, with default of 24 seconds.
Answer delay Set the delay time of outbound connection ranging from 10-60 seconds, with default of 12
seconds. Also see Delay sending 200OK on page Phone/Line > Line.
Off-hook for Used for binding an FXO (Line) port with an FXS (Phone) port. For inbound calls to an FXO
rejection (Line) port, if the associated FXS (Phone) port is busy, the gateway will hang up after off hook
according to the time set by the parameter, so as to refuse the upcoming call. The duration of the
off hook is 500~5000 milliseconds, with a default of 600 milliseconds.
On-hook protection Protection period following hang up of FXO (Line) port. During this period, gateway ignores
time any voltage variation of line. Value range is 100~5000 milliseconds, the default is 400 in
milliseconds.
Polarity detection. Choose whether to activate the detection of reverse polarity signal of FXO (Line) port.
Note the detection will work only when the trunk supports polarity reversal.
Busy detection
Repeat Gateways will regard the busy tone signal with the repeat times specified here as a hang-up
signal. Default is 2, effective range is 2 ~ 5(cycle).
On-time Set duration of busy tone signal, the default is 350 in milliseconds.
Off-time Set the interval time of busy tone, the default is 350 in milliseconds.
Detect Default is -23(dB), effective range is -15 ~ -29 (dB).
dual-frequency busy
tones
After login, click the label of Advanced > RADIUS to open this interface.
Title Explanation
Primary server Set IP address and port number of preferred Radius server.
Note: if the port number is not configured yet, please use Radius default port number of 1813.
Key Set the share key to be used for encrypted communications between Radius client and server.
Note: the share key should be configured the same for both client and server side
Secondary server Set the IP address and port number of standby Radius server. When the fault appears in
communications between gateway and preferred Radius server, the gateway will automatically
activate standby Radius server.
Note: in case of no configuration of port number, use default port number of 1813.
Key The share key for communications between Radius client and standby Radius server.
Note: the key should be configured the same for both client and server side
Retransmit timer Set the amount of overtime on response after transmission of Radius message, the default is 3
seconds. The retransmission will be performed If no response is given after the timeout.
Retransmit times Set the times of retransmission of Radius message when no response is received default is 3
times.
CDR type Inbound: Set whether to send RADIUS charge message for inbound calls;
Outbound: Set whether to send RADIUS charge message for outbound calls;
Answered: Set whether to send RADIUS charge message when calls are connected;
Unanswered: Set whether to send RADIUS charge message for unanswered calls.
2.6.7 Encryption
After login, click the label of Advanced > Encryption to open this interface.
Title Explanation
T.38 encrypt Choose whether to encrypt T38 data. By default, this is not selected.
RTP encrypt Choose whether to encrypt RTP voice pack, the default is 0.
0: No encryption
1: Entire message
2: Header only
3: The data body only
Signal encrypt Choose whether to encrypt signaling. By default, this is not selected.
Encryption method Set the gateway encryption method, default is 7. The optional parameters as below:
2:TCP not encrypted
3: TCP encrypted
6: UDP not encrypted
7: UDP not encrypted
8: Using keyword
10: RC4
13: Encrypt13
14: Encrypt14
16: Word reverse(263)
17: Word exchange(263)
18: Byte reverse(263)
19: Byte exchange(263)
20:VOS
Encryption key You may obtain it from service provider
Session Border
Proxy
Server Set the IP address and port number of session border proxy server. The character of “:” must be
used between IP address and port number. Server address could be set into IP address or domain
name. When domain name is used, DNS service must be activated as shown in the page of
Network, and DNS server must be configured.
Example: 201.30.170.38:1020 or sbc.com:1020.
Signaling port Signaling port assighment of the gateway, the default value is 4660. Signaling port number may
be set at will, but can not conflict with other ports of equipment.
After login, click the label of Advanced > Tones to open this interface.
Figure 2-23 Call progress tone configuration interface
Title Explanation
Country/Region There are progress tone plans for several countries and regions which are pre-programmed in
gateways. Users may also specify the tone plan according to the national standard. Gateways
provide tone plans for the following countries and regions:
China; the United States; France; Italy; Germany; Mexico; Chile; Russia; Japan; South
Korea; Hong Kong; Taiwan; India; Sudan; Iran; Algeria; Pakistan; Philippines;
Kazakhstan.
Dial Prompt tone of off-hook dial tone.
2nd dial Used for the second stage dial tone.
Message waiting Used for prompt of voice mail, or when the subscriber line is set with “Don’t Disturb Service and
Call Transfer”.
Busy Used for busy line prompt.
Congestion Used for notification of call set up failure due to resource limit.
Ring back The tone sent to caller when ringing is on.
Disconnect Used for reminding the subscriber of off-hook and no dialup status of the phone.
Call waiting Used for notification in call waiting.
Confirmation Used for confirming function keys being entered.
Indicates 440 Hz single frequency tone, repeated twice in terms of 300 milliseconds on and 10
seconds off.
950/333,1400/333,1800/333,0/1000
Indicates repeated playing 333 milliseconds of 950 Hz, 333 milliseconds of 1400 Hz, 333 milliseconds
of 1800 Hz, and mute of 1 second.
The feature codes consist of system feature codes and service feature codes. The system feature codes
are used for acquiring gateway information, and the lattser is used for users to activate and inactivate
supplementary services.
After login, click the label of Advanced > Feature codes to open this interface.
The following are the examples of the dialing rule for the feature codes:
Using *xx (dial * and 2 digits number) to activate a service
Using #xx (dial # and 2 digits number) to cancel a service
This is illustrated with the following defaults for various parameters, which may be modified according
to requirements.
Figure 2-24 Feature codes configuration interface
Title Explanation
System feature
codes
Query IP address The function key for determining the IP address of gateway, with a default of ##. Dialing this
key, users can hear the gateway voice the IP address and system-software version number.
Narrative: if the gateway is only equipped with FXO (Line) port, connect FXO(Line) port
through the PBX extension line or PSTN direct line, and dial the number of this line
accordingly, press ## immediately after hearing the second dial tone, users may thus hear the IP
address and system software version number of the gateway.
Query phone The function key for determining the phone number of this subscriber line, with default of #00.
number By dialing this key, your will hear the phone number of the subscriber line voiceed by the
gateway.
Title Explanation
Service feature
codes
Activate CFU The function key for activating unconditional call forwarding, with a default of *60. Dialing
this key will activate unconditional call forward of the line and set the destination number
for call forwarding.
User operation: Off hook → press *60 →enter the destination number.
Users can determine the latest destination number set by dialing *60*.
Note: it’s required to enable call forwarding service before using this function (please see the
instructions on the relevant configuration of subscriber line).
Deactivate CFU The function key for deactivating unconditional call forwarding, with default of #60.
User operation: Off hook → press #60 → hang up.
Activate CFB The function key for activating call forwarding on busy,with default of *61. Dialing this key
may activate CFB, and specify the destination number.
Note: it’s required to enable call forwarding on busy service before using this function (please
see the instructions on relevant configuration of subscriber line).
Deactivate CFB The function key for deactivating call forwarding on busy, with default of #61.
User operation: Off hook → press #61 → hang up.
Activate CFNR The function key for activating call forwarding on no answer, with default of *62. Dialing the
function key may activate call forwarding on no answer and specify destination number.
Note: it’s required to enable call forwarding on no answer service before using this function
(please see the instructions on relevant configuration of subscriber line).
Deactivate CFNR The function key for deactivating call forwarding on no answer, with default of #62.
Activate CRBT The function key for activating color ringback tone, with default of *80. Subscribers may select
their favorite color RB tone by using this key.
Note: it’s required to start color ring service before using this function (please see Phone for
how to assign the feature to the phone).
User operation: Upon off hook, the subscriber may press the function key (e.g. *80),
then, input the two-digit index numbers of color ring;
*80* is used for hearing and inquiring the color ring that has been previously set.
Deactivate CRBT The function key for deactivating the color ring, with default of #80. The subscriber may use
such key to recover the normal ring of phone.
User operation: Off hook → press #80 → hang up.
Activate forking The function key for activating the double-ring/forking feature, with default of *75.
Deactivate forking The function key for deactivating the feature, with default of #75.
Activate DND Activate “Don't Disturb”, with default of *72. With DND selected, the gateway will reject all
coming calls by sending busy tone to the caller.
Note: It’s required to start “Don't Disturb” prior to using this function (please see the
instructions on relevant configuration of subscriber line).
Deactivate DND The function key to cancel “Don't Disturb”, with default of #72. Dialing the function key may
recover normal ringing upon the arrival of incoming calls.
Enable speed dials Define the function key of dial, with default of *74. This key allows the user to build a table of
2-digits(20~49)speed-dial numbers.
Note: It’s necessary to get the dial-up service under way before applying this function (please
see Phone for hwo to assign the feature to the phone).
User operation: Upon dialing the function key (*74), dial the two-digit speed dial
followed by the expanded number terminated with #.
Speed dial prefix The prefix number for applying abbreviated dialing, with default of **. The said prefix should
be added ahead of abbreviated dialing numbers when using abbreviated dialing.
User operation: off hook → dial the prefix number of abbreviated dialing(**)and dial
abbreviated dialing number (20).
Title Explanation
Suspend call waiting The function key for cancelling the call waiting feature for next call, with default of *64.
Dialing this function key will temporarily shield the user from a call-waiting distraction for next
call, avoiding the possible intervention.
Note: the function key works only for single cancel, if to cancel the call waiting completely,
please refer to the instructions on relevant configuration of subscriber line.
Blind call transfer Function key of blind call transfer,with default of *38.
User operation: During the call, tap the phone hook switch or press R button→ dial *38→ dial
the called number and then hang up.
Audit CRBT The function key for hearing the color ring,with default of *88.
User operation: Off hook → press *88 → input color ring number.
3-way *1
2.7 Status
2.7.1 Call status
After login, click Status > Call Status to open this interface.
Figure 2-25 Interface of call status
Title Explanation
Line There are six types of line status, On-hook, Off-hook, Ringing, Maintenance, Disconnect,
Parallel line in-use.
Call The call state includes Idle, Ooutpusling, Ring, Entering number, In progress, Ring back, Talk,
Near end hung up, Far end hung up, and Timeout.
After login, click Status > Call history on FXS to open this interface.
After login, click the label of Status > Call history on FXO to open this interface.
Figure 2-27 Interface of call on FX0
After login, click Status > SIP message count to open this interface.
Figure 2-28 Interface of SIP message count
2.8 Logs
2.8.1 System Status
Title Explanation
Login User Info Show the IP address and permissions of the login user. The numbers following the IP address
show the online permission level of the user: 1- administrator; 2 - operator; 3 – viewer. The
viewer can only read the configuration.
When more than one administrator logs in at the same time, the first login’s
permission level is 1; others are 3; also, when more than one operator logs in at the
same time, the first one’s permission is 2, others are 3.
For example:
Login User Info >>>>>
1) 192.168.2.247 1
Title Explanation
SIP Registration Info Show registration status:
Not enabled: The registration server’s address is not entered yet;
Latest response: The latest response message for the registration. 200 means
registered successfully;
No response: No response from registration server. The cause may contribute to 1)
incorrect address for the registration server; 2) IP network fault; or, 3) the
registration server is not reachable.
For example:
SIP Registration Info >>>>>
---- Not enabled ----
SIP Registration Info >>>>>
Contact: <sip:[email protected]:1003;user=phone>
latest response: 200 (timeout-555)
Contact: <sip:[email protected]:1003;user=phone>
latest response: 200 (timeout-555)
Latest Call Info Show the latest call.
Call Context Info Show the call status.
Rtp Context Info Show the voice channel related to the calls.
For example:
Rtp Context Info >>>>>
3) created, call =e011
After login, click Logs > Call Message to open this interface.
Figure 2-30 Call Message interface
After login, click Logs > System Starup to open this interface. Log files can be downloaded through
this interface.
Figure 2-31 Interface of System Starup
After login, click Logs > Manage Log to open this interface. Log files can be downloaded through this
interface.
Figure 2-32 Interface of Manage Log
Title Explanation
Log level Select the log file level of gateway, default is 4. The higher the level the more details the log
file will be.
Note: Log level should be set to 4 or lower when gateway is used in normal operation, avoiding
reducing the system performance.
System log server Set the IP address of the system log server.
Local log port The port used to send logs.
Log server IP address of debugging log server.
The procedure of downloading log files described hereof is only applicable to release 1.9.x.238 of MX
series or updated version of software.
2.9 Tools
2.9.1 Change password
After login, click Tools to open this interface. Only administrator is entitled to change the password of
login.
For changing administrator password, it’s required to enter new password into New password field and
Confirm new password field, then click Submit.
The password being used by the operator will be displayed as hidden codes, which could be changed by
the administrator at any time. The administrator is allowed to change the operator’s password by
entering the new password into Operator password > password.
Figure 2-33 Interface for password changing
After login, click Tools > Export data to open this interface. The download procedure is similar to the
After login, click Tools>Import data to open this interface. Operating procedure is the same as that of
software upgrade.
Figure 2-35 Interface of import data
2.9.4 Upgrade
After login, click Tools > Upgrade to open this interface. The software upgrad procedure is presented
as below:
Step1 Obtain the upgrade files (tar.gz file), and save the file onto a local computer.
Step2 Click Tools > Upgrade to access to the page of software upgrade.
Figure 2-36 Interface of upgrade
A few minutes are needed to upgrade the gateway. Don’t operate the gateway during this period.
Step7 After success in upgrade, the following dialog will appear, click Confirm.
Step8 The gateway is on the progress of reboot when the interface cannot be displayed.
Step9 Wait for about two minutes, and access the interface of gateway management system, click
Version info and check the software version.
For MX100 and MX120 gateways, the software upgrade operation must be conducted on an 100M
Ethernet port.
After login, click Tools > Restore factory settings to restore the factory settings.
The factory settings are designed based on common applications, and therefore, no need to modify them
in many deployment situations.
After login, click Tools > Restart to restart the gateway, making modified configuration come into
effect.
In most cases, there is no need to reset the gateway, and the modified parameters will come into
effect upon confirming the Submit.
After login, click Tools > Reboot to restart the gateway. As this is a system wide reset, it takes longer
time.
Generally, it’s sufficient to restart software when the gateway confirms to reset; the system reboot will
be required only when network settings of the gateway are changed.
After login, click Tools > TDM capture to open this interface. This tool can be used to capture the
voice stream from the Phone or Line interface. The capture starts from the off-hook if it is a Phone
interface or from the ringing if it is a Line interface,and is ended on on-hook or call release. When the
call lasts longer than 200 seconds, only the first 200 seconds of voice stream will be captured. The
voice file is stored on the gateway in PCMU format.
Figure 2-41 Interface of TDM capture
Step1 Select the analog line ID to which you want to perform the capture.
Step2 Click Start to initiate the capture procedure.
Step3 Make the test call.
Step4 Click Stop to terminate the capture procedure. You will be notified for donwload.
After login, click Tools > Ethereal capture to open this interface. You are allowed to capture up to
three IP voice data files, each with up to 2M bytes. The data files are stored on the gateway in
dump.cap format under catalog /var/log.
2.11 Logout
After login, click the Logout at top right to exit the gateway management system and return to the
login interface.
3 Appendix
3.1 MX120 system operation state
Table 3-1 MX120 system operation state