User Manual MG900

Manual of WEB Configuration
from NICEUC Company

(Used for all devices)
V2.0
Shenzhen NICEUC communication Tech Co., Ltd
Address: Room 1101-1102, Yizhe Building, Yuquan Rd, Nanshan District,
Shenzhen City, Guangdong, China (Postcode: 518052)
Tel: 86-755-26722761
Fax:86-755-26526970
Email: sales@nicecomm.com.cn
sales@niceuc.com
Website: http://www.nicecomm.com.cn
http://www.niceuc.com
http://nicecomm.en.alibaba.com
Shenzhen NICEUC Communication and TECH CO., LTD Manual of Web Page
Index
Chapter 1 Introduction of devices ........................................................................................... - 6 -
Chapter 2 Introduction of WEB Configuration ....................................................................... - 6 -
2.1 Connect devices by Web browser .................................................................................. - 7 -
2.2 Introduction for interface of web configuration ............................................................. - 7 -
Chapter 3 Operations for device ............................................................................................. - 8 -
3.1 Look for configuration ................................................................................................... - 8 -
3.2 Save configuration ......................................................................................................... - 8 -
3.3 Set to Default ................................................................................................................. - 8 -
3.4 Save to file ..................................................................................................................... - 9 -
3.5 Load from file ................................................................................................................ - 9 -
3.6 Reboot ............................................................................................................................ - 9 -
Chapter 4 Look for status ......................................................................................................... - 9 -
4.1 Device Details ................................................................................................................ - 9 -
4.2 Hardware Resources .................................................................................................... - 10 -
4.3 PCM trunk status.......................................................................................................... - 11 -
4.4 Analog line status ......................................................................................................... - 13 -
4.5 VoIP channel status ...................................................................................................... - 14 -
4.6 GSM line status ............................................................................................................ - 15 -
Chapter 5 Network Configuration ......................................................................................... - 15 -
5.1 Configure IP ................................................................................................................. - 15 -
5.2 DNS Setting ................................................................................................................. - 16 -
5.3 NTP setting................................................................................................................... - 16 -
Chapter 6 Work configuration ................................................................................................ - 17 -
6.1 System .......................................................................................................................... - 17 -
6.1.1 System function ............................................................................................. - 21 -
6.2 Rule (Call Rule) ........................................................................................................... - 25 -
6.2.1 Length Rule .................................................................................................... - 26 -
6.2.2 Number Convert Rule ................................................................................... - 28 -
6.2.3 Router Rule .................................................................................................... - 29 -
1) Trans to Truck (Transfer to PCM Trunk) ....................................................... - 31 -
2) Trans to VoIP.................................................................................................. - 33 -
3) Trans to FXO (MG device don’t have this function) ..................................... - 34 -
4) Trans to Wireless (MG device don’t have this function) ............................... - 35 -
5) Trans to DTMF Key (MG device don’t have this function) .......................... - 37 -
6) Trans to Computer Operator (OPR) (MG device don’t have this function) ... - 38 -
7) Trans to Extension (MG device don’t have this function) ............................. - 39 -
8) Queuing (Enter to queue) (MG device don’t have this function) .................. - 40 -
9) Function (MG device don’t have this function) ............................................. - 41 -
10) Call Pick-up (MG device don’t have this function) ..................................... - 41 -
11) Trans to FXS (MG device don’t have this function) .................................... - 42 -
12) CTI Ctrl (Control) (MG device don’t have this function)............................ - 42 -
13) Call in Holding (MG and IAD device don’t have this function) ................. - 42 -
sales@niceuc.com supports@niceuc.com -2-

14) Conference ................................................................................................... - 43 -

15) Transfer to Trunk (Limit length of called number) ...................................... - 45 -
16) Number Change Notes (Query DB) (For device PBX only) ....................... - 46 -
6.2.4 Router Line Group ......................................................................................... - 47 -
6.2.5 Rule limit ......................................................................................................... - 47 -
6.3 PCM ............................................................................................................................. - 49 -
6.3.1 ISDN PRI ........................................................................................................ - 51 -
6.3.2 SS7 .................................................................................................................. - 53 -
6.3.3 R2 (/CAS) ....................................................................................................... - 59 -
6.3.4 V5.2 ................................................................................................................. - 62 -
6.3.5 Q.SIG............................................................................................................... - 64 -
6.3.6 Virtual caller .................................................................................................... - 64 -
6.3.7 Number Plan .................................................................................................. - 64 -
6.4 Analog Line .................................................................................................................. - 66 -
6.4.1 FXS .................................................................................................................. - 67 -
6.4.1.1 User Number .......................................................................................... - 67 -
6.4.1.2 User Group ............................................................................................. - 67 -
6.4.1.3 Call Priority ............................................................................................ - 68 -
6.4.1.4 Out Gain and In Gain ............................................................................. - 68 -
6.4.1.5 User Password ........................................................................................ - 70 -
6.4.1.6 Dial Function .......................................................................................... - 70 -
6.4.1.7 Service Functions ................................................................................... - 70 -
6.4.1.8 Signal Cycles .......................................................................................... - 74 -
6.4.1.9 Ring Mode.............................................................................................. - 75 -
6.4.2 FXO ................................................................................................................. - 76 -
6.4.2.1 Signal Detection ..................................................................................... - 76 -
6.4.2.2 Detect DTMF CID ................................................................................. - 77 -
6.4.2.3 Out Gain and In Gain ............................................................................. - 78 -
6.4.2.4 Service Functions ................................................................................... - 79 -
6.4.2.5 Refuse Call-in......................................................................................... - 82 -
6.4.3 Other Line Types (PBX don‟t have this function)................................ - 82 -
6.4.3.1 PTT lines ............................................................................................ - 82 -
6.4.3.2 E&M.................................................................................................... - 86 -
6.4.3.3 Magnet................................................................................................. - 90 -
6.4.4 Module Configure (PBX don‟t have this function) ..................................... - 91 -
6.4.5 Ring-back Tone .............................................................................................. - 92 -
6.4.6 Tone Setting (PBX don‟t have this function) .............................................. - 92 -
6.5 VoIP (PBX don’t have this function) ............................................................................ - 93 -
6.5.1 Server Parameter .......................................................................................... - 96 -
6.5.2 FAX ................................................................................................................ - 100 -
6.5.3 VoIP User ...................................................................................................... - 102 -
6.6 Wireless Line.............................................................................................................. - 103 -
6.6.1 Mobile Information ....................................................................................... - 103 -
6.6.2 BCCH Show ................................................................................................. - 104 -

6.6.3 Out Gain and In Gain of Wireless line ...................................................... - 105 -

6.6.4 Call Back ....................................................................................................... - 106 -
6.6.5 Hot Dial ......................................................................................................... - 106 -
6.6.6 IMI and Use Limit......................................................................................... - 107 -
6.6.7 Wireless Switch-board Number ................................................................. - 107 -
6.6.8 SMS Operation ............................................................................................ - 108 -
6.6.9 Command Operation................................................................................... - 110 -
6.6.10 USSD Operation ........................................................................................ - 110 -
6.6.11 Call Statistic ................................................................................................ - 112 -
6.6.12 Fee Operation ............................................................................................ - 112 -
6.6.13 SIM Rotation .............................................................................................. - 113 -
Chapter 7 System Function ................................................................................................. - 115 -
7.1 SIP-T / SIP-I............................................................................................................... - 115 -
7.2 Media Control of Ring-back Tone.............................................................................. - 115 -
7.3 Close Early Media ...................................................................................................... - 116 -
7.4 Setting for Calls overtime .......................................................................................... - 117 -
7.5 SNMP ......................................................................................................................... - 117 -
7.6 Intelligent Analysis Call Forward .............................................................................. - 118 -
7.7 Smart Call-back.......................................................................................................... - 118 -
7.8 SMS Transmit ............................................................................................................ - 118 -
7.9 Network Limit ............................................................................................................ - 119 -
7.10 Non Tonal Detect ..................................................................................................... - 119 -
Chapter 8 System Manage .................................................................................................. - 119 -
8.1 Telnet Account ........................................................................................................... - 119 -
8.2 Access Control ........................................................................................................... - 119 -
8.3 Change Password ....................................................................................................... - 120 -
8.4 Update Firmware........................................................................................................ - 120 -
8.5 Update MSP ............................................................................................................... - 121 -
8.6 Update Voice .............................................................................................................. - 121 -
8.7 Debug Trace ............................................................................................................... - 121 -
8.7.1 Introduction of common commands ......................................................... - 121 -
Appendix 1 Description of INI File ...................................................................................... - 123 -
Appendix 2 Description for Function of Keys on Phone .................................................. - 123 -
Appendix 3 Make and Load Ring Tone .............................................................................. - 125 -
Appendix 4 Information Name for SS7 Signaling ............................................................. - 127 -
Appendix 5 SS7 link .............................................................................................................. - 130 -
1. One SS7 Signaling Link to Manage Multiple Devices (or cards) ................................ - 130 -
2. Two SS7 signaling links to manage three devices (or three E1/DSP cards) ................ - 133 -
Appendix 6 FAQ .................................................................................................................... - 136 -
1. Network Failure ........................................................................................................... - 136 -
2. E1 Failure ..................................................................................................................... - 136 -
3. PRI Failure ................................................................................................................... - 136 -
4. SS7 Failure ................................................................................................................... - 137 -
5. Failed to connect PcmMonitor ..................................................................................... - 137 -

6. No voice of SIP calls .................................................................................................... - 137 -

Appendix 7 Line Order of cables......................................................................................... - 137 -
1. Console cable ............................................................................................................... - 137 -
2. Network cable .............................................................................................................. - 138 -
3. E1 cable ........................................................................................................................ - 138 -
4. Analog FXS/Extension cable ....................................................................................... - 139 -
5. E&M cable ................................................................................................................... - 140 -
6. PTT cable ..................................................................................................................... - 144 -
A.7.1 An example for Connection of PTT Card....................................................... - 145 -
A.7.1.1 Physical Connection ............................................................................ - 145 -
A.7.1.2 Line Order of Connection Cable ......................................................... - 146 -

Chapter 1 Introduction of devices

WEB configuration for all devices from NICEUC Company are same, it uses the unified frame.
According to the different functions and application, there are four kinds of devices.
·MG series: VoIP Media gateway/E1 trunk gateway, support E1/T1 and SIP, product model
number: NC-MG900 and NC-MG1000.
·IAD (Integrated Access Device): Analogue Voice Gateway and wireless GSM/CDMA/WCDMA
Gateway, support E1/T1, FXO/FXS, E&M, GSM/CDMA/WCDMA and SIP. Product model
number: NC-MG160, NC-MG320, MG08W-64SIM, NC-MG320W, NC-MG640, NC-MG640W,
NC-MG930, NC-MG930W.
·IPPBX: IP Integrated Gateway, support E1/T1, FXO/FXS, E&M, GSM/CDMA/WCDMA and
SIP, and can be built-in SIP server, provide with more value-added services like as CTI and
Recording etc. Product model number: NC-MG320, NC-MG640, NC-MG930.
·PBX: Voice Switch; support E1/T1, FXS/FXO, and GSM. Product model number:
NC-AD300R, NC-AD300D, etc.
In the following description, if one parameter is not a common parameter, there will mark and
distinguish.
Chapter 2 Introduction of WEB Configuration
WEB refers to web page, network, internet and other technology fields. It uses HTTP which
provides subscribers with visual interface. Because of simplicity, now it is widely used in each
field. Our devices provide with WEB service, subscribers can configure and monitor the
devices by opening the web browser only, there is no need to install other extra software, and
it‟s very easy and convenient.
When the connection of network is ready after the installation of devices, there can use web
browser to connect devices. The default IP for all of our devices from factory is
“192.168.16.254”; the default port is 80 for web services.
Our web service supports most of web browsers, such as Microsoft IE, Google Chrome and
Mozilla Firefox, etc. To be better in showing, please use Google Chrome.

2.1 Connect devices by Web browser
Enter the IP of devices (such as the default IP “192.168.16.254”) into address bar of browser,
there will pop up to remind the login, and then enter the user name and password. At present,
our devices provide with one account only to login. User name is “admin” which cannot be
changed, default password is “nice” which can be changed by subscriber.
2.2 Introduction for interface of web configuration
After login successfully, come into the interface of web configuration. Descriptions for the
whole frame of the interface are as follow:

1. Logo 3. Language
2. Name
of device
4. Operations
for device
5. Index for
functions 6. Contents
1. Logo: company logo of NICEUC; or we can make customer‟s logo here.
2. Name of device;
3. Language of web configuration. There support two languages only, or you could select that
there show English only (change by web resource database)
4. Operations for devices, some related operations.
5. Index for functions: tree structure; Click “+” on the left, then will unfold sub key.
6. Contents: for each sub key of each function.

Chapter 3 Operations for device
3.1 Look for configuration
Data of device is saved in ROM. When booting, there will read configuration from ROM and
load into RAM.
Click “Read ROM”, to look for the configuration data that saved.
Click “Read RAM”, to look for the configuration data that running.
If read successfully, there will remind that reading of all configuration data complete.
Note: ROM is the hard disk of device, RAM is the memory of device.
3.2 Save configuration
After changed the configuration data, there need to save the data, so that the changing will be
effective.
Click “Write ROM”, to save the configuration data into ROM, it will be effective when reboot.
Click “Write RAM”, to write the configuration data into RAM, parts of data will be effective
immediately.
If wrote successfully, there will remind that writing of all configuration data complete.
Suggest：After changed the configuration data, it’s better to write into both ROM and RAM.
3.3 Set to Default
Click “Set to Default”, here can set all data to default value. When it is the first time to install the
device, firstly set to default, then change the data according to the actual requirement.
Attention: this operation is only to set the data to default value, separately there should
do “write ROM” or “write RAM”, therefore the configuration can be saved into device.

3.4 Save to file
The configuration data of device can be saved as file into PC, to backup.
Click “Save to file”, then there will remind to select the format of file, at present there only
support format “INI”; if the speed of network is slow, you can select “Compressed” to save the
traffic. In normal case, there is no need to select “Compressed”. Click “Ok”, there will pop up
saving or opening the file, select “save” to save it.
Attention: before saving, please “Read ROM” or “Read RAM” in advance.
3.5 Load from file
In same place when there should change the device, or in same project when there should use
several sets of devices, “Load configuration data from file” is necessary, it will be timesaving,
security, no mistake.
Click “Load from file”, there will point out “select file” and “format of file”. Click ”select file”, there
will ask to open the loaded file. Please pay attention to format, if it is compressed, then there
must select “Compressed”, otherwise there will be a mistake. At last, click “ok” to load. If
successfully, there will show “loading from file is successful”.

Attention: Loading is only to load the configuration data into web interface,
separately there need to “Write ROM” or “Write RAM”.
3.6 Reboot
Some configuration data need to reboot to be effective. Click “Reboot”, the device will
hot-reboot. The result is same as the command “reset”, the device will reboot after 3 seconds.
Chapter 4 Look for status
4.1 Device Details
Click “Device Details” in index area, there can look for the details of this device, such as
hardware, software, etc. For example:
App version 8.14,Made on Feb 13 2014, 17:58:32 Version of software and

made date
soft info:931 Software type
sid:0x0,123456 Serial number of device
dev type:30 Hardware type
dev id:0x00 Hardware ID
dev version:0x33 Hardware version
AUTH Authorization information
OK,E1=4,USER=208,GSM=64,VOIP=128,REG=1024,SAPI=0,OID=42419. for the quantity of lines
931
AUTH FC: CAS V52 SS7 PHLNK CTILNK SIGMIX CONF REC SAPI EXT Authorization information
SNMP for functions
Firmware version: v10_25_03_02 Version of MSP software
SPU version: C64V_7_9_2 SPU Version
Hardware type of MSP
MSP type(21): M82354,channels=64 and the quantity of
channel
4.2 Hardware Resources
Unfold the “Device Details” and click “Hardware resources”, to look for the line resources of
device.
·Resources Count: total resource. Look for the total count of the device.
If the count of one resource is 0, it means that the device don‟t have this resource.
·PCM trunk: if there is E1/T1 port on device, there can look for the status of PCM trunk as
follow:
sales@niceuc.com supports@niceuc.com - 10 -
Color green means the connection of PCM is normal; others mean there is a problem in
connection of PCM. Each icon represents one E1/T1 port, keep the mouse stop on each icon,
there will show the details of E1/T1 port.
·Analog Line: if there are analog lines on device, there will show the following list of analog
lines.
Each icon represents one line, keep mouse stop on icon, then there can show the types of
lines, each type is with different color.
In the above screen-shot, from top to bottle, they are FXO(external line) in blue,
FXS(internal line/extension) in red, High-Resistance Recording in green, PTT(Push To
Talk/RoIP) in purple, E&M line in dark blue, Magnet line in brown.
·Wireless Line: if there are GSM, CDMA, WCDMA and other wireless lines, there will show the
following list for wireless line:
Each icon represents one line, the symbol on the right show the signal strength of wireless line.
Keep mouse stop on the icon, and then there can show detailed status of wireless line.
4.3 PCM trunk status
If there is E1/T1 port on device, please unfold “Device Details” and click “PCM trunk status”, to
look for the detailed status of E1/T1 port.
The icon on the left is the physical connection status of PCM; color green means clock
synchronization is normal, color red means there is a warning in physical connection, color
dark gray means invalid or forbidden. Next the number from 0 to 31 represent the 32 time-slots
of E1/T1. Keep mouse stop on icon, and then there show the detailed status. Click the icon
under the time-slots number, there will pop up the details of this time-slot.
If there configure PRI, there can look for the status of PRI link as follows:
Color green means normal, red means fault. Keep mouse stop on there, then there can show
the detailed status.
If there configure SS7, there can look for the status of SS7 link as follows:
Color green means normal, red means fault. Keep mouse stop on there, then there can show
the detailed status.
About the details of each icon please look for the following picture:
There is a button “Refresh”, click it to refresh the status manually, also there can set the time
interval to let it refresh automatically.
The value of time interval can not only select, but also can fill with it, then press “enter”.
Click “BER Statistics”, to switch the list of BER Statistics(BER: bit error rate), then to look for
the case of bit error.
Click “Query Data”, there will purge the previous data, and re-calculate the bit error.
PCMID, PCM ID.
CRCE, Error of CRC-4.
FER, FAS/ NFAS Bit or Pattern Error.
COFA, the alignment of new-found basic frame is different with the previous one.
OOF, Out of basic frame synchronization.
LVC, Bipolar Violation (BPV) Error (In AMI decoding) or HDB3 code Violation CV (Error) ) (In
HDB3 decoding).
4.4 Analog line status
If there is analog line on device, unfold the “Device Details” and click “Analog line status”, to
look for the detailed status of analog lines. There are two forms: Table and Graph, click them to
switch. In form table, look for “using state”, “call step”, “Caller number”, “Called number”,
“Ringing Duration”, “Talking Duration” and other detailed information.
Table View as follows:
Graph as follows:
There is a button “Refresh”, click it to refresh the status manually, also there can set the time
interval to let it refresh automatically.
The following picture shows the meaning of each icon of line status:
4.5 VoIP channel status
If the device supports VoIP, unfold “Device Details” and Click “VoIP channel status”, to look for
the detailed using status of VoIP channels. Here is same as “Analog line status”, there are two
forms: Table and Graph.
4.6 GSM line status
If there is GSM line on device, unfold “Device Details” and Click “GSM line status”, to look for
the detailed using status of GSM line. Here is same as “Analog line status”, there are two
forms: Table and Graph.
Analog line status, VoIP channel status and GSM line status, the content under these three
items are same.
Chapter 5 Network Configuration
5.1 Configure IP
There are two LAN ports on device, LAN 0 and LAN 1. Only LAN 0 can be used for VoIP
call. Click “Network”, change IP on the right, and only support IPv4. The new version will
support DHCP.
The secondary IP can not only be configured on LAN 0, but also can be configured on LAN 1.
When select “used by-eth 0”, the secondary IP will configure on LAN 0; When select “used
by-eth 2”, the secondary IP will configure on LAN 1;
Attention: Changing IP of LAN 0 will be effective by only clicking “OK”. But

changing the secondary IP will be effective by clicking “ok”, doing “write ROM”,
and rebooting device.
Attention: After changed the IP of LAN 0, SIP calls will not work well, please
reboot device.
If there need several IP, unfold “Network” and click “Net Route”, there are maximal 5 IP routes.
Take the above picture as sample, when visit 192.168.2.XXX, there use 192.168.2.1 as the
address of gateway.
5.2 DNS Setting
If there need to use domain name system, then there need to configure the IP address of
DNS. Unfold “Network” and click “DNS Setting”, there can configure maximal 3 DNS.
If use broadband router and there is a function of DNS agent, then there can fill with the
internal network address of router as DNS.
5.3 NTP setting

Sometimes there will use something related with time, for example, call detailed recording,
then such a item will ask the accuracy of time. If connect with Internet, there can adjust
time from Internet. Unfold “Network” and click “NTP setting” to configure. For example:
Server address: when fill with “default”, mean look for the default time server.
Time zone: default time zone from Internet.
Refresh synchronization period: the time interval for adjust the time. Fill with 0, means
there will be one adjustment when reboot or write RAM. Take minute as unit.
Chapter 6 Work configuration
Before changing configuration, we suggest to read firstly the original configuration, then
change, in this case there can avoid the wrong operation. After configured the work data, save
the configuration as file to back up.
6.1 System
Some main system data configure here.
Click “System”. Icon with question mark in blue on the right is the explanation for each item.
Telnet Listen Port, specify the port of telnet service, usually configure as 5204.
Telnet Time Out, configure the valid time length for the connection of telnet. If over this time,
the connection will force to disconnect. “0” means disable time out.
PCM Monitor, configure the network port to monitor PCM signaling (UDP receiving port), it‟s
used for connecting the tool PCMMonitor.
PBX Language, when select English, PBX is playing the voice, will switch automatically to the
index position of English Voice. (MG device don‟t have this data)
Country Tone, different country is with different standard of tone. At present, support four
standards from America, India, China and Russia. Besides, this tone can be customized. (PBX
device don‟t have this data)
DSP Setting, configure the function and usage of DSP. (Only used for PBX device)
Different device is with different quantity of DSP. Functions are as follows:
Voice, Used for recording or playing voice.
MFC/DTMF, Used for detecting DTMF, or used for signaling R2 to receive/send number.
Conference, it is used for multiple way conference and conference management.
Caller ID, Used for Caller ID, please ignore this function, it‟s useless.
MFC/DTMF (ig), it is one variant of MFC/DTMF.
DTMF06, high accuracy of the detection for DTMF.
MixRec, Used for Mix the recording of two-way calls.
CDR (call details recording) send to: configure that send the CDR to the specified port of
specified IP address. Port usually uses 7200.
Because of number convert, perhaps the caller and called number when call in/call out will be
different, therefore need to record the number when call in. Select “CDR According Call In”,
means use the number when call in, otherwise use the number when call out.
Device support both TCP and UDP to receive CDR at same time. If use TCP, device is the
service terminal, there can select maximal 3 client terminals. If using UDP, there only send to
one specified IP. Port is as the UDP receiving port from receiver.
The original format of CDR is text strings and is end with "\ r \ n". Each string separated by a
space. Field is with fixed length, if the text string is not long enough, will add the space to this
fixed length.
In the text string, there are four kinds of line types, 0 is E1,1 is analog line, 2 is VoIP lines, 3 is
wireless lines.
Call duration is seconds.
If using UDP to receive CDR, there are four formats.
Format 1
line 1 Line 2 Caller Called Talk

String logo End
type type number number time
Length 1 1 1 1 1 1 15 1 20 1 5 2
Description R space space spae Left Space Left Space right “\r\n”
String Example: “R 1 0 88889010 98888 5”
The sample indicated that calls from analog lines to digital trunking, calling number is
88889010, the called number is 98888, call duration is 5 seconds.
Format 2
Line 1 Line 2 Caller Called Talk

String logo End
Type number Type number number number time
Length 1 1 1 3 1 1 3 1 15 1 20 1 5 2
Description C Space space space left space left space right “\r\n”
String Example :“C 1 18 2 1 9010 9208 37”
The sample indicated that calls from analog lines number 18 to VoIP channel 1, calling number
is 9010, the called number is 9028, and call duration is 37 seconds.
Format 101
Line 1 Line 2 Line 1 Called Talk

String logo End
Type Type number number time
Length 1 1 1 1 1 1 15 1 20 1 5 2
Description R Space space space left space left space right “\r\n”
String Example :“R 1 0 009 10000 14 ”

The sample indicated that calls from analog lines number 9 to digital E1, the called number is
10000, and call duration is 14 seconds.
Format 240
Line 1 Line 2 Caller Called Talk

String logo End
Type number Type number number number time
Length 1 1 1 4 1 1 4 1 20 1 60 1 8 2
Description R Space space space left space left space right “\r\n”
String Example :“R 1 7 3 12 9203 13700000000 22 ”

The sample indicated that calls from analog lines number 7 to GSM line 12, calling number is
9203, the called number is 13700000000, and call duration is 22 seconds.
The above format is effective only for UDP.
If using TCP to receive CDR, there are five formats.
129, device as TCP client terminal, CDR format is same as format 1 of UDP.
Other value, device as TCP service terminal, the format as follows:
“\rCaaaa1, aaaa2, sssss, nnnnn, bbbb, cccc\n”
Beginning at”\r”; the identity Character “C” as logo
aaaa2 means: line number
ssss means :end calling time
nnnn means: talk_duration time
bbbb means: caller number
cccc means: called number
The end: a carriage returns “\n”
Call monitor, use the sending port and address of CDR to send the calls data.
Select “Send All messages”, send all messages that monitored. (MG device don‟t have this
parameter)
Select “Send Main Messages”, send main messages only that monitored. (MG device don‟t
have this parameter)
Select “Not Send CDR”, only send monitored messages, don‟t send CDR. (MG device don‟t
have this parameter)
By default, use UDP to send monitored message; if format configures as 136, that means “use
TCP to send message, device as TCP client terminal.
Log send to-destination; send the debugging information to the specified IP address and
port, used for tracing debugging.
6.1.1 System function
Main global function configure here.
Auto Record, enable and disable; (need to run a software in a PC to receive voice data)
Enable, automatically send all the voice data of device to the recording sever, but don‟t include
the voice data between extensions. If want to include the voice data between extensions,
please select “Enable (including extension between)”. (MG device don‟t have this
parameter)
DTMF Tx Amplitude (Tx: transmit), adjust the amplitude for sending DTMF. The primary tone
amplitude can program from -6.2 dBoV to -69.2 dBoV. Unit of this parameter is -0.1 dBoV,
range is from 62 to 692. (0 dBm0 ≈ -6.2 dBoV) (PBX device don‟t have this parameter)
DTMF Tx Cycle, adjust the cycle for sending DTMF. (PBX device don‟t have this parameter)
No Answer Overtime, unit is second. Configure 0, means disable this function. Otherwise, it
over this time, there is no answer, the call will be disconnected. (PBX device don‟t have this
parameter)
Max Talk Duration (Max: Maximal), unit is second. Configure 0, means disable this function.
Otherwise, if the talk duration over this time value, the call will de disconnected. (PBX device
don‟t have this parameter)
Intelligent Analysis Call Forward, sometimes, E1 on MG device will connect with the pbx,
there configure call forward on PBX, but the pbx can not send three number out, in this case,
there need our MG device to analyze and process.
For example, A call B, B call forward to C, call route as follows:
1. Send call “A to B” to E1.
2. Got a call “A to C” from E1;
Select this parameter, MG will judge this call is a call forward, if route is converted to SIP, will
use string “Diversion” to take number B. (PBX device don‟t have this parameter)
Send CID with date time, when device send FSK information of CID, whether take with date
and time. (PBX device don‟t have this parameter)
SIP-T or SIP-I, in SIP message, takes with ISUP message data of SS7. Select “Pcm->VoIP”,
convert the ISUP message of SS7 that received from E1 to txt, and take into SIP message;
Select “VoIP->Pcm”, transmit the SS7 message that collected from SIP message into E1.
(PBX device don‟t have this parameter)
SNMP setting, select “Enable” to enable SNMP. (PBX device don‟t have this parameter)
Server IP address, specify the IP address of SNMP software, used for sending Trap
message.
Server port, specify the receiving port of SNMP software, used for sending Trap message.
Trap send PCM Sync State, send status of E1/T1 clock synchronization by Trap message.
Trap send PRI Link Step, send status of PRI link by Trap message.
Trap send SS7 Link Step, send status of SS7 link by Trap message.
Trap send Line Call Information, send call status of all lines by Trap message.
Trap send PCM Sync State (Detail), send detailed status of E1/T1 clock synchronization by
Trap message.
The following system function is used for PBX only.
Disable listen when use Convert Rule for automatic auxiliary dial-up, a special
application. When use convert rule, NC-AD300D can make automatic calls and send key,
then enter into automatic IVR procedure of the other side, at last send the true called
number to call out. For example, process of 800 phone card, there can use NC-AD300D
to complete automatically. After select this parameter, subscriber cannot hear any voice of
IVR, until connect with the called number finally. (This parameter is used for PBX only)
Call monitor, a special application. Send monitoring message of calls by the network
configuration of CDR, usually used for the simple secondary development. There are
three mode, incoming, outgoing and both way. The applications of these three modes are
E1 High-Resistance Parallel Collecting, by using high-resistance isolator to connect the

collected E1 line, receive the signaling data, then analyze and process, at last compose a
call. if monitor it own self call message, please fill with the value manually. When configure
as 17, means “monitor simple message”; when configure as 49, means “monitor detailed
message”. (This parameter is used for PBX only)
(UDP) SS7 Monitoring Date--Send To, this is a special application. Use NC-AD300D as a
device which is used for collecting signaling, by high-resistance isolator, receive the link data
of SS7, then use UDP to send the link data to the specified address and port. This function is
used for secondary development. (This parameter is used for PBX only)
CTI-UDP target (CTI application, the connection target of UDP), this is a special
application. Send CTI message by UDP to the specified address and IP. Keep Live Seconds,
used for judging the connection status. (This parameter is used for PBX only)
Signaling Transfer--PCM Map Block, this is a special application. When use as signaling
converter, in some application, subscriber need to know the feedback of PCM from the other
side. Divide PCM into group A and B, group A as SS7, and group B as PRI or R2. When all
PCM in group are with problem, device will block automatically the PCM in group B, to notice
subscriber. Vice versa. For example, PCM0，PCM1，PCM2 use R2, PCM3, PCM4，PCM5 use
SS7, the configuration should be as: Select 3, 4, 5 in group A; select 0, 1, 2 in group B. (This
parameter is used for PBX NC-AD300R and NC-AD300D only)
6.2 Rule (Call Rule)
Define the call processing flow of the device.
First of all, grouping the incoming line, then add the corresponding rule. When configure,
Please taking device as the center to judge the incoming and outgoing calls.
Click “Add new group”:
Then pop up a configuration page:
Firstly select “Line Type”, secondly select “Line Range” (Attention: the last end value is not in
the range), at last select the rules for these lines.
Attention please: the line type of different device is different; the quantity of line is also
different.
Rules are Length rule, number convert rule and router rule. They are three separate tables. In
all of three rule tables, there is one "group number" which is used to specified group of rule. In
table of line group, cite each rule by group number. "ConvertRuleID(In)" and
"ConvertRuleID(Out)", they use the same one table of number convert, the difference is the
front is used for call in, the back is used for call out. The effective value of group number is
from 0 to 254; if don't need to handle, please fill with 255.
When a call is coming, firstly judge the type of incoming line, and then look up range according
to the type in the rule table. As the above screenshot, for example, the incoming call is from
the 5th time slot of PCM0, so the line type is trunk, the line ID is 5, match to the line of serial
number 0 in the group line table of Rule. In that way, when judging the number length, look up
the rule of number 0 in the length rule table. There is filled with 255 in the ConvertRuleID(In)
table, so no need to handle. Then look up the rule of number 0 in the RuoterRuleID table to
process. The processing for ConvertRuleID(Out) is after checked the router; if the router
transfer to VoIP, so it will look up according to the value of the line (with number 1) in VoIP line;
if the router transfer to trunk, it will look up according to the value of the line (with number 0) in
trunk line.
To configure conveniently, WEB interface provide with the function of filter processing. In the
group line table of Rule, if configured rule group, then the group number of rule will become a
button, click it and enter the configuration of the corresponding Rule. In the three rule tables,
click “Go to group line table”, there can transfer back to the group line table of rule.
Attention: all parameters in Rule will be effective immediately after “Write ROM”
and “Write RAM”.
6.2.1 Length Rule
When device is processing the received number, there need to judge the number length of
various case.
Upfold “Rule” and click “Length”, configure length rule on right, to add, modify, delete, or other
operations.
Click “Addition” or “Modify”, there pop up configuration page.
·RuleID: Used for citing in the Rule table.

·SID: Prefix of a number, it is the judgment conditions for processing rules.
·Length rule: Search by dividing section; divide a number into several sections to judge length.
There can divide into maximal 8 sections (or 8 grades). When searching, firstly from the
beginning of the grade with smallest value. if the "FindingResult" is "Finded, End", then stop,
directly process according to the range between "MinLength" and "MaxLength". If the length is
less than "MinLength", then it is timeout for waiting; When receiving number is ended because
of timeout, and the length is still less than the "MinLength", then reject the call. If the length is
more than or equal to the "MaxLength", then the receiving number is ended. If "FindingResult"
is "continue to find", then after subtract "ThisLength" from SID, and continue to search next
grade. If don‟t search the matched "SID" when searching next grade, and if configured “Next
length default”, then process according to default length. Otherwise it failed to search.
See the configuration as showed in the above screen-shot, for example, search in Rule ID 0,
Number is „17909075526520000‟ then searching progress are as follows: First, find rule in
grade 0, there is no matching SID; Then continue to find rule in grade 1, match with the SID 17
of order #18; "This length" of order #18 is 5, so when find next grade, the number that should
be found is changed to 075526520000. Continue to find rule in grade 2, match with the SID
0775 of order #19, “This length” of order #19 is 4, so when find next grade, the number that
should be found is changed to 26520000. Continue to find rule in grade 3, there is no grade 3.
But in last grade (order #19), the default length is 8, 26520000 is just with 8 digits, then the
„finding‟ is successfully over.
The progress of finding of number „*8‟ are as follows: firstly find rule in grade 0, match with
order #1, the “Min Length” is 2 digit, the finding result is “Finded, end”, so that no need to find
further, the „finding‟ is successfully over.
The progress of finding of number „2345678‟ are as follows: firstly find rule in grade 0, there is
no matching SID; Then continue to find rule in grade 1, match with the SID 2 of order #6; "Min
length" of order #6 is 1, "Max length" of order #6 is 8, the digit of number „2345678‟ is 7, this is
more than Min length, but is not up to Max length, Device will be in the status of waiting,
continue to wait for the next key. If over time, there is no next key, then the finding is ended.
6.2.2 Number Convert Rule
Sometime the application case is complicated, there need to add/reduce number

according to rules, so should use number convert rule.
Upfold “Rule” and select “Convert”, configure convert rule on right, to add, modify, delete
and other operations.
·Rule ID: used for citing in the Rule table.

·SID: Prefix of a number, it is the judgment conditions for processing rules. (When Rule ID is
less than 128, means “judge as called number”; when Rule ID is more than 127, means “judge
as caller number”.)
·Removed Length: the length that removed from the front of a number. Prefix: add prefix
number in the front of a number. Postfix: add postfix number in the back of a number. It‟s
possible that the caller and called number convert at same time.
For example, in group rule ID 132, if the SID of caller number is at the beginning of 0, then
remove 1 digit from the front of caller number, remove 4 digits from the front of called number.
Attention: in prefix or postfix, “&” means “insert the previous caller number; “@” means
“insert the previous called number; “$” means “insert the original called number (need to
support by Signaling)”.
6.2.3 Router Rule
Define the processing situation of calls. Unfold “Rule”, and select “Router”, configure convert
rule on right, to add, modify, delete and other operations.
·Rule ID: Used for citing in the Rule table.

·Class: Priority, specify the priority of routing rules, will be related to the priority of analog
circuit parameters in the SLC->Line Part table. When priority of extension is more than priority
of route, then can make calls. (MG device don‟t have this parameter)
·SID: Prefix of a number, it is the judgment conditions for processing rules. (When Rule ID is
less than 128, means “judge as called number”; when Rule ID is more than 127, means “judge
as caller number”.)
Sometimes there are so many number to match and judge, but the capacity of router rule table
and number convert table is limited ( maximal 128pcs). When format of SID is “cdb, L, G”,
means to look up CDB list. L is the length of phone number in CDB list, G is group number in in
CDB list. (CDB, it is a list of Special SID)
Unfold “Rule” and click “CDB”, to configure CDB list.
·Sequence: when there are many rules with same matched conditions, the action will be
followed from the sequence small to large, so when the small sequence failed, route can goes
to larger number sequence, and so on; when there are some routes are impassable, there can
choose other route, thus can back up for multi-routes.
Under case of default configuration, device will judge automatically the result of call-out, and
determine whether to carry out secondary route. If select “secondary router enable”, then force
to enable secondary router on the line.
·White list G ID (Group ID): Limit the calls. White list: only the number in white list is allowed
to call. Here specified group ID. Unfold “Rule” and click “White list”, to configure white list.
·Black list G ID (Group ID): Limit the calls. Black list: only the number in Black list is not
allowed to call. Here specified group ID. Unfold “Rule” and click “Black list”, to configure Black
list.
For device PBX only, there is a special function as follows.
When the range of the group ID of black white list is from 100 to 200, this means “use CDB” as
black white list. When the group ID of SID is same as this group ID, then will match. Find in two
times, the first time reduce one digit to find, the second time find the full number; if found at the
first time, then stop finding.
When the group ID of black white list is 253, this means “use CDB” as black white list. Find
according to the group ID of router related with group ID of CDB. Find in two times, the first
time reduce one digit to find, the second time find the full number; if found at the first time, then
stop finding.
When the group ID of black white list is 255, find database, caller convert. Default, name of
database list is cright, there are four fields: id, caller, called, and flag. Connect device to
database via another separate software “wcdrrx”. Device will send message to “wcdrrx” by
CDR port, “wcdrrx” will collect caller number from the received message, then find the field
“caller” in this database list, if matched, then send field “called” back to device, after received,
device will judge field “flag” to determine whether continue to carry out router, if continue, then
will convert the caller number from the calls to field “called”, and carry out route.
When the group ID of black white list is 252, find database, called convert. Default, name of
database list is cright, there are four fields: id, caller, called, and flag. Connect device to
database via another separate software “wcdrrx”. Device will send message to “wcdrrx” by
CDR port, “wcdrrx” will collect caller number from the received message, then find the field
“called” in this database list, if matched, then send field “caller” back to device, after received,
device will judge field “flag” to determine whether continue to carry out router, if continue, then
will convert the called number from the calls to field “caller”, and carry out route.
In configuration of “wcdrrx”, if parameter “RDSwapQuery” set as 1, then when finding, will
swap-find the caller number and called number. For example, when find called converter in
database, device send message to “wcdrrx”, “wcdrrx” collect caller number 8000 from the
received message, if there is a record in database: field “caller” is 8000 and “called” is 9000,
then matched successfully; send 9000 back to device, after received, device will judge field
“flag” to determine whether continue to carry out router, if continue, then convert the called
number from calls to 9000, and carry out route.
When use white list, field “flag” as 0 means: reject, 1 means: allow;
When use black list, field “flag” as 0 means: allow, 1 means: reject.
·SS7 Answer Signal: For some signal, whether parameter of charge has, for example, for
SS7, select “charge” to charger, or don‟t select.
·Action: Specify how to process the calls. There are several options as follows:
1) Trans to Truck (Transfer to PCM Trunk)
Transfer calls to PCM trunk to call out, there are several detailed cases. (6 cases)
When call-out, called number can be transmitted one by one, and also can be transmitted after
st
received completely. Therefore, there are two options: (1 ) “Trans to Trunk (Called number
nd
is transmitted one by one; „Ext Caller‟ as caller (Only for FXS call out)” and (2 )
“Trans to Trunk (Called number be transmitted after received completely; „Ext Caller‟ as
caller (Only for FXS call out)”. (Ext caller: external number)
For device which has analog lines (such as IAD, IPPBX and PBX), because each subscriber
has two numbers: External number and internal number, when call out, there need to select
rd
which number as caller number. Therefore, there are two other options: (3 ) “Trans to Trunk
(Called number is transmitted one by one; „Int Caller‟ as caller (Only for FXS call out)”
th
and (4 ) “Trans to Trunk (Called number is transmitted after received completely; „Int
Caller‟ as caller (Only for FXS call out)”. (Int caller: internal number)
th
(5 ) Trans to Trunk (Bind first): a special application. There can regularly assign one E1/T1
time slot channel to per analog line, when calls, firstly use the bind channel. (Usually used for
EM signaling as user side, used to correspond time slot)

th
(6 ) Trans to Trunk (By virtual caller): a special application. There can configure one virtual
number on each E1/T1 time slot channel, when transferring calls, find the corresponding time
slot channel according the virtual number, use the found time slot channel to call out. (Usually
it is used for exchange terminal of EM signaling and used to distribute calls)
For above six cases, the meanings of router content are same. Click the blue word in “Modify
Router” to set router. (Such as )
Router setting, specify the range of pcm trunk channel that used when call out.
A) Specify PCM; sequentially select the time slot channel from specified PCM to call out.
PCM ID, ID of PCM, for example, uses PCM 3, please fill with 3.
B) Specify Line Range, sequentially select range of time slot channels from specified PCM to
call out.
[Start-end), the starting channel is contained, but the ending channel is not contained.
Time-slot numbers of PCM are numbered uniformly. Time-slot number of PCM0 is from 0 to 31,
Time-slot number of PCM1 is from 32 to 63, and so on. For example, [start-end) set as [1-3),
then the used channel are two: 1 and 2.
C) Specify PCM group
Specifying PCM group is the best common router way, also flexible. In the PCM group you can
select rotating in group or sequence in the group, also you can specify the direction is
ascending or descending.
PCM group, specify a PCM group, there can be multiple PCM, unfold “Rule” and click “PCM” to
set the corresponding PCM.
·Rotating In Group: Select the time-slot in rotation from the specified range in PCM group,
select different time-slot per time.
·Sequence In Group: Each time always select the time-slot from the beginning according to
the sequence of member in PCM group, to find the free time-slot that can be used.
·Ascending: In specified range, from low to high, to find which time slot is free that can be
used.
·descending: In specified range, from high to low, to find which time slot is free that can be
used.
·Dual Seizure Control: Because the speed of PCM trunk call is very fast, if two sides select
one same channel to call at same time, then there is a crash, as dual seizure. To avoid dual
seizure, there should use some tactics. The normal usage is selection of odd line and even line.
For example, the local side select odd line has priority; the other side select even line has
priority, in this way, so that can avoid the dual seizure.
2) Trans to VoIP
Transfer calls to VoIP to call out. Only support SIP protocol. (PBX device don‟t have this
action)
For device which has analog lines (such as IAD, IPPBX and PBX), because each subscriber
has two numbers: External number and internal number, when call out, there need to select
which number as caller number. Therefore, there are two other options: “Trans to VoIP
(„ExtCaller‟ as caller (Only for FXS call out)” and “Trans to VoIP („IntCaller‟ as caller
(Only for FXS call out)”.
Router setting, specify the address and port to send calls. Click the blue word in “Modify Router”
to set router. (Such as, click “domain:0”; )
Address: IPV4 address or domain. When fill with domain, send calls according to the register
account in “server”, at this time, “port” is used to specify the order number of register account.
Attention: register account, it is in “VoIP->Server”, there can register into five servers.
When fill address with “127.0.0.1”, send calls to VoIP extensions of this device, there will find
the registered user in “VoIP User” of this device. (For IP PBX only, worked as sip server)
When fill address with “strk:xxx”, use the registered account from analog trunk IAD to call out;
xxx is registered account. (For IP PBX only, worked as sip server)
3) Trans to FXO (MG device don‟t have this function)

Transfer calls to FXO to call out.
Trans to FXO (Bind first), used in special situation. Assign one analog trunk line FXO for each
analog user line FXS, when call, firstly use the bind line. Usually it‟s used for bridge mode.
Router setting, specify the analog trunk FXO line that used to call out. Click the blue word in
“Modify Router” to set router.
(Such as, click “line 23”; )
A) Specify line; specify one FXO line to call out.
Line number, take number 0 as beginning, FXS and FXO uniformly numbering.
B) Specify Line Range; specify one range, in this range select one free line to call out.
[Start-end), the starting line is contained, but the ending line is not contained.
C) Specify Line Group, it is a common and flexible way.
Line Group, specify one FXO line group. There can be multiple FXO line in this group, unfold
“Rule” and click “FXO Group” to configure the FXO line that need to use.
·Rotating In Group: Select the line in rotation from the specified range in line group, select
different line per time.
·Sequence In Group: Each time always select the line from the beginning according to the
sequence of member in line group, to find the free line that can be used.
4) Trans to Wireless (MG device don‟t have this function)
Transfer call to wireless (GSM/CDMA/WCDMA) to call out.
Trans to wireless (Bind first), it is used in special situation. Assign one wireless line for each
analog user line FXS, when call, firstly use the bind line.
Router setting, specify wireless line that used to call out. Click the blue word in “Modify Router”
to set router. (Such as, click “line 0”; )
A) Specify line; specify one wireless line to call out.
Line number, from 0 as beginning, Wireless board is in sequence automatically from left to
right.
B) Specify Line Range; specify one range, in this range select one free line to call out.
C) Specify Line Group, it is a common and flexible way.
Line Group, specify one wireless line group. There can be multiple wireless line in this group,
unfold “Rule” and click “wireless” to configure the wireless line that need to use.
5) Trans to DTMF Key (MG device don‟t have this function)
Usually it‟s used for when call in from analog trunk FXO line or wireless line, make a secondary
transfer to the actual called number. Firstly device plays dial tone, and wait DTMF key from
users.
Selected “DTMF Key Trans”, and Click the blue word in “Modify Router” to enter “router
setting”.
·Wait Delay: Set waiting time for receiving DTMF key, if over this time, there will not wait. Unit
is millisecond.
·Max Length (maximal length): if set as 0, then process according to rule group that specified
in “Length Rule ID (Subsequent processing use)”; otherwise, process by this specified max
length. Recommendation value is 0.
·End key: Specify which key to end, usually set as #.
After DTMF Key was over, then convert number according to the rule that specified in
“Convert Rule ID (Subsequent processing use)” and process it according to the rule that
specified in “Router Rule ID (Subsequent processing use)”.
6) Trans to Computer Operator (OPR) (MG device don‟t have this function)
Device provides with a simple and automatically voice flow for telephone switchboard, also
can configure simple IVR voice flow.
Selected “OPR” and click the blue word in “Modify Router” to enter “router setting”. Router
setting: specify detailed flow parameters.
·Detect DTMF Key: If don‟t select, it will play voice when transfer. After played three prompt
voices, use original called number to match and carry out the three Rule groups that
subsequent processing use; if selected there will use the received DTMF key as new called
number.
·Wait Delay: Set waiting time for receiving DTMF key, if over this time, there will not wait. Unit
is millisecond.
·Max Length (maximal length): If set as 0, then process according to rule group that specified
in “Length Rule ID (Subsequent processing use)”; otherwise, process by this specified max
length. Recommendation value is 0.
·End key: Specify which key to end, usually set as #.
·Loop Times: Specify loop times of voice prompt. If over this times, and there is still no DTMF
key, end calls.
·Welcome Prompt Voice Index: Play one time only, usually content is „welcome to call
xxx(NICEUC) company‟.
·Main Prompt Voice Index: Repeatedly play, usually content is “prompt of DTMF Key”.
·Fail Prompt Voice Index: After DTMF Key, if the next process is failed, then there will play
prompt voice again.
When above prompt voice index are configured as 0, means „use the default flow of computer
operator‟. There are six prompt voices on OPR, content are as follow:
Number Content of Voice Example
0 Welcome prompt Welcome to call NICEUC company.
1 Operation prompt Please dial extension number, find number
please dial “0”
2 Busy prompt The line is busy.
3 No answer prompt There is no answer, please dial later.
4 Not existed number prompt The number you dialed is not existed, please
check it and dial.
5 Operation prompt after failed Please dial later or dial other number.
to transfer extension
When there is a call to come in, automatically answer it, and play welcome prompt voice,
next repeatedly play main operation prompt voice, at same time enable “detect DTMF
Key”, after DTMF key is ended, convert number according to the rule that specified in
“Convert Rule ID (Subsequent processing use)” and process it according to the rule that
specified in “Router Rule ID (Subsequent processing use)”. If router failed, will play “failed
prompt voice”, then back to flow, re-play main operation prompt voice; if router succeed, end
the flow.
·Related Operation: a special application in special case, it use to inform “Phone Link”.
7) Trans to Extension (MG device don‟t have this function)
When call in, find analog extension line or VoIP extension according to called number.
Router setting, each extension has two number, so need to select “Match Ext Caller” or “Match
Int Caller”.
When device find extension, use “suffix matching”, so that can match the case with area code.
For example, the called number of call in is 075526520000, if the internal number is 26520000,
this case can also be matched. There is no need to add area code “0755” for internal number.
8) Queuing (Enter to queue) (MG device don‟t have this function)
When call in, firstly let the call to queue, secondly assign to ring on extension according to
rules.
Select “Queuing” and click the blue work in “Modify Router” to enter “router setting”.
Router setting, specify the line member that need to queue.
·Line Group: Specify one FXS line group. There can be multiple FXS line in group. Unfold
“Rule” and click “FXS group” to configure FXS line group.
·All Ring: All of line members in group will ring.
·Only select the login user line: it is a special application in special case, and it is used to
login CTI Link or Phone Link, as agent in call center.
Default case, calls in queuing will be in waiting status until answer by extension. If select “Limit
Queue Wait Number”, there will calculate the quantity of extension that can be used, if the
waiting call in queue is over than the quantity of extension, then the exceeded calls can limit by
“Can Over Number”, if over this value, then reject the call.
When caller user is in queue and waiting for the answer from extension, there can play voice to
it, specify voice by “Waiting Voice Index”, and specify playing times by “Play Repeats”.
Queuing (Date Time), Carry out this router only in active date and time. Separately select
from Sunday to Monday, each day there are time periods can be configured.
For this function, firstly make the date time of device accurate, then device can judge whether
the call from this moment is in active time.
9) Function (MG device don‟t have this function)
Specify the operation function code of key on phone.
Usually set as “**”. Press key on phone, there can carry out the command of device. Here
need to configure one group ID of Convert Rule. In this group, there should have a rule which
delete “**”, so device can recognize the function operation code.
Detailed function of commands, please check Appendix 3.
10) Call Pick-up (MG device don‟t have this function)
Specify function key to pick up the call on other extensions of device.
Usually set as “*8”, use “*8” to pick up the call on other extensions. Here need to configure one
group ID of Convert Rule. In this group, there should have a rule which delete “*8”.
When there are multiple extensions which are ringing at same time, if want to pick up one
specified extension, please use “*X”, “X” is SID of the extension number, Beside need “convert
rule” to delete “*”, then device can recognize specified extension by the back number.
11) Trans to FXS (MG device don‟t have this function)
Transfer calls to specified FXS line.
Select “Trans to FXS” and click the blue work in “Modify Router” to enter “router setting”.
Router setting, specify the assigned FXS line when call in. There are three ways:
A) Specify line; specify one FXS line to call out.
Line number, take 0 as beginning, FXS and FXO uniformly numbering.

B) Specify Line Range; specify one range, in this range select one free line to ring.
C) Specify Line Group, specify multiple FXS lines that can be used.
Line Group, specify one FXS line group. There can be multiple FXS line in this group, unfold
“Rule” and click “FXS Group” to configure the FXO line that need to use.
·Rotating In Group: Select the free line in rotation from the specified range in line group, select
12) CTI Ctrl (Control) (MG device don‟t have this function)
Transfer calls to CTI (secondary development interface) to control.
Routers setting, transfer CTI Group ID to CTI, to distinguish different application.
13) Call in Holding (MG and IAD device don‟t have this function)
This is one special application in special case. Make a special processing to the calls from E1,
and carry out operations such as answer, or play voice, etc.
Select “Call in Holding” and click the blue work in “Modify Router” to enter “router setting”.
Router setting: specified detailed operation. Here can set “Answer/No answer”,
“Recording/No recording”.
·Play Silence: No voice can be heard.
·Play Ring Back Tone: The caller will hear ring back tone; the type of ring back tone can be
customized.
·Play Voice: Play the voice that specified by Voice Index; Play repeats: specify the time of
playing voice.
14) Conference
Configure parameters related with conference function.
Select “Conference” and click the blue work in “Modify Router” to enter “router setting”.
Router setting set detailed parameter of conference.
“The offset of the conference room number at the called number” and “The length of
conference room number at the called number” are used to collect the conference room
number. For example, “Offset” is 3, “Length” is 2, and user call 99902 to enter conference,
after offset, “999” is deleted, collect 2 lengths, so the room number is 02.
The mode of joining conference: three modes, “Talk in conference”, “Listen (only) in
conference” and “Speak (only) in conference”.
PBX device don‟t have the following three parameters.
Block DTMF: filter DTMF Key.
Trans coding, pass through coding.
Enhance, properly adjust parameter, to get better effect of conference.
15) Transfer to Trunk (Limit length of called number)
Transfer calls to PCM trunk to call out according to some length rule.
Select “Transfer to Trunk (Limit length of called number)” and click the blue work in “Modify
Router” to enter “router setting”.
Router Setting, Specify the length rule of number and the range of PCM trunk time-slot
channel that used for call out.
A) Specify PCM; sequentially select the time slot channel from specified PCM to call out.
PCM ID, ID of PCM, for example, uses PCM 0, please fill with 0.
B) Specify PCM group

Specifying PCM group is the best common router way, also most flexible. In the PCM group
you can select rotating in group or sequence in the group, also you can specify the direction is
ascending or descending.
PCM group, specify a PCM group, there can be multiple PCM, unfold “Rule” and click “PCM” to
set the corresponding PCM.
·Rotating In Group: Select the time-slot in rotation from the specified range in PCM group,
select different time-slot per time.
·Sequence In Group: Each time always select the time-slot from the beginning according to
the sequence of member in PCM group, to find the free time-slot that can be used.
·Ascending: In specified range, from low to high, to find which time-slot is free that can be
used.
·descending: In specified range, from high to low, to find which time-slot is free that can be
used.
·Dual Seizure Control: Because the speed of PCM trunk call is very fast, if two sides select
one same channel to call at same time, then there is a crash, as dual seizure. To avoid dual
seizure, there should use some tactics. The normal usage is selection of odd line and even line.
For example, the local side select odd line has priority; the other side select even line has
priority, in this way, so that can avoid the dual seizure.

st
·Called SID 1 (Set): It is the first SID of called number. Set/gather the possible number. For
example, fill with 25; called number with 2 or 5 should set/gather here.
st
·Called SID 1 is black list: if select, means “the first SID of the called number which is same
st
as called SID 1 will forbidden to call; otherwise, allow to call.
nd
·Called SID 2 (set): It is the second SID of called number, set/gather the possible number.
For example, fill with 37, means “SID of called number is X3 or X7 (X is the first SID)”.
Consider of the above example, here means “four SID: 23, 53, 27, 57”.
nd
·Called SID 2 is black list: if select, means “the second SID of the called number which is
nd
same as called SID 2 will forbidden to call; otherwise, allow to call.
·Length of called number must be equal or more than Minimal Length, and be equal or less
than Maximal Length.
16) Number Change Notes (Query DB) (For device PBX only)
A special application, it is with use of Changing number notification system.

Select “Number Change Notes (Query DB)” and click the blue work in “Modify Router” to enter
“router setting”.
When there is a call from E1, there can select “Answer” or “No Answer”, then send the
request “query Database” by the connection of CDR network, the Change Number Notification
System will query DB and feedback result; Device will play the corresponding prompt voice
according to the result. There are four prompt voices: two successful prompt voices, one
prompt voice for empty number, one prompt voice for failure, and here can set the times of
playing.
6.2.4 Router Line Group
Unfold “Rule”, there are group setting for each kind of line.
1) PCM Group
Set member of PCM group.
2) FXS Group (MG device don‟t have this function)
Set member of analog FXS line group.
3) FXO Group (MG device don‟t have this function)
Set member of analog FXO line group.
4) Wireless Group (MG device don‟t have this function)
Set member of wireless trunk line group.
Description for setting: group ID is fixed, there only need to set the members of this group.
Click in the list of member or click “Modify”, there will pop up a page “Member Setting”.
In page of “Member Setting”, members in group are on left, members out of group are on right.
Firstly select member, then click button “Add” to add, button “Delete” to delete, “Up” and
“Down” can adjust order of members.
6.2.5 Rule limit
1) White List
Only allow the numbers in white list to call.
2) Black List
Don‟t allow the numbers in black list to call.
One number can belong to maximal 10 groups, can batch add.
3) VoIP Call Restriction (PBX device don‟t have this function)
Restrict the source IP of SIP calls.
Received Address, Selected to examine source IP address of SIP message.
RTP Address, Selected to examine source IP address of media flow.
SIP From Address, Selected to examine the IP address of field „From‟ of received SIP
message.
White List of IP Address, List of allowed IP address, parameters are IP address and IP Mask.
IP Mask determines the range of IP address. For example, IP Mask: 255.255.255.255, this is
one single IP; IP Mask: 255.255.255.0, this is a range of IP address.
For example, if want to limit one single IP:192.168.1.5, fill IP address with 192.168.1.5, fill IP
Mask with 255.255.255.255; if limit a range of IP from 192.168.1.0 to 192.168.1.255, fill IP
address with 192.168.1.0 and IP mask with 255.255.255.0.
6.3 PCM
Click “PCM”, and set on right.
Clock source, specify the source of primary clock of PCM. If configure as “Internal (Free
Run)”, then device will provide with clock, but the accuracy of device is a little low, so usually
please select external clock. When select external clock, device will select clock from one
PCM port (PCM port is specified by configuration, usually it is PCM 0), then transmit clock to
other PCM ports on this device.
Clock Source from backplane, transmit clock by backplane bus when cascade of boards.
(PBX don‟t have this function)
When clock source changed, reset phasic, it‟s related with hardware, usually here is no
need to enable.
All E1/T1 mode, used for MG device MG1000 only.
PCM Synchronization detailed alarm: RED, RAIV, AIS, RMAIV, TS16AISV, and TS16LOSV.
PCM Mode: E1 or T1. (PBX don‟t have this function)
Attention: T1 should be supported by hardware. When use T1, parameters is different

in PCM Setting. There is no parameter of Impedance and CRC4, there will be a
parameter of Framing mode, quantity of time slot of T1 is 24.
PCM to packet, Adjust voice gain from PCM to packet. (PBX don‟t have this function)
Packet to PCM, Adjust voice gain from packet to PCM. (PBX don‟t have this function)
PCM option list, set related attribute of PCM.
PCM ID, ID number of PCM ports. Order of PCM port, please check the hardware device.
Impedance, it is line impedance. The E1 line use coaxial cable to connect, usually there use
two kinds of impedance: 120 ohm or 75 ohm.
"CRC4": check the cyclic redundancy of 4 bytes of physical frame. When connect with device
from foreign countries, usually there need to enable.
Framing Mode, used for T1 only, there are Super Frame (SF), Extended Super Frame (ESF),
Digital Multiplexer (DM), and Switch Line Carrier (SLC)-96.
Signaling Type, Set signaling type of PCM port. Now support the following kinds of signaling:
A) ISDN PRI, or Primary Rate ISDN (Primary Rate Access), also known as Digital One
(DSS1) signaling, in domestic usually use “30 B channels plus one D channel (30B + D)”.
B) SS7 signaling, SS7 signaling is a common international standard for common channel
(Common Channel) signaling system, which uses a layered functional structure and message
communication mechanism, the most suited for the use of modern digital communication
network. Currently device from NICEUC support TUP (Telephone User Part) and ISUP (ISDN
User Part).
C) Number one signaling, CAS (Channel Associated Signaling), called as R2 also, there are
mode: MFC and DTMF.
D) V5.2, connect AN (Access Network) access network and the LE (Local Exchange) the local
exchange network V-interface.
E) Q.sig, similar with PRI.
Click “Modify”, enter into setting page. Here can modify detailed parameter of signaling and
special application.
Attention: Modification of signaling will be effective after “write Rom” and reset.
Other function, set other special function.
Virtual Caller Function, device can provide with a lot of virtual caller number, they can used
for call in or call out, select the corresponding option to set. There are 4 using way, Fix Time
slot, map one -for-one to caller number according to time slot channel; Sequence, in whole
virtual caller list, take turns to select number in sequence; Random, in whole virtual caller list,
randomly select number; Time Slot Sequence, in whole virtual caller list, take turns to select
number in sequence, but should correspond with number of time slot.
Unfold “PCM” and click “Virtual Caller”, to configure virtual caller number. Here can add or edit
one by one, and can also add or edit in batch.
Bind SIP domain, (PBX don‟t have this function), can bind the status of PCM with SIP domain.
Binding Registered, when there is a problem with PCM, automatically log off service of SIP;
when PCM recover, automatically re-register SIP server. Vice versa, if SIP failed to register,
then disconnect PCM link; if SIP register successfully, then recover PCM link. Binding Option,
device sends message “OPTION”.
6.3.1 ISDN PRI
Click “Modify” and Select Signaling Type as ISDN PRI, Signaling Particular as follows:
Set Time Slot, set time slot of link; E1 mode, default is time slot 16; T1 mode, default is time
slot 23. Usually here it‟s no need to set.
Net Mode, there are two options: network side or user side. Net mode of two connected side
cannot be same.
Location, usually it‟s no need to set. Default is “2_Public network serving the local user”.
Over time of received (ms): When receiving number, if the number is passed by bit, will wait
this value of ms(millisecond) for next bit. If longer than this time and did not receive next bit,
then the number receiving has been completed.
Work Mode, usually choose "Normal". Mode “Listen” is used for signaling collection by
high-impedance; Mode “Record” is used for high-impedance recording. (MG device don‟t have
this function)
Phone Number Parameters, Here can set “Caller Screening Indicator”, “Caller
Presentation Indicator”, “Type of Caller Number”, “Type of Called Number”,
“Numbering Plan Identification” and “Transfer Capability”.
CLI sending control, is used for controlling whether send caller message or not.
Default, device will pass through the phone number parameters from above grade, if there is
no corresponding parameter from above grade, and then will use the value that specified here.
If selected “Force”, then device will not pass through the parameters from above grade,
directly use the value that specified here.
If there is something wrong with modification, please click “default” to recover default value, or
click “Normal” to use the normal value.
If there is a request that different phone number should use different type of caller number or
called number, then need to “Find in Number Plan Table”. Find the corresponding value
according to the configured SID in Number Plan Table.
Ring Message, usually use “ALERTING”, also can select “PROGESS” as Ring Message.
Local provide ring back tone (ring back tone don‟t pass through), Default, device will pass
through ring back tone from above grade; If selected this feature, then our device will provide
ring back tone.
Description for “When trunk side received ready message of media, and do not open the
media channel” and “Always open media channel when received call proceeding”, please
check the Appendix 3.
Bcc Preferred, Selected, and then device on other side will control and assign time slot
channel of calls.
Using Virtual Caller as Caller, When call in, according to caller number, find the
corresponding time slot channel in virtual caller table.
Using Virtual Caller as Called, When call out, according to called number, find the
corresponding time slot channel in virtual caller table.
Map Time Slot, (it is used for NC-AD300XD only, used for map time slot one-for one in MCU
control board and PCM trunk board).
6.3.2 SS7
When use SS7, firstly add SS7 Link parameters. Unfold “PCM” and click “SS7 Link” to
configure SS7 Link parameters.
Link ID, Index number of SS7 Link. When PCM use SS7, it is used for specifying link.
Link Type (Connect method), Set the carrying type of link. There are following several types:
·1_Hand Link, link from one time slot of E1/T1.
·2_Net Client, connect link from network client side.
·3_Net Server, connect link from network server side.

·4_Net Server and Client, connect link from network server+client side. Two connected
device use symmetric interconnection mode.
·5_SGW (Signaling Gateway), device work as server of transferring link, transfer link to
network.
·6_GWC (used for PBX only), Split one link into multiple links for subordinate device to use.
Link from network by UDP, When connect link from network, default use TCP; if selected this
item, then will use UDP.
Maximum length of UDP frame (It is used for NC-AD300D only), this function is used for
secondary development.
Application Type, there are two types: TUP and ISUP.
Over time of received (ms), if calling number is received by-bit, over this time the receiving
will be finished. Unit is millisecond.
Service Information Octet (SIO), SIO (8 bytes) contains SSF and SI.
SIO
Sub-service field（SSF） Service indication code （SI）
D C B A D C B A
00 International Network Spare 0000-SNMM Signaling network management messages
01 Spare for International 0001-SNTM Signaling network testing and maintenance
(for international use only) messages.
10 National Network 0010-Spare
11 Reserved for National 0011-SCCP (Signaling Connection Control Part)
use 0100-TUP (Telephone User Part)
0101-ISUP (ISDN User Part)
0110-DUP (call and circuit-related messages)
0111-DUP (facility registration and cancellation)
1000-Reserved for MTP Testing User Part
1001-1111-Reserved of no use
Here we can set SSF only, usually there don‟t modify it, Default is 80. When device is used in
other foreign countries (out of China), usually set it as C0 or 01.
SS7 link can build network, the common way is dual link to share the loading.
“Link parameters 1”, configure parameters of link 1; “Link parameters 2 (Used for link
group network)”, configure parameters of link 2.
Link Time slot, Number of time-slot that be used when connect link from E1/T1.which time
slots have the SS7 link.

Signaling Link Code (SLC), the logical link number, must be consistent with the SS7
equipment on other side. Click to Edit.
Sub Mail, When connect link from network, it is used for determining port of TCP service or
call port of client. When configure as “0” or “255”, it will automatically set, value of port is equal
to “7320+ link ID”; otherwise, value of port is equal to “7320+ mail ID”.
Signaling Gateway IP, it is IP address of sever side. Set IP address of server that client side
will connect with.
When Link Type is “2_Net Client”, device can connect two different servers at same time. “Sub
Mail” and “Signaling Gateway IP” in link parameters 1 will address to server 1. “Sub Mail” and
“Signaling Gateway IP” in link parameters 2 will address to server 2.
When Link Type is “3_Net Server”, device can not only work as server of SS7 link, but also can
work as client side to connect other server. “Sub Mail” in link parameters 1 will address to port
of TCP Service of SS7 Link Service. “Sub Mail” in link parameters 2 and “Signaling Gateway
IP” in link parameters 1 will address to other server.
Link Type “4_Net Server and Client” is a special way of “Net Server”. Configuration of
parameter of server and client are same as “Net Server”. Usually it is used for mode of dual
link, only symmetric inter-connection of two devices.
Link Type “5_SGW” is another special way of “Net Server”. Configuration of parameter of
server and client are same as “Net Server”. But this type supports that client side use different
OPC and DPC, like an agent server of link.
Link Type “6_GWC” is similar with opposite use of “5_SGW”. Firstly get link from network, then
transfer to E1/T1, then assign to next grade. Link time-slot is used for the connection with
device from next grade, “Sub Mail” and “Signaling Gateway IP” is used for getting link from
server, configuration way is same as “Net client”.
Primary link and secondary link of device itself, plus link of network connection, so one link
group supports maximal 4 links.
Signaling Point Code (SS7 signaling code), When fill, please pay attention to the conversion
between decimal and hexadecimal. Here provides with multiple formats, please select the
corresponding format according to the actual situation and fill.

(Bit) length, specify the length of the signaling point codes. Usually there use 24-bit in
domestic and international use 14-bit.
OPC, Origination Point Code; DPC, Destination Point Code; STPC, Signal Transfer Point
Code.
Phone Number Parameters, Here can set “Calling Party‟s Category”, “Caller Screening
Indicator”, “Caller Presentation Indicator”, “Type of Caller Number”, “Type of Called
Number” and “Transfer Capability”. Usually select the default value. If there is a request that
different phone number should use different type of caller number or called number, then
need to “Find in Number Plan Table”. Find the corresponding value according to the
configured SID in Number Plan Table.
Default, device will pass through the phone number parameters from above grade, if there is
no corresponding parameter from above grade, and then will use the value that specified here.
If selected “Force”, then device will not pass through the parameters from above grade,
directly use the value that specified here.
If there is something wrong with modification, please click “default” to recover default value, or
click “Normal” to use the normal value.
Caller add „F‟ and Called add „F‟, configure according to actual situation. Different request of
different PBX, if select, then use „F‟ to complete number.
Local provide ring back tone (ring back tone don‟t pass through), Default, device will pass
through ring back tone from above grade; If selected this feature, then our device will provide
ring back tone, rather than pass through the next level.
Description for “Do not open the media channel of SIP when the media channel of trunk
side is ready”, Please check Appendix 3.
“Transmit the extension information in the signaling message”, (used for PBX only), pass
through message when signaling convert from E1/T1 to E1/T1.
this function)
After configured the parameters of Link, please come to configuration of parameters of
signaling on PCM port. Click “PCM” and “Modify” on right, and then select SS7 in “Signaling
Type”.
Link ID, the Link ID in this PCM.
CIC(Circuit Identification), circuit number of each time slot in E1, should be same as the
other side. This CIC number is the order number of PCM in one link ID, numbering of time slot
in PCM will be automatically increased in sequence. Numbering of time slot in PCM is equal to
“CIC multiply 32 then plus number of time slot. For example, CIC set to 0, then time slot 0-31 in
this PCM will be from 0 to 31; if the CIC set to 1, then the time slot 0-31 in this PCM will be
numbered from 32 to 63; if the CIC set 2, then the time slot 0-31 in this PCM will be numbered
from 64-95; and so on.
Blocked Time Slot, Specify which time slot is blocked, don‟t use.
6.3.3 R2 (/CAS)
Click “PCM” and “Modify” to enter “PCM Setting”. Select “CAS/R2/E&M” at parameter
“Signaling Type”. Two ways: MFC and DTMF
If use Chinese standard R2 signaling, configuration is very simple. After selected “R2”, then
click “MFC” and select “Chinese R2”, there is no need to configure other parameters.
Besides, our devices support two other standard R2 signaling: Syrian R2 and ITU-R2.
If the R2 signaling is non-standard, please select “DL Setting (ABCD bit)” to set each value
for line signaling; if select “DL Mask Enable”, then here can limit the valid value of “ABCD bit”
by “DL Mask”, to filter the unnecessary bit.
this function)
“Call ID Enable”, selected and asks the other side to send caller ID. Position, specify when
„received called number of which position‟, to ask caller ID. For example, position is 2, 6123
call 7890, the order of receiving number is 7, 8, 6, 1, 2, 3, 9, 0. If we put caller number at front,
and called number at back, separate them by “->”, then the procedure of receiving number
are as follows:
->7
->78
6->78
61->78
612->78
6123->78
6123->789
6123->7890
Play busy tone by local, used for special case, if device on the other side cannot play busy
tone, selects it, and then our local device will play busy tone.
If use DTMF, the configuration is a little complex, usually used for special case, such as E&M
signaling.
PCM MUX, used for fixing connection mode, let E1 use as analog line, extend analog line by
E1.
User Side, selected, device will work as terminal. In Router rule, configuration should be
“when off-hook, directly use E1 time slot to connect end office PBX, dialing by user will process
by end office PBX”. In Router rule, use “x” as SID; in “Analog Line”, fill “FXS Dial Next Key
Over” and “FXS Play Busy Tone Duration” with 0, then will achieve “when off-hook, directly use
E1 time slot”. If user line want to use fixed E1 time slot, there need to configure analog user
line, in router rule, enable “Transfer to Trunk (Bind first), and bind the corresponding E1 time
slot, so analog line will correspond with E1 time slot one-to-one.
Send DTMF Called, when used as user side, device will send called number to end office PBX
by DTMF (only used for device PBX only). If selected, there is no need to use “Off-hook
directly use E1 time slot”, let local device firstly receive called number and send to end office
PBX.
Received DTMF Called, when don‟t select “user side” and device will use as end office PBX,
after selected “Received DTMF Called”, used for defect DTMF key from user side, and receive
dialing number from user side.
Send Call ID by In-band, when don‟t select “user side” and device will use as end office PBX,
after selected “Send Call ID by In-band”, will send caller number of FSK or DTMF to user side
by voice channels.
Using Virtual Caller as Caller,
1. Device as user side, when call in, according to time slot channel, find the corresponding
phone number in virtual caller table as caller number of call-in.
2. Device as end office PBX, when call in, according to time slot channel, find the
corresponding phone number in virtual caller table as caller number of call-in.
Using Virtual Caller as Called, Device as user side, when call in, according to time slot
channel, find the corresponding phone number in virtual caller table as called number of
call-in. It is equal to hot-line number, usually don‟t use.
Map Time Slot, (it is used for NC-AD300XD only, used for map time slot one-for one in MCU
control board and PCM trunk board).
6.3.4 V5.2
When use V5.2, firstly please configure link parameter of V5.2. One single unit device supports
maximal one V5.2 interface and two links.
Unfold “PCM” and Click “V5.2” to configure related parameters.
V5 ID and V5 Variable are the main parameters of V5.2 interface; they must be same as
opposite side.
Interface; there are two types of interface, AN side and LE side. Interface must be different as
opposite side. If opposite side work as AN side, then our device must work as LE side; if
opposite side work as LE side, then our device must work as AN side.
Access network (AN), it is an abbreviation from the Service Node Interface (SNI) and the
associated user-network interface (UNI) between the number of transmission entities (such as
the line facilities and transmission facilities), is for the supply of telecommunications services
to provide the necessary carrying capacity the implementation of the system. One AN can be
reused, exchange connectivity and transport functional components.
LE is a local switch acronym, it‟s a subscriber line through the AN terminating switches.
Logic C Path ID, one main parameter of V5.2 link, must be same as opposite side.
Protection Link Switch time, when there is a failure with primary link, and failure time is over
the time that configured here, there will automatically switch to secondary link, Unit is
Millisecond. “Polarity Reversal” can exactly judge whether calls are connected or not
connected.
When as LE side, device can control the way of sending CID when call to AN side. Default,
send by FSK, if selected “Disable FSK”, then don‟t send FSK caller. Select “Enable INIT
CLIP” or “Enable STB CLIP”, to send caller number by way of signaling.
L3 address, is EFaddr types of PSTN signaling or control protocol message in the third layer
of the address. The aim is for the user port or public control functions provide a unique
reference.
Unfold “PCM” and click “L3 Addr (Address)”, to see L3 Address Table.
One Phone number is corresponding to one address. Click "Phone Number" column to add
number, or edit number by clicking the existed number. After edited, please press “Enter” to
finish modification. Click "Batch Set" to set serial number in batches.
After configured parameters of V5.2 interface, come to configure signaling parameters on
PCM. Click “PCM” and “Modify” to enter “PCM Setting”. Select “V5.2” at parameter “Signaling
Type”.
Specify primary link, secondary link or no link on this PCM port.
Logic ID (CIC), identity V5 within the PCM-link a logical sequence of numbers, both sides
must be consistent.
When PCM carry primary link or secondary link, Blocked time slot: designated which time
slot is C-channel. When you select no link, useless slots can be blocked.
6.3.5 Q.SIG
Configuration of its parameters are same as ISDN PRI, please see the description of ISDN
PRI.
6.3.6 Virtual caller
Device provide a group virtual caller number, it can be used for in-bound or out-bound on
E1/T1. Totally support 1280 virtual caller number. At first, set numbers in virtual number table,
secondly, set the using way on PCM.
Unfold “PCM” and click “Virtual Caller”, then set virtual caller number on right. Click "Phone
Number" column to add number, or edit number by clicking the existed number. After edited,
please press “Enter” to finish modification. Click "Batch Set" to set serial number in batches.
Unfold “PCM” and click “Modify” in “PCM option list”, and enter “PCM Setting”, to set “Virtual
Caller Function” in “Other Functions”.
Virtual Caller Function, device can provide with a lot of virtual caller number, they can used
for call in or call out, select the corresponding option to set. There are 4 using way, Fix Time
slot, map one -for-one to caller number according to time slot channel; Sequence, in whole
virtual caller list, take turns to select number in sequence; Random, in whole virtual caller list,
randomly select number; Time Slot Sequence, in whole virtual caller list, take turns to select
number in sequence, but should correspond with number of time slot.
6.3.7 Number Plan
When signaling is ISDN PRI or SS7, sometimes the type of number needs to modify.
Number Plan, it is used for setting organization way of numbers. If different number need
different number plan, at this time need to find the number plan table. In Type of Caller or
Called Number, select “Find in Number Plan Table”, means to find Number Plan Table.
Unfold “PCM”, select “Number Plan”, and click button “Addition” to add SID and type for a
number.
Signaling Type, specify this rule which is used for SS7 or PRI.
SID, prefix of called number.
Type of Number, for SS7, there are following options:
·TUP
0, Subscriber number;
1, Spare, reserved for national use;
2, National (significant) number;
3, International number;
·ISUP
0, Spare;
1, Subscriber number (national use);
2, Spare, reserved for national use;
3, National (significant) number;
4, International number;
For PRI, contains two parts, Type of Number (high 4 bits) and Numbering Plan Identification
(low 4 bits).
·Type of Number:
0, unknown;
1, international number;
2, national number;
3, network specific number;
4, subscriber number;
6, abbreviated number;
7, reserved for extension;
·Numbering Plan:
0, unknown;
1, ISDM/telephony numbering plan (Recommendation E.164[19]/E);
3, Data numbering plan (Recommendation X.121 [21]);
4, Telex numbering plan (Recommendation F.69 [22]);
8, National standard numbering plan;
9, Private numbering plan;
15, Reserved for extension;
6.4 Analog Line
Set related parameters for Analog Line. (MG don‟t have this function)
Now there are 6 kinds of analog lines, Internal line/Extension (FXS), External Line (FXO),
High-impedance Recording, Push To Talk (PTT), E&M and Magnet.
Firstly, click “Read RAM”, secondly unfold “Device Details” and click “Hardware resources”,
and then look for type of analog line. Usually analog line use a card with 4 lines, different types
use different colors to distinguish.
In this picture, the first card is FXS, the second card is FXO, the third card is High-impedance
Recording, the fourth card is E&M, the fifth card is PTT, the sixth card is Magnet, the seventh
card is forced FXS, the eighth card is forced FXO.
Unfold “Analog Line” and select “line-characteristic”, click “Modify” in parameters table of
analog line on right, to configure the correct type.
Device will automatically recognize the line module when booting. If failed to automatically
recognize, then it will initialize according to configured type. If you are using a Force type, such
as "FXS (forced)", it will not automatically recognize and directly initialize according to
configured type.
Click “Batch Set”, to enter “Line Batch Setting”, the first option is “Line Type”, here can set
analog line in batch.
Next, separately describe each line type.
6.4.1 FXS
FXS, External line, used for connecting phone, usually there is feeder.
6.4.1.1 User Number
Each line has two caller numbers, External caller number and Internal caller number. External
number is usually used for external calls, length of number is maximal to 20 digits; Internal
number is usually used for internal calls, length of number is maximal to 8 digits.
Unfold “Analog Line”, and select “Line-characteristic”, set in line type table. Click “Modify” of
each line, to set number for corresponding line; if numbers are continuous, please click “Batch
Set”.
In “Line Batch Setting”, select “Int Caller” or “Ext Caller”, set start and end of line range, fill “Int
Caller” or “Ext Caller” with the first number, click “OK”, then the numbers will be automatically
in ascend order.
If here fill “step value of change” with minus number, then the numbers will be in descending
order.
6.4.1.2 User Group
Unfold “Analog Line”, and select “Line-characteristic”, please click “Batch Set”.
Assign extension line into different companies and departments.
Extension lines under one same company can call each other; extension lines under different
companies cannot call each other. Extensions can be picked up by users only from one same
department, extension cannot be picked up and transferred by users from different
departments.
Company ID, fill with number value, same number value as one group.
6.4.1.3 Call Priority
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set call priority.
There are two priorities, low priority and hight priority. Users can switch “Use high priority” or
“Use low priority” by phone command or modification of configuration.
More big value, more high priority; user line can use all of router rules that its own priority is
more than router priority.
For example, there are 4 route rules for extension, priority value of these 4 rules are0, 2, 5, 7;
Low priority of extension is 3, high priority is 11; when extension use low priority, it can only use
the front two rules; when extension use high priority, it can use all of 4 rules.
6.4.1.4 Out Gain and In Gain
Click “Analog Line”, here can set out gain and in gain of global line.
Out Gain corresponds to adjust the volume that heard by other end, In Gain corresponding to
the volume of local end.
Out Gain (Output Volume), adjust volume that heard in receiver of phone.
For PBX, value range is from 0 to 25. Number 0 is default number 15 as -5.00 DB. Number
from 1 to 25, it‟s at the beginning of -12.00 DB, change is based on gain of 0.50 DB; the
maximal value is 0.00 DB.
For IAD and IPPBX, value range is from 0 to 2000. Number 0 is default number 1460 as -3.500
DB. Number from 1 to 2000, it‟s at the beginning of -76.450 DB, change is based on gain of
0.50 DB; the maximal value is 23.500 DB.

In Gain (Input Volume), adjust volume that spoken in microphone.
For PBX, value range is from 0 to 25. Number 0 is default number 7 as 3.00 DB. Number from
1 to 25, it‟s at the beginning of 0.00 DB, change is based on gain of 0.50 DB; the maximal
value is 12.00 DB.
For IAD and IPPBX, value range is from 0 to 2000. Number 0 is default number 1880 as -0.400
DB. Number from 1 to 2000, it‟s at the beginning of -94.350 DB, change is based on gain of
0.050 DB; the maximal value is 5.600 DB.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set “Out Gain”
and “In Gain” of each line.
Out Gain (Output Volume) of each line
For PBX, value range is from 0 to 25. Number 0 is default number 15 as -5.0 DB. Number from
1 to 25, it‟s at the beginning of -12.0 DB, change is based on gain of 0.5 DB; the maximal value
is 0.0 DB. When fill with -1 or 255, means “Use global set”.
For IAD and IPPBX, value range is from 0 to 200. Number 0 is default number 146 as -3.50DB.
Number from 1 to 200, it‟s at the beginning of -76.00 DB, change is based on gain of 0.50 DB;
the maximal value is 23.50 DB. When fill with -1 or 255, means “Use global set”.
In Gain (Input Volume) of each line
For PBX, value range is from 0 to 25. Number 0 is default number 7 as 3.0 DB. Number from 1
to 25, it‟s at the beginning of 0.0 DB, change is based on gain of 0.5 DB; the maximal value is
12.0 DB. When fill with -1 or 255, means “Use global set”.
For IAD and IPPBX, value range is from 0 to 200. Number 0 is default number 188 as -0.40 DB.
the maximal value is 5.60 DB. When fill with -1 or 255, means “Use global set”.
Attention: When modify volume, it’s better that change of increase and decrease is based on
default value, no much changes to avoid voice distortion or “cannot hear”. Modification will
be effective after “Write RAM” and reboot.
6.4.1.5 User Password
When use phone commands or Phone Link, there need password to verify. Password is made
of four digits, fault is 0000.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set password for
each user.
6.4.1.6 Dial Function
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set some related
functions for dialing.
Usually, when dialing, by pressing key “#” to end dialing, “#” will not be sent out. If here select
“Always bypass “#”, then it will bypass “#” also.
When select “Bypass “#” when SID is “*” or “#”, device will automatically judge the number
that dialed by user, if the SID is “*” and “#”, will bypass “#”, other case will not bypass.
Forbid call extension, to forbid the function of “extension call each other”, this extension
cannot call other extensions; also other extensions cannot call this extension.
6.4.1.7 Service Functions
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set service
functions. (Unfold button on right of “Service Function”) Because of less use, default these
parameters will be hidden.
1) Hot Dial
After off-hook, user want to automatically call one number, please select this function, and fill
with the number.
In new version, only filled with hot line number (no need to select it), then enable hot-dial
function. After off-hook, there will wait for one time (time is decided by “FXS Dial Next Key
Over”), and call hot line number. If during the waiting time user dialed, then will not carry out
function of hot-dial. If selected hot dial and filled with number, there will be no wait after
off-hook and immediately call hot line number.
2) Call forward
There are 3 types for call forward, Call forward without any condition, Busy forward and No
answer forward.
Select the way of call forward that want to set, then fill with corresponding number.
Call forward without any condition, any call to this line will automatically forward to this “call
forward number”.
Busy forward, Call forward when the line is busy: only when this line is busy, call will forward
to this number (Busy forward number);
No answer Call forward: only when there is no answer on this line, call will forward to this
number (No answer forward number).
For these 3 types of call forward, if call-out from E1, when using ISDN PRI or SS7, there is
special format of call message. For default, there contains three numbers in message, caller
number (original caller number), original called number (user number of this line (select
internal number or external number according to router)), called number (the number which
will forward to).
Click “Analog line”, there are parameters that can specify the number in message.
When select "Call transfer use original called number as the caller number", in message
there only use two numbers: caller number (user number of this line (select internal number or
external number according to router)), called number (the number which will forward to).
For default, when user make a flash to call forward, the call-out message use two numbers:
caller number(internal number of this line) and called number(the number that user dialed after
flash).
When select "Call transfer always use the Ext caller as the caller number", when call
forward, the number of call-out will all force to use the external number of this line.
When select “Call transfer always use the Int caller as the caller number”, when call
forward, the number of call-out will all force to use the internal number of this line.
3) Do not Disturb
When don‟t answer any calls, please enable function of “Do not Disturb”. After selected “Do
Not Disturb”, this line will be busy.
4) SMS Function
This function is based on hardware; firstly there should be wireless (GSM/CDMA/WCDMA)
module which is inserted with SIM card on the device, and the SIM card can normally receive
and send message.
There are three types of SMS notices, Send message when no answer, Send message
when busy and Send message when block. Firstly selected them, then fill with mobile phone
number that needs to inform in “SMS alert number”.
5) Bind Line
When call out from analog FXO trunk, wireless trunk and E1 trunk, there can use the function
of "Bind line".
Firstly, select the line type that want to bind, then fill with the bind line/channel number. Each
user calls out from a stationary external line. This is used for special case, usually used for
bridge-connection mode.
Bridge-connection, disconnect the original line, and insert our device among the original line,
use our device to bridge-connect the original line, at same time add other lines to call out, take
away the call traffic. Special case is bridge-connection of analog line, for example, Line 0 is
FXO, connect with the original FXO trunk line of user, line 16 is FXS, connect with the original
extension of user, E1 connect with new operator line. When user call out from E1 and other
user call this user from analog FXO, for user, not only keep the original number, but also save
the expense of calls.
If select “Bind Line (FXO) (Bridge-connection of analog line), when internal line is busy, then
will block the bind analog FXO line, to forbid the call-in. When internal line call out from the
bind FXO line, default, the action of “Press flash” on internal line will pass through into FXO
line, it means that there will be an action “flash” on analog FXO trunk line; if selected “Disable
flash pass to FXO”, during calls, the action of “Press flash” on internal line will not pass
through into FXO line, then it will process by local side, such as call hold, or call forward, etc. If
selected “Disable flash process”, then don‟t carry out any process like call hold/call forward
when detected flash.
6) Ring-back
For calls from E1 or VoIP, if media is passed through, then here can play voice as ring-back
tone. The ring-back tone should be loaded into voice database of device in advance, order
number of voice file is as voice index number.
Enable function of ring-back tone, firstly select “Ring back Play Voice”, then fill with voice
index number in “Voice Index”. “Play repeats”, repeat times to play voice, if selected “Infinitely”,
then will play all the time, and it will not stop until the call is answered.
Besides, unfold “Analog Line” and click “Tone Setting”, to set global ring back.
7) Recording
Recording function is based two conditions, license of recording and run service software of
recording. Then configure recording function “enable recording” to recording.
Unfold “System” and click “Function”, to set global recording. “Enable Auto Record”, device
can make recording for all calls (except the calls between extensions). “Enable (Including
Extensions between)”, this will make recording for calls of extensions.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set recording for
each line.
“Auto recording (Enable)” and “Auto recoding (Disable)”, can be used together with global
parameter “Auto record”.
8) Call waiting (PBX don‟t have this function)
When analog user line FXS is busy in call, if there is a new call to call this FXS, under the case
that there don‟t enable of function “call waiting”, this new call will directly back because of busy.
If enable call waiting, then the new call will be holding on and waiting, the busy line will heard
reminding tone.
Click “Analog Line”, to set global parameter “FXS Call Waiting Enable”.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to control “call
waiting” for each line.
6.4.1.8 Signal Cycles

Adjust parameters of analog line, so that device can adapt the working environment better,
and work stably. Unfold “Analog Line” and click “Tone Setting”, set parameters on right.
For internal line FXS, there are 3 “Signal Detected Cycles”, unit is millisecond.
Pick up, fill with 0, it means „use the default value 50‟; over this value, the line will pick up.
Flash, fill with 0, it means „use the default value 70‟; Value 1 to 9 mean ‟do not detect flash‟,
Value over 10 mean ‟detect flash‟. The time range to detect flash is from flash cycle to hang up
cycle.
Hang up, fill with 0, it means „use the default value 300‟; over this value, the line will hang up.
Such as Hang Up 300 ms means that after 300ms the line will hang up.
Take default value as an example, after phone picked up about 50ms, line pick up; if the time
of flash on phone is more than 70ms, and less than 300ms, then line will detect as flash; if the
time is less than 70ms, there is no detection; if the time is over 300ms, then line will hang up.
When internal line FXS pick up and dial, device will wait for user to press key. Click “Analog
Line”, there are two parameters can set.
FXS Dial first Key Over (Over time for FXS dialing first key): if the user picks up phone and
over this time there is no dial, the line will be busy. Setting it as value 0, it means there will
always wait and no busy. Usually set as 15000ms.
FXS Dial next Key Over (Over time for FXS dialing next key): if during the process of
dialing, and over this time there is no next dial from user, then there will judge the dialing is
over. Setting it as value 0, it means there will always wait and no busy. Usually set as 5000ms
When the FXS Dial First Key Over and FXS Dial Next Key Over are all set to value 0 at the
same time, it means that the call will be sent out immediately when picking up, at this time in
Router rule, SID should use "x".
FXS Play Busy Tone Duration, when FXS is disconnected, if user don‟t hand up, then will
heard busy tone, this parameter specify the duration time to play busy tone. Value 0, don‟t limit.
Otherwise, there will limit the duration time, over this time, it will be silence.
6.4.1.9 Ring Mode
Device can specify ring mode under different calls. Unfold “Analog Line” and click “Ring
Mode”, set parameters on right.
There are following several cases of call sources (call sources: where is the call from):
Call out back, the call is from interior of system, such as the call controlled by CTI.
Call from PCM, call from PCM trunk.
Call from FXO, call from analog trunk FXO.
Call from FXS, call from internal extension FXS.
Other undefined calls, please set according to “Default”.
Duration time of ring can be different under different modes. Value 0 as default 25 seconds, the
maximal value is 199 seconds, usually fill with 40 seconds.
6.4.2 FXO
FXO is used for connecting phone cable, like telephone control analog phone line. Usually it is
called as analog trunk.
6.4.2.1 Signal Detection

Click “Analog Line”, set related parameters on right.
FXO Receive Cid Idle Over (Idle time of rings that FXO receive CID): When there is a call
on analog trunk (FXO), the device will detect its ringing and receive Caller ID (CID). If the
interval time that FXO ring is over this time (which has became a normal ringing), there will
judge CID had been already sent. If less than or equal 200ms, there will not detected FSK CID.
Recommended value is 2000. Fill it with value 0, use default value 2200.
FXO Receive Cid Ring Times (Times of ring that FXO receive CID): If times of rings that
FXO receive CID is over this times value and have not yet received CID, the line will not wait.
Recommended value 2, if fill it with 0, use default value 2.
FXO Wait Dial Tone Seconds (Time that FXO wait for dial tones) : When call out through
analog trunk (FXO), there need to detect the dial tone to call out. If over this time, there have
not yet detected dial tone, then stop to detect; If this value is less than 2000 milliseconds, then
there don‟t detect dial tone, line will call out after specified delay time.
FXO Auto Answer Delay (Delay time that FXO automatically answer) : when call out FXO,
if this value is 0, it means there don't detect polarity reversal until connected . Otherwise, there
will be a delay to connect. For example, as here, fill with 6000, then the call will be connected
after the delay of 6000ms.
FXO Auto Detected Interval (cycle that FXO automatically detect): here can set a interval
time to detect FXO lines at fixed time. Value 0, there is no need to detect.
FXO Busy Tone Circles (Circles that FXO detect busy tone): until signal of busy tone reach
up at this cycle, it will be recognized as busy tone.
Unfold “Analog Line” and click “Tone Setting”, in “Signal Detected Cycles”, Ring, and fill with 0,
use default value 4500 milliseconds. When there is a call-in from FXO, and there is no answer
of this call, FXO line is ringing, if caller hang up, FXO line will stop to ring, device will detect
ring after over this time, if there don‟t detect ring, judge that caller had already hanged up.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set in “Analog
line parameter setting”.
Pulse Dialing, when call out from FXO, send number by pulse dialing. (PBX don‟t have this
function)
Polarity reversal signal detection, when FXO Auto Answer Delay set as 0, if selected this
parameter, it‟s necessary that there don't detect polarity reversal until connected; if don‟t select,
polarity reversal signal is not necessary, when there don‟t detect it, there will be a delay to
connect.
Non Tonal Detection Enable, if there is no polarity reversal signal, there can judge whether
called answer or no answer by Non Tonal, this is better than “wait for delay to connect”, but it‟s
not too accurate. (PBX don‟t have this function)
6.4.2.2 Detect DTMF CID
When line type is FXO, can specify whether detect DTMF CID, if don‟t detect DTMF CID,
then default is “detect FSK CID”.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set in “Analog
line parameter setting”.
When need to judge Open Key and Close Key of DTMF CID click “Analog Line” to set “DTMF
CID Flag”. Open Key and Close Key are selected only from “ABCD*#”, and use one digit of
them. Usually open key is C, close key is D.
6.4.2.3 Out Gain and In Gain
Click “Analog Line”, here can set out gain and in gain of global line. (For FXO, only PBX device
have global parameter)
Out Gain, adjust volume that heard in FXO when call is from FXO.
For PBX, value range is from 0 to 25. Number 0 is default number 15 as -5.00 DB. Number
from 1 to 25, it‟s at the beginning of -12.00 DB, change is based on gain of 0.50 DB; the
maximal value is 0.00 DB.

In Gain, adjust volume that heard in FXS when call is from FXO.
For PBX, value range is from 0 to 25. Number 0 is default number 7 as 3.00 DB. Number from
1 to 25, it‟s at the beginning of 0.00 DB, change is based on gain of 0.50 DB; the maximal
value is 12.00 DB.
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set “Out Gain”
and “In Gain” of each line in “Analog line parameter setting”.
Out Gain (Output Volume) of each line
For PBX, value range is from 0 to 25. Number 0 is default number 15 as -5.0 DB. Number from
1 to 25, it‟s at the beginning of -12.0 DB, change is based on gain of 0.5 DB; the maximal value
is 0.0 DB. When fill with -1 or 255, means “Use global set”.
For IAD and IPPBX, value range is from 0 to 200. Number 0 is number value 94 as -3.50DB.
the maximal value is 12.40 DB. When fill with -1 or 255, means “don‟t set”.
In Gain (Input Volume) of each line
For PBX, value range is from 0 to 25. Number 0 is default number 7 as 3.0 DB. Number from 1
to 25, it‟s at the beginning of 0.0 DB, change is based on gain of 0.5 DB; the maximal value is
12.0 DB. When fill with -1 or 255, means “Use global set”.
For IAD and IPPBX, value range is from 0 to 200. Number 0 is default number 115 as 0.00 DB.
the maximal value is 12.75 DB. When fill with -1 or 255, means “don‟t set”.
Attention: When modify volume, it’s better that change of increase and decrease is
based on default value, no much changes to avoid voice distortion or “can not hear”.
Modification will be effective after “Write RAM” and reboot.
Remind: Sometime FXO can not FSK CID, possible reason is gain of FSK signal is too
small or too big, please properly increase or decrease “Out Gain” and “In Gain” of
FXO.
6.4.2.4 Service Functions
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set service
functions. (Unfold button on right of “Service Function”) Because of less use, default these
parameters will be hidden.
1) Hot Dial
For FXO, there is a call to call in, line will ring. Device recognizes call-in by detecting ring,
therefore there is no actual called number. Hot Dial, it‟s equal to specify a called number.
When device detected FXO ring, if configured hot dial, then will call hot line number.
Enable this function, please select “Hot Dial” and fill with hot line number.
2) Trunk Centrex
A special application, achieve a function “multiple lines use one number only” or “Bind multiple
FXO line into one number only”.
When device use analog trunk to access, sometimes user need to announce one number only,
if analog trunk work as Centrex /PABX etc, which the line has function of extension of PBX,
such as pick up or transfer. So that device can achieve this function.
Assume that, originally user has 4 external lines, external line numbers are 27398000,
27398001, 27398002 and 27398003, and the corresponding extension number is 8000, 8001,
8002 and 8003. Before don‟t use our device, user can only use these 4 telephone lines to
make calls, there are only 4 extensions, the calls among extensions are free (Usually calls
among extensions are free). Along with addition of person, 4 extensions are not enough,
therefore it‟s necessary to install device to add quantity of extension, at same time user hope
to announce one number only 27398000, take it as number of switchboard, customer (caller
from PSTN) only need to dial 2739800.
For example, there are 4 FXO lines and 28 FXS lines on device. If 4 external lines connect with
4 FXO line according to number order, 27938000 connect FXO 0, 27938001 connect FXO 1,
27398002 connect FXO 2, 27398003 connect FXO 3.
4 external FXO lines use one external number only by Trunk Centrex. There are two ways to
achieve this function.
1. by pick up on extension
There should make following configuration on device. Unfold “Rule” and click “FXO Group”, in
FXO group, assign 4 FXO lines into one same group. For example, Group ID 0 as follows:
At same time, please pay attention to line order in group, the FXO line of switchboard number
should be in the first row as line 0.
For the FXO line 0 worked as switchboard number, please select function “FXO queue”
(Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set “FXO
queue” in service functions), and fill “Ext Caller” with the key for pick up from original
extension, such as fill with “ *72# ”. Ext caller of other FXO lines don‟t fill with it, should be
empty.
After finished configuration, “Write ROM” and reboot device, to let configuration be effective.
When there is a call to switchboard number 27398000, device will use other free FXO lines to
pick up switchboard line, and let switchboard be free, so that switchboard can receive calls
again, it‟s equal to four lines share one switchboard number. Besides, other three external
FXO lines can be directly called; there is no effect for them.
2. by transfer on extension
There should make following configuration on device. Unfold “Rule” and click “FXO Group”, in
FXO group, assign 4 FXO lines into one same group. For example, Group ID 0 as follows:
At same time, please pay attention to line order in group, the FXO line of switchboard number
should be in the first row as line 0.
For the FXO line 0 worked as switchboard number, please select function “FXO queue”
(Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set “FXO
queue” in service functions), and don‟t fill with its “Ext Caller”, let external number of FXO Line
0 be empty. For other FXO lines please fill their “Ext Caller” with original extension number,
such as 8001, 8002, 8003, etc.
After finished configuration, “Write ROM” and reboot device, to let configuration be effective.
When there is a call to switchboard number 27398000, device will transfer call to other free
FXO lines, and let switchboard be free, so that switchboard can receive calls again, it‟s equal
to four lines share one switchboard number. Besides, other three external FXO lines can be
directly called; there is no effect for them.
6.4.2.5 Refuse Call-in
Click “Analog Line” to set.
FXO Call-in be refused and set busy, When FXO line receive call, FXO line will ring, at this
time it‟s impossible to refuse call with no answer. After selected this function,if enable this
function, firstly device will pick up this FXO line, secondly hang up, in this case there can
refuse call.
6.4.3 Other Line Types (PBX don‟t have this function)
6.4.3.1 PTT lines
PTT is shorthand of “Push to Talk”. (Also known as Press-to-Transmit, is a method of having
conversations or talking on half-duplex communication lines, including two-way radio, using a
momentary button to switch from voice reception mode to transmit mode.)
PTT card is designed and manufactured by NICEUC Company; we own proprietary intellectual
property rights. It is a card that make NICEUC gateway achieve function of RoIP (Radio over
IP). PTT card provides with 4 lines RoIP; each RoIP line supports standard audio input/output,
and PTT controls level to input/output; it is used for connecting Vehicle Radios or Digital Mobile
Radios.
PTT card can be used in devices NC-MG320W, NC-MG320, NC-MG640, NC-MG930, and is
compatible with any user slot of them; together with FXS, FXO, GSM/CDMA/WCDA, E1, E&M,
SIP and so on, incorporate powerful and systematic integrated access gateway system (or
IPPBX system).
6.4.3.1.1 Look at position of PTT lines
When there is a PTT card in devices, you could enter into web configuration page of our
devices, and check where PTT lines are by following steps: unfold “Device Details”, and click
“Hardware Resources”. PPT lines are purple with one capital work “P”. PTT (Push To
Talk/RoIP) in purple)
As above picture shows, PTT lines is in slot 2 of device NC-MG930. PTT line orders are: 32, 33, 34, and
35.
6.4.3.1.2 Configuration
On web page, under “Analog line” to configure global parameter of PTT lines.
Take above picture as an example.

PTT Silence Detection Cycle (Downlink): If here configure less than 100ms, there will disable
this detection function. Default is 400ms, means if over 400ms, there will judge the PTT line is
silent.
PTT talk control key definition: through key to control PTT talks. “*=ON”, means key “*” is used
to obtain rights of talks; “#=OFF”, means key “#” is used to release rights of talks; “*1*ON”,
means key “*1* is used to obtain rights of talks; “*2*=OFF”, means key “#” is used to release
rights of talks.
PTT OFF detected release delay: delay how long to release talks.
PTT ON protection time: for example, here fill with 300seconds, when time of PTT ON is over
300 seconds, there will automatically release talks, to protect and avoid burning-out vehicle
radio station.
In chapter 2.2.1, there show PTT line orders are: 32, 33, 34, and 35.
Under “line-characteristic”, find number 32, 33, 34, and 35, then click “Modify” to modify
configuration of each PTT lines.
As above picture show, Default “Active low” (Active low level), LED lights “OUT” AND “IN” on PTT panel
will be off.
Hot dial: fill with 9000, (Any FXS extension number of MG930 is ok).
6.4.3.1.3 Rule configuration
Under “Rule”, make groups: FXS and PTT.
Click “Router Rule ID 1” to configure router rule of FXS group
Click “Router Rule ID 2” to configure router rule of PTT group.
Make 32 connect with 33; under this case, LED lights “IN1、OUT1、IN2、OUT2” are off; at this
point, call 9999 from 9001, 9000 will ring and answer; then LED lights “IN1” and “OUT2” on
PTT panel will be on; when 9000 and 9001 talk, 9001 press “#” to release rights of talks, then
LED lights “IN1” and “OUT2” on PTT panel will be off; 9000 press “*” to obtain rights of talks,
then LED lights “OUT1” and “IN2” will be on.
When 9000 and 9001 talk, 9000 press “#” to release rights of talks, then LED lights “OUT1”
and “IN2” will be off.
6.4.3.2 E&M
At present, E&M module can be used in product NC-MG320, NC-MG640 and NC-MG930.
E&M (earth & magneto, or ear & mouth) is a type of supervisory line signaling that uses DC
signals on separate leads, called the "E" lead and "M" lead, traditionally used in the
telecommunications industry between telephone switches.
E&M was originally developed to allow PABXs in different geographic locations to
communicate over an analog private circuit. Some digital interfaces such as Channel
Associated Signaling also use versions of E&M signaling. E&M is considered an obsolete
technology for new installations, which generally use Basic Rate (BRI) or Primary Rate (PRI)
digital interfaces.
Signal Interface
Our E&M module can be worked as both PBX and TLE (The Line Equipment) sides.
Variants of E&M
Supports three variants of E&M: Type I, Type II and Type V.
When variant is I or II, the connected devices on two sides must be PBX and TLE (The Line
Equipment). When Variant is V, there is no request for the connection device on two sides,
they can be same or can also be not same; to unify, when variant are these three types, the
connection device on two sides are PBX and TLE.
Number of Wires: 2-wires and 4-wires.
4-wire E&M uses a 4-wire (2-pair) transmission path for the voice signal. 2-wire E&M uses a
single pair for both transmit and receive voice signal.
Configuration of “2 line EM or4 line EM” is as same as the end device.
Signal Mode: E&M, ADASE and DTMF (DTMF: immediate start, wink start and delay start).
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set signal mode
in the page “Analog line parameter setting”.
Configuration of Signal mode is as same as the end device.
The configuration of type “PBX” / “TLE” is for DTMF only.
E&M
DTMF PTK
E&M DTMF defines three methods of “start” signaling: wink start, delay start and immediate
start.
Wink Start - when the originator goes off hook, the other end transmits a short (140-290ms)
off-hook signal and returns to on-hook. The originator detects the wink and then sends the
dialed digits. The other end goes permanently off-hook (seized) when the call is answered.
Delay Start - the originator goes off-hook, waits a pre-defined delay, and then checks for
on-hook from the other end before sending the digits.
Immediate Start - The originator goes off hook, waits 150ms and then sends the dialed digits.
On configuration, there can specify delay time, unit is millisecond. DTMF PKT mode, there
need to configure “Virtual Caller Offset” and “Terminal ID”, if work as PBX side, please
configure “Signal ID” also.
·Under DTMF mode, E&M module work as PBX side, When there are incoming calls, use
number of Virtual Caller “by Virtual Caller Offset + by Terminal ID” as called number. When
there is no configuration of virtual number, please use hot-dial number as called number.
·Under DTMF mode, E&M module work as TLE side, when there are incoming calls, use
number of Virtual Caller “by Virtual Caller Offset + by Terminal ID” as caller number. when
there is configuration of virtual number, directly use internal number as caller
number
DTMF mode, one TLE can connect with multiple PBX. In one same group, each PBX has
different SelfID, but has same PEER ID. PEER ID is the SELF ID of TLE.
The TLE and PBX in one same group, they should use same VCBASE (by Virtual Caller
Offset).
In one same group, there is one TLE only, but there can be multiple PBX.
ADASE
Trunk-ADASE is a 4-wires (transmission pair & receiver pair) E&M trunk for two-sided use with
INBAND signaling. This is a standard from Russian Power Energy Department.
Please pay attention to the configuration of parameter “hot dial”.
When call-out, if there is no called number, or the called number is equal to default hot dial
number, then use “DISPATCH” mode, and don‟t send the called number. Otherwise, send the
called number by in-band signaling, don‟t send caller number.
When call-in, under “dispatch” mode, because there is no information of called, use hot dial as
called number.
Out Gain and In Gain of E&M
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” and choose “EM” in
Line Type in the page “Analog line parameter setting”, then adjust out gain and in gain of each
E&M line.
Out Gain (Output Volume), adjust volume that heard in receiver of phone.
Value range is from -18 to 6. Number 0 is default number 15 as -11.00 DB. Number from 1 to
49, it‟s at the beginning of -18.00 DB, change is based on gain of 0.50 DB; the maximal value
is 6.00 DB.
In Gain (Input Volume), adjust volume that spoken in microphone.
Value range is from -14 to 10. Number 0 is default number 15 as -7.00 DB. Number from 1 to
49, it‟s at the beginning of -14.00 DB, change is based on gain of 0.50 DB; the maximal value
is 10.00 DB.
6.4.3.3 Magnet
Unfold “Analog Line”, and select “Line-characteristic”, please click “modify” to set use way of
magnet line in the page “Analog line parameter setting”.
Ring disconnect, after detected ring, then disconnects.
Alter after disconnect, Alarm after disconnect.
Ring Count, count of ring.
Ring Interval, Interval time of ring.
6.4.4 Module Configure (PBX don‟t have this function)
Click “Analog Line” to configure.
FXS power save disabled (disable mode of FXS power save), FXS power check disabled
(disable mode of FXS power check), FXS load patch disabled (disable mode of FXS load
patch) and FXS broadcast disabled (disable of FXS broadcast), these four parameters are
decided by the hardware functions related with FXS. Usually, don‟t set.
Hot Plug Detection Cycle, how many interval times to detect hot-plug. Usually configure as 5
seconds.
Adjust the line order; to be compatible with previous device, so we add this parameter, it is
used for adjusting numbering order of line.
0, to disable.
3, each group with 4 lines become reverse order, such as previous order is 0, 1, 2, 3, 4, 5, 6, 7,
after become reverse order, the order is 3, 2, 1, 0, 7, 6, 5, 4;
"15": each group with 16 lines become the reverse order, such from 0, 1, 2 …14, 15, after
became reverse order is 15, 14, 13 … 2, 1, 0.
FXS Ring Power Type, fill with 0 to use DC 48V and AC 12V, fill with 1 to use AC 60V.
FXS Disable Caller ID, if selected, internal FXS extension will not send caller ID.
Long Line setting: sometimes user line need to extend hundreds and thousands kilometers,
at this time need to use the long line mode to let FXS module work normally. Select “Used long
line”, then there are options for module:
"0.0": the first module of the first user board; "0.1": the second module of the first user board;
"1.2": the third module of the second user board; Increase progressively.
6.4.5 Ring-back Tone
User can make customized ring-back tone according to the following steps:
1. Convert the audio file to the database file that device can use, more details please check the
Appendix 3;
2. Load the audio file into device; more details please check the Appendix 3.
3. Configure parameters, unfold “Analog Line” and click “Tone Setting”.
Device can specify the ring-back tone under three kinds of calls. "Ext", circumstance of calls
from external „FXO‟, „E1‟ or VoIP‟, "Int", circumstance of internal calls among FXS each other,
"Trans", circumstance of call transfer by flash hook;
When the specified voice index number cannot find in the device, device will automatically use
the standard ring-back, therefore, if don‟t want to use music as ring-back tone, please fill
“Voice Index” with -1 or 255.
6.4.6 Tone Setting (PBX don‟t have this function)

Each different countries has different standard of tone. There are four standards from USA,
India, China and Russia. Click “System” to configure. If use standard from other countries,
please select “Custom”.
Usually, tone contains Dial Tone, Busy Tone, Ring back Tone and Congestion Tone.
Unfold “Analog Line” and click “Tone Setting”, to configure the custom tone.
Generation Custom is used for „Device play and send tone‟; Detection Custom is used for
„Device detect the received tone‟.
6.5 VoIP (PBX don‟t have this function)
At present, VoIP calls only support SIP protocol.
Click “VoIP” to configure.

Codec supported by our device：G.723, G.729, pcma (G.711 Alaw), pcmu (G.711 uLaw),
G.726-16, G.726-32, GSM, GSM-EFR, AMR, AMR-WB, iLBC.
On left, they are the enabled codec; on right, they are the disabled codec. Priority of codec is in
sequence from up to down, click “Up” and “Down” to change priority of codec.
Codec G.723, rate supports two kinds 5.3k and 6.3k.
Codec G.726, reorder supports IETF and ITU.
iLBC, rate supports two kinds 13.3k and 15.2k.
Packet Interval (FPP), configure interval time that RTP send packet, unit is millisecond.
DTMF, specify which way to transmit DTMF, there are three ways: INBIND, RFC2833 and
SIPINFO; INBIND, transmit in bind; SIPINFO, transmit by sip information. Pay attention to the
value of DTMF Payload must be the same as the opposite side. Usually, use RFC2833,
Payload use 101.
RTP Mode, set up function of penetrating private network, there are two modes: signal and
recved. "Signal", don't penetrate (all are in public network, please use signal), "recved",
penetrate, (when the opposite side is private network, use recved). When configure as
"recved", the equipment will send data back according to the received data packet.
Check RTP Source IP Addr and Port: check the source IP Address/Port of RTP packet; to
strictly check the source address and port of RTP, then exclude the illegal media data.
RTP Broken Second: in a specified period of time, when there is no RTP data, it will
automatically disconnect. "0" means do not enable this feature.
Voice Activity Detection, in order to save the bandwidth traffic, there increase a function of
voice activity detection; enable this function, device will automatically do compression of mute.
Configure parameters according to requirements.
Echo Cancellation, Network Echo Cancellation Tail Length and Echo Path Filter Length,
they are used for configuration of echo cancellation.
IP Re-trans (IP Re-transmit): When device transmit calls to VoIP, after transmitted message
INVITE, if in the time configured in Re-trans Timer, there don‟t receive the response message,
then device will re-transmit. If there is still no response, then add the times of re-transmission;
please configure the times in “Re-trans Max”. If the system is failed to send call 4 times (the
interval time will increase 500ms per time) from the first route, then the system will select the
second (or spare) route.
Jitter Buffer: buffer the voice jitter; Configure parameters, so that you can adjust the voice jitter.
Delay Min: the minimum time of delay;
Delay Max: the maximum time of delay;
Delay Init: the initial value of delay;
Adaption Period: Adaption Period;
Deletion Threshold: delete the limit of threshold.
6.5.1 Server Parameter
Configure related parameters of gateway, gatekeeper or sever. Unfold “Analog Line” and click
“Server” to configure. Now our device can register into 5 SIP servers at same time, 5 SIP
servers are from domain 0 to domain 4.
Domain, fill with IP or domain, following with colon to add number of port, for example,
ippbx.org:5068. If there is no port number, default is 5060.
User name, or Display Name.
Password, used to register and authenticate.
Authorization User Name, used to authorize user name to register and authenticate.
Option User Name, used to send message “OPTION”.
Keep alive, keep activity how long, it‟s used for test of network, unit is millisecond. If select
“Reg”, regularly send empty message. If select “Opt”, there will regularly send message
“OPTION”.
Re-registration interval, it is Interval time of re-registration. Set “Regist (registration)
Expires”, if expired, will re-register. When register, there will receive refreshing time of
registration that requested by server. Device takes the minimal time as refreshing time.
Fail Retry, if failed to register, here specify interval time to retry registration.
Local port, set the local network port that used by SIP protocol, UDP and TCP are signal port
of SIP, and RTP is media port.
STUN (Session Traversal Utilities for NAT) Server: the address and port number of STUN
server that used in network. In order to solve the problem of establishment of communication
(for H.323/MGCP/SIP traversing NAT) and as called, client side will send STUN Server an
inquiry about their own address after conversion, the STUN server will return the source
address in the received request packet as the information to client side, so the client side get
the address information after conversion. Then, the client side can directly fill with this
converted address in the source address of the protocol packet of H.323/MGCP/SIP,
meanwhile can also directly register this converted public IP address when is registering on
terminal.
Out Call Proxy Server, the address and port number of out call proxy server that used in
network.
URL Mode, set format of SIP message.
Tel format, use format <tel:XXXXXXXX>.

RequestUri
Caller Display Name
FromURL
Called Display Name
ToURL
Contact
Enable Privacy: header (RFC3323), used to limit caller display, for more information,
please check RFC3323.
Enable History: header, used to carry the original number.
Disable redirection, used for NAT, usually doesn‟t use this item.
To Uri as Old called, used to identify the non-standard call transfer.
Add P-Asserted-Identity; add related information of P-Asserted-Identity into message.
Add user=phone, add user=phone into message.
SIP Encrypt; use a particular encryption algorithm to encrypt the SIP protocol or RTP voice
data.
When register failed, use the second route, if there configured second route, after failed to
register, then it will not try to call again and directly use second route.
IP to IP calls, proc (process) by application layer, generally, please select. Otherwise the IP
calls will not process by application layer, it will directly transfer and process by MSP.
IP to IP calls, RTP peer to peer, for calls with both sides are IP, default, media is transited via
server; after select this item, media will transmit from peer to peer, and not via server.
180 message with SDP, when use 180 as ring-back message, usually it will not take media
information; after select this item, 180 message will take media message like 183.
Response ALERTING always use 180, force to use 180 messages as ring-back message.
Fixed DTMF Payload, not automatically adapt, when the value of DTMF Payload is different
on both of caller and called side, device can automatically adapt; if selected this item, force to
use its own configured value, and don‟t mind the configure on opposite side.
Reliable provisional response (PRACK), if opposite side send a request „100rel‟, then
device use „PRACK‟ to response.
INVITE without 100rel, (need to “Write ROM and reboot” to be effective).
Check the SIP from IP, for calls, strictly checks the IP address where is it from in SIP
message. (For device IPPBX only)
Only registered user can call, strictly limit calls, only the registered user can call. (for device
IPPBX only)
Always request authentication for calls, strictly limit calls, each call response 401, and
request authentication. (For device IPPBX only)
Enable BLF server, enable function of BLF. (For device IPPBX only)
Remote side is on NAT mode, when there is one user side from one private network to register,
please enable this function. (For device IPPBX only)
SIP From with Port, add colon and port number behind the „From address‟ in SIP message.
(For device IPPBX only)
SIP to with Port, add colon and port number behind the „To address‟ in SIP message. (For
device IPPBX only)
6.5.2 FAX
Configure related parameters of fax. Unfold “Analog Line” and click “Fax” to configure.
Fax Mode, T.38, By Pass (/Pass through), T.38 Signal By Pass.
Fax Protocol, udp local TCF and udp transferred TCF.
Udp EC, usually configure as 1.
Fax use different RTP Port, RTP port of fax is different with RTP port of voice. After selected,
when switch VBD, RTP port will change, usually increase by 2 from the original port value.
Local, use own port; Remote, use port from opposite side.

Auto (Automatically) switch to VBD, Switch to VBD after V.25, VBD Detail, Switch VBD not
send reINVITE, Switch to T38 only detected CED, different case enable different options.
FAX connection speed limit (limit connection speed of fax), set the connection speed of fax.
T.38 packet loss concealment, enabled, replace bad lines by empty (white) lines or last good
(correct) lines.
Prevent ECM faxes in T.38, usually don‟t select.
Disable T.38 spoofing Mechanism, usually don‟t select.
Fax Switch Delay, used for NAT case, delay how long to switch fax, send a period of voice
packs to activate NAT. Range is from 10 to 2000ms.
Fax Active event (fax activation detection), set trigger condition for recovery of calls after fax
ended. Several signals: Local CNG, Remote CNG, Local ANS, Remote ANS, Local V21,
Remote UDPTL.
Recover talk after fax (FAX Receive), used for when receive fax.
Recover talk after fax (FAX Send), used for when send fax.

6.5.3 VoIP User
VoIP user, or SIP Subscriber, this function is for device IPPBX only.
Device support many accounts to register as terminal, maximally, there can register 1024
terminal accounts into our device.
Select “Register All Phone”, to enable the registration function that allow VoIP user to register
into SIP sever. Under some cases, need to select “Call to register not use contact”, in the
register message that sent, will not use field “contact”.
For IAD, “Contact Phone” is used to associate VoIP user account with analog extension
phone.
For IPPBX, when local use as SIP server, fill “SIP Server Address” with “*”, to assign account,
SIP terminals which register into can be as VoIP extension. Here can configure “Internal
caller”, “External Caller”, “Company group”, “Department Group” and “priority
value/class” for VoIP extensions, they have same functions like analog extension. There is no
use for “contact phone”

6.6 Wireless Line
Configure and look up wireless line (GSM/CDMA/WCDMA, etc) on device.
Click “Wireless”, to configure related parameters on right.
Wireless incoming ring detection cycle, (how long detect ring per time) when call in from
wireless trunk, detect whether there is ring signal at this interval time, to judge ring end or not.
Value 0, don‟t detect; Recommended value is 9000. Unit is millisecond.
Wireless dial out wait ring back delay, select value from options or fill with value, unit is
millisecond.
6.6.1 Mobile Information

Unfold “Wireless” and click “Mobile Information” to look up detailed status of wireless line.
IMEI, abbreviation of International Mobile Equipment Identity, it is mobile equipment
international identity number, and made of electronic serial number with 15 digits of numbers.
IMSI, abbreviation of International Mobile Subscriber Identification Number, it is a symbol to
distinguish mobile users, store in SIM card, used to distinguish effective information of mobile
user. The length of IMSI is not over 15 digits, use number “0~9”, made of “MCC+MNC+MSIN”.
MCC is country code of mobile user in 3 digits, for example, Chinese MCC is 460; MNC is
number of mobile network, maximal made of 2 digits, is used to identify mobile communication
network of mobile user; MSIN is identity number of mobile user, is used to identify mobile user
in one mobile communication network. MSIN is made of “CC+MOM1M2M3+ABCD”, CC is
assigned by different operators, there is corresponding relation between M0M1M2M3 and
H0H1H2H3, four digits ABCD can be freely assigned.

SMSC, abbreviation of Short Message Service Center, show number from short message
service center.
Carrier, Carrier name of SIM used now.
Status, registration status of SIM, can check it is local or roaming.
Signal Level, signal lever of wireless line; more bigger value, more stronger signal.
LAC, abbreviation of Location Area Code, it is one area that set for paging and cover one
geographical area. Early, divide by administrative regions, now more flexible, divide by paging
traffic. When traffic paging in one LAC reach to early-waring threshold, then must divide. In
order to make sure the location of mobile station, each cover area of GSMPLMN are divided
into many location areas, LAC is used to mark different location areas. One location area can
contain one or more communities.
Cell, base station unit under location areas.
Operator Select, if SIM card support the switch between multiple operators, then can select
operator by clicking “Operator Select” as follows:
After switch operator, device will remember option of this time.
6.6.2 BCCH Show

BCCH: Broadcast Control Channel.
Unfold “Wireless” and click “BCCH Show” to specify BCCH that used by wireless line.
Click “Refresh”, to immediately refresh show. If want to change BCCH, please click “Detail”
behind of each line to enter modification page as follows:

Choose specified BCCH and click “Lock”, after successfully locked, the locked BCCH will
change into red as follows:
Click “Unlock”, recover to freely choose, device will dynamically and automatically choose the
BCCH that can be used.
6.6.3 Out Gain and In Gain of Wireless line
Click “Wireless” to set out gain and in gain for global wireless line.
Global Parameters:
Speak Out Gain, fill with -1 or 65535, to use default value 2; 0 is 0DB, 1 is as -6DB, 2 is -12DB,
3 is as -18DB.

Mic In Gain, fill with -1 or 65535, to use default value 0. 0 is as 0DB, 1, 2, 3, 4, 5, 6, 7 are as
6DB, 12DB, 18DB, 24DB, 30DB, 36DB, 42DB.
Unfold “Wireless” and select “Port Configuration”, then click “Parameters Modify”, here can set
out gain and in gain for each line. Value range is same as global parameters.
6.6.4 Call Back
When internal extension A call external number B and there is no answer or busy, device can
automatically record extension number A and the unconnected number B, in the life time that
kept for call back, if detect there is call in from external number B, then device will directly
transfer call to internal extension A. Otherwise process the call in normally.
How to enable this function, please take the following steps:
Firstly, unfold “Wireless” and select “Port Configuration”, then click “Parameters Modify”, to
choose “Hot dial”; secondly, click “Wireless”, to set “Call Back Deep”; value 0, to disable this
function, the maximal value is 100.
Call Back Keep Life, specify the life time that kept for call back.
Prompt: choose “Enable” and “Disable”, to automatically finish the above steps.
6.6.5 Hot Dial
Wireless line is similar with FXO line, where there is a call-in, line will ring. Device recognizes
the call-in by detecting ring; therefore there is no actual called number. Hot Dial, it‟s equal to
specify a called number. When device detected call in from wireless line, if configured hot dial,
then device will directly call hot line number.
Firstly, unfold “Wireless” and select “Port Configuration”, then click “Parameters Modify”, to set
“Hot dial” in the page “Mobile line setting”. Choose “Hot dial” and fill with the number that want

to call.
6.6.6 IMI and Use Limit
Lock network /IMI, limit the network number of SIM card that used in device. If specified,
there can install and configure the SIM cards only which are correspondent with network
number. Unfold “unfold “Wireless” and select “IMI”, configure on right.
Bind IMEI, limit device can only use the specified SIM card, to prevent illegal misappropriation
of device. Unfold “Wireless” and select “Port Configuration”, then click “Parameters Modify”, to
set “Bind IMEI” in the page “Mobile line setting”. Fill “Bind IMEI” with the last 10 digits of IMSI.
Lock SIM card / PIN, limit SIM card can only use in this device, to prevent illegal
misappropriation of SIM card with special charge rate. Unfold “Wireless” and select “Port
Configuration”, then click “Parameters Modify”, to fill “PIN” with correct PIN of SIM card in the
page “Mobile line setting”.
Some commands for setting that can be used when connected from Telnet:
Set PIN, command: gsmpwd (port, “old password”, “new password”)
Lock, command: gsmlck (port, “password”)
Unlock, gsmunlck (port, “password”)
6.6.7 Wireless Switch-board Number
When use mobile/wireless number as access number, there can public one mobile number
only, and use multiple line together for access. How to enable this function; please check the
following detailed steps:
Public number of the first SIM card, Enable “Call forward” on the line of this SIM card, for
other lines, fill “External Caller” with it own mobile number of SIM card. Besides, these several
cards used for wireless switch must be in one same group. Unfold “Rule” and select “Wireless
Group”, add these lines into Group ID 0.
After accomplished configuration, when there is a call on the first SIM card, will transfer to the
second SIM card; if the second SIM card is busy, will transfer to the third SIM card; Only all of

other lines are busy, the first SIM card will be used.
6.6.8 SMS Operation
If want to send and receive SMS, then there need to configure parameter IMI. Unfold
“Wireless” and select “IMI”, configure parameter according to operator of wireless access.
(Click “Addition” to add)
Only SMS center number is correct, there can receive and send SMS; if not clear about it,
please ask operator.
1) Send SMS
Unfold “Wireless” and select “Send SMS” as following picture:

There can select many lines, each line is divided by comma. Click “Add Line” to look for which
lines can be used, choose the lines that you want to use, and click “OK”, the chosen lines will
be automatically add into “Line”.
To, destination mobile phone number that you want to send SMS. There can add multiple
destination mobile phone number, divided by semicolon.
Message, it is content of SMS.
Finally, click “Send” to send SMS, after wait for a while, there will return the result of sending.
2) Receive SMS
Unfold “Wireless” and select “SMS Received” as following picture:
Select “Line Number” and click “Refresh, to look for content of SMS that received by this line.

6.6.9 Command Operation
There can operate to send command AT, it‟s used for debug.
Unfold “Wireless” and select “Command Operation” as follows:
Select “Line Number” and enter command AT into “Command”, click “Send” to carry out the
command, “Result” is showed below. Click “Refresh” to refresh result.
6.6.10 USSD Operation
By command USSD, to carry out some functions, such as check balance of SIM card. This
feature should be supported by operator.

Unfold “Wireless” and select “Send USSD” as follows:

Fill with USSD Request command into the specified line; if all of line should carry out same
command, please select “All”, fill with USSD request command and click “Copy to selected”
(copy to the line that selected), then completely fill with command into the lines that are
selected. Click “Clear all” to clear out all request empty. Click “Send” to carry out command.
Result will be showed on raw “Reply”. Click “Refresh Results” to refresh raw “Reply”.
Unfold “Wireless” and select “USSD Fee”, there can set related parameters for USSD to
automatically recharge.
USSD query period, how long to query balance one time, unit is second.

Balance inquiry USSD command, command to inquiry balance. Each operators has different
commands, subscriber need to fill with it own-self.
Get value from offset of reply message, offset specified position from the reply message of
USSD command, and get balance value. Usually, one Chinese word occupies two bytes.
Recharging threshold, when balance is lower than threshold value, there carry out
recharging command.
Prepaid recharge USSD command, USSD command to recharge. Each operators has
different commands, subscriber need to fill with it own-self.
6.6.11 Call Statistic
Unfold “Wireless” and select “Call Statistic”, to look up call situation and balance minutes.
6.6.12 Fee Operation
Unfold “Wireless” and select “Port Configuration”, click “Rotation Control”, make configuration
in the page “Mobile line setting”.
Limit Call Duration (Per Call), there will enforce to cut call if the call duration of per call is
over configured time. Attention: please select “Call cut in limit (Per Call)” also.
Limit Call Duration (Total), if total call duration is over this configured time, this line will be
forbidden to use.

Alarm Threshold, when balance of call duration is less than this configured time, there will
send alarm by SMS to mobile phone.
Auto Reset, automatically reset remaining call duration, means that get total call duration of
GSM line. Automatically reset by time or by after (call duration is ) used over.
Reset Period, by date, by week, by month.
Next Reset time, specify time for next resetting.
Or manually do resetting operation by button “Fee Restore”.
6.6.13 SIM Rotation
Unfold “Wireless” and select “Port Configuration”, click “Rotation Control”, make configuration
in the page “Mobile line setting”.

SIM Slot Enable, enable SIM card slot.
Switch to next SIM when current SIM detect failed, when failed to detect current SIM,
automatically switch to next SIM.
Block when current SIM detect failed, when failed to detect current SIM, this current SIM will
be blocked.
Switch to next SIM when current SIM register failed, when failed to register current SIM,
automatically switch to next SIM.
Block when current SIM register failed, when failed to register current SIM, this current SIM
will be blocked.
Send SMS Notify when be locked or register failed, when this current SIM is locked or
failed to register, next SIM will send notify to specified number by SMS.
Switch to next when consecutive failures, when SIM failed to make consecutive calls
(usually two calls), automatically switch to next SIM.
Block when consecutive failures, when SIM failed to make consecutive calls (usually two
calls), this current SIM will be blocked.
Before using the SIM card must be activated, if selected, SIM cannot be used until activate.
VAD Enable, if selected, device will enable VAD detection.
Change IMEI by SIM switch count, when switch count reach to configured count,
automatically change IMEI. Attention: please select “Auto create IMEI “.

Chapter 7 System Function
7.1 SIP-T / SIP-I
Unfold “System” and click “Function” to set.
Function of SIP-T, can carry message data of SS7 ISUP in SIP message. If select "Pcm->VoIP", so
the message packets of SS7 ISUP received from E1 will converted to text strings, and carry it into
SIP message. If select "VoIP->Pcm", the message of SS7 that collected from SIP message, will
forward into E1.
7.2 Media Control of Ring-back Tone
Device can provide with local ring back tone. When use the signaling ISDN PRI or SS7, there
can configure to play the ring back tones by local, rather than pass through the ring back tones
from the next level.
If use the signaling ISDN PRI, click “PCM” and “Modify” on right, choose “ISDN PRI” in
Signaling Type in page “PCM Setting”, then select “Local provides ring back tone” as follow:
If use the signaling ISDN PRI, unfold “PCM” and select “SS7 link”, then click “Modify” to enter
page “Modify the Link”, then select “Local provides ring back tone” as follow:
In some special application, sometimes there should open media channel in advance.
For signaling ISDN PRI, default, when transfer the calls from VoIP to PCM trunk, if received
message "CALL PROCEEDING", and there contains parameter "Media is ready" in the
message "CALL PROCEEDING", then please open the media channels at VoIP side; if there is
no parameter "Media is ready" in the message "CALL PROCEEDING", then there don't open
the media channels at VoIP side until there received the message of "ALERTING" or

"PROGRESS". If select “When trunk side received media ready message, do not opening
the media channel”, even there contains parameter "Media is ready" in the message "CALL
PROCEEDING", there will open media channel at VoIP side after received message
“ALERTING” or “PROGRESS”. If select “Always open media channel when received CALL
PROCEEDING”, there will open media channel at VoIP side once received message “CALL
PROCEEDING”, rather than open media channel after received message “ALERTING” or
“PROGRESS”.
For signaling SS7, default, after transferred the calls from VoIP to PCM to call out, immediately
open media channel at VoIP side, rather than wait for message “ACM”. If select “Do not open
the media channel of SIP When the media channel of trunk side is ready”, there will not
open media channel at VoIP side until received message “ACM”.
7.3 Close Early Media
For SIP calls, in some cases, if the other side cannot process early media, and ask our device
don‟t reply message of "183", only reply message of "180".
If use signaling ISDN PRI, please select “Local provides ring back tone” and “When trunk
side received media ready message, do not opening the media channel”, to achieve this
function.
If use signaling SS7, please select “Local provides ring back tone” and “Do not open the
media channel of SIP When the media channel of trunk side is ready”.

7.4 Setting for Calls overtime
Unfold “System” and Click “Function” to set:
No Answer Overtime, limit the maximal waiting time-length of calls. After sent calls, if over
this time, there is still no answer, and then end calls. Range is from 0 to 600, unit is second. 0,
disable. If enable this function, please set it as 90s.
Max Talk Duration, limit the maximal talk time-length/duration. After connected the calls, if
over this time, then end calls. 0, disable. Unit is second.
7.5 SNMP
Unfold “System” and Click “Function” to set:
Select “Enable”, to enable function of SNMP.
Server IP Address, specify the target IP address where Trap message send to.
Server Port, specify the target port where Trap message send to. Usually it is 162.
Send state information of our device by Trap messages. Following state information can be
sent by trap message: PCM Sync State, PRI Link Step, SS7 Link Step, Line Call
Information and PCM Sync State (Detail); select to enable.

7.6 Intelligent Analysis Call Forward
Sometimes, E1 on device connect with PBX, there configure call forward on PBX, but the PBX
can not send out the three number, at this time there need our device to analysis and process.
For example, A call B, B call forward to C, route as follows:
1. Send call "A to B" to E1 side;
2. Receive a call "A to C".

Unfold “System” and Click “Function” to select “Intelligent Analysis Call forward”, then our
device will judge the call as call forward, if router is “Transfer to SIP”, will use field “Diversion”
to take number B.
7.7 Smart Call-back
In a company or an enterprise, sometimes multiple extensions share one external direct line
number only, when extension make call-out, if the called does not answer, later when called
make call-back to the direct line number, user hope this call-back can directly assign to the
own phone, and don‟t transfer it by colleague.
Click “Analog Lines”, and select “Smart Call-back Enable”, to enable this function. When
extension use switch-board number as caller number to call out PSTN, if called user don‟t
answer, after a time, when called user call back to switch-board number, device will
automatically assign this call to the extension.
At present, this function is effective for the route rule “Queuing”.
7.8 SMS Transmit
If there is wireless line on device, then wireless line can transmit SMS with SIP side each
other.
Unfold “VoIP” and click “Server”, select “SIP Message Trans SMS” on right, to enable this
function.

When SIP send text message, device automatically recognize, and use SIM card to transmit
SMS out. When SIM card receive SMS, if there want to transfer message content to SIP side,
there need to configure a special router. Add a router in router group of wireless line, fill SID
with “smsrx”, and select “Transfer to VoIP”.
7.9 Network Limit
Click “Network” to configure on right. Under “Secondary IP Router”, here can limit network
function of device, include telnet, SNMP, WEB configuration and Configure tool, only use
which IP to connect. Select “IP1”, there can only connect these network function of device via
the first IP segment; if select “IP2”, there can only connect these network function of device via
the secondary IP route that configured here. Default, there “don‟t limit”.
7.10 Non Tonal Detect
Unfold “System” and click “Function”, then configure under “Non Tonal Detect”. There can
judge whether called answer when call out from FXO or wireless SIM card by properly adjust
parameter.
Chapter 8 System Manage
8.1 Telnet Account

Specify certificated account and password when connected from Telnet, maximal 10 accounts.
Unfold “System Manage” and select “Telnet Account”, to add, modify and delete on right.
8.2 Access Control

To be device‟s safe, here can set IP address white list that used for access device, can set
maximal 128 IP addresses.

Unfold “System Manage” and select “Access Control”, to add, modify and delete on right.
When IP Mask is 255.255.255.255, to limit one single IP.
Once filled with, there are only the IP in list can access device, the IP which are not in list, will
be rejected to connect.
Attention: for MG800, if want to enable this function, there must change the parameter
“flags” into “0x20000” in Boot.
8.3 Change Password

Unfold “System Manage” and select “Change Password”, to change login password for web
interface.
8.4 Update Firmware

Unfold “System Manage” and select “Update Firmware” to update firmware.
Click button “Select file”, and select firmware file, after clicked “OK”, then start to load file; if the
file is big, and network speed is a little slow, it‟s possible to need some time to complete
loading.
After finished loading, there will show the size of file. There are two flash chipset to save
firmware, default there use NOR FLASH, NAND FLASH as backup. Click button “Write” to
write firmware into flash chipset, click button “Cancel” to cancel operation.
After started “Write”, there will show rate of progress as follows:
Speed of “write” into NOR FLASH is a little slow, there need some time, but speed of “write”

into NAND FLASH is fast. After completed “write”, then can see:
After saw “Update OK”, there can reboot device to be effective.
Attention: different device has different firmware. Please select correct firmware file when update.
8.5 Update MSP

Unfold “System Manage” and select “Update MSP” to update MSP. Usually, it‟s not necessary
to update MSP. The process of updating MSP is same as updating firmware, only file is
different.
8.6 Update Voice

Unfold “System Manage” and select “Update Voice” to load file of voice database. The
progress is similar with “update firmware”. Here will not remind the writing destination, after
complete loading, will automatically start to write.
8.7 Debug Trace
8.7.1 Introduction of common commands
All of commands in our device should distinguish capital or small letters.
Command Parameters Description
Look up the basic information of device
ver Query CSP software version
mspver Query MSP software version
pboot Look up basic configuration
shsid Read the serial number
sgmtime Show time
ipconfig Read IP address

Read IP address
ifconfig eth0; For example: “ifconfig Configure IP address, similar with Linux
“192.168.16.253:FFFFFF00”” command
Look up the resource configuration
LineGrpShow Look up call control table
CvTableShow Look up number convert table
LenTableShow Look up number length rule table
RtTableShow Look up the router rules table
PcmAppShow Number of PCM Look up PCM parameters
Ss7AppShow Number of link Look up SS7 link parameters
Operation Command
reset Reset equipment
cboot Modify basic configuration
ldcsp Update CSP firmware
settime Hour, minute, second Set time
Year, month, date, hour, minute,

setdate Set date and time
second
Enable LAN port to send

logNetEn
messages
logNetDis Close LAN port to send a message
logTerEn Enable console port to send a message
logTerDis Close console port to send a message
setlapp Trace levels (0 ~ 7) Trace global call
setltab Trace levels (0 ~ 7) Trace table condition of call control
setlsip Trace levels (0 ~ 7) Trace information of SIP Call
setlreg Trace levels (0 ~ 7) Trace information of SIP registration

Appendix 1 Description of INI File
Work parameter / Device configuration usually save as text file in format of INI. INI is plain text
file, data content are text string, provide by text file or string. Data in a row to state one
segment or item, separate rows by enter or line break.
Format of configuration parameters:
[Segment]
ItemName = ItemValue ; (description)
Segment is paragraph name, ItemName is item name, and ItemValue is parameter value of
the item.
Segment allows using hyphen to indicate one group configuration of different content with
same attribute. Use “-number” to indicate hyphen, for example, [segment-0] indicate the first
paragraph of same attribute, [Segment-1] indicate the second paragraph of same attribute.
Segment indicate the beginning of one paragraph, next paragraph begin; also indicate the last
paragraph ended.
Item Name also allows using hyphen. Difference comparing with Segment is to use
ItemName[number] to indicate. For example, ItemName[0] indicate the first item of same
attribute, ItemName[1] indicate the second item of same attribute.
Item Value must be in specified item of specified segment. One parameter in number value
allow to use decimal or hexadecimal to indicate, Ox is indication way of hexadecimal.
Semicolon indicate ending of one row or beginning of description, there can add necessary
description content behind semicolon “;”. System will not process the description content.
Appendix 2 Description for Function of Keys on
Phone
Keys Function Description
**00# Get line number

**01# Get line caller ID
**19*Old password *new Subscriber line to change password
password#
**10*0 password# Unlock access (high-limit)
**10*1 password# Lock access (with a low limit)
**11*1# Set temporary Do not disturb (off-lock)
**11*0# Cancellation of the provisional don‟t disturb
**12*1# Set do not disturb
**12*0# Cancel do not disturb
**13*SUB PHONE# Unconditional call forwarding
**13*0# Cancel unconditional call forwarding
**14*SUB PHONE# Busy transfer
**14*0# Cancel busy transfer
**15*SUB PHONE# No answer transfer
**15*0# Cancel No answer transfer
**20*01# Objects to join the meeting
**20*01# Object to attend the meetings
**96# Device version ID
**97# Device serial number
**98* new IP address# Set device address, the format is
XXX*XXX*XXX*XXX
**99# Attach device IP address
Operation Ways to answer, transfer and conference by PBX
Extension pick up and press “*8” to answer any calls. (Operation codes can be specified by
router configuration)
Extension pick up and press “*extension number”, to answer specified calls. (Operation codes
can be specified by router configuration)
Suppose A is any call, B, C and D are extensions. When A call PBX, B answer, B can transfer
it or hold a conference by patting “flash”. Operations are as follows:

B pat “flash” (gently pat, if pat heavily, will disconnect the call), at this time, A keep status of
calling, and hear ring back tone; B press extension number of C, if C is free, C will ring, then B
hear ring back tone, A still hear ring back tone; if C is not free, B will hear busy tone, at this
time B only pat “flash” again and can talk with A again.
After B pat “flash”, if C answered, at this time B and C talk, A still hear ring back tone; if B hang
up, then A and C will take. If B pat “flash” again, then A, B and C will talk in three-ways and
enter into conference. After entered into conference, B or C can pat “flash” and press
extension number of D, and invite D into conference, and so on, invite all extensions into
conference one by one.
Attention: conference operation need to authorize by hardware.
Appendix 3 Make and Load Ring Tone
There can use the recording software (such as Windows system with a tape recorder) to
record tone. Saved as wav sound files, format must be used CCITT A-Law, 8kHz, 8 bit, mono,
7kb / sec. After recording finished, use NiceVocMake.exe to convert format.
Run NiceVocMake.exe, interface is as follows.

Click " " button, will add the recorded wav file, select the added item, click can
play the tone, and click can re-select the file, and , button can
move the position of this tone up or down, Then click , to save these tone file
into the BIN file.
Unfold “System Manage” and click “Update Voice”, load the made voice BIN file into device,
after loaded, please reboot device to be effective.
Default, there are 31 voices, content are as follows:
Index Chinese English

0 欢迎词(如：你好，XX 公司) thanks for your call
请直拨分机号,查号请拨零 please dial extension number, press 0 to
1
（操作提示语） manual service
2 用户忙 the number you are dialing is busy now
the number you are dialing does not answer
3 无人应答
the phone
4 无此号码,请改拨其他号码 the number you are dialing is not exist
5 请稍后再拨,或改拨其他号码 please dial later, or dial other number
6 音乐(用于回铃音) music
7 音乐(用于等待音) music
8 零 zero
9 一 one
10 二 tow
11 三 three
12 四 four
13 五 five
14 六 six
15 七 seven
16 八 eight
17 九 nine
18 十 ten
19 年 year
20 月 month

21 日 date
22 点 Hour
23 分 minute
24 号为您服务 Answer for you
25 操作成功 operation successful
26 操作无效 operation failed
27 嘀（声音） di
28 请在提示音后留言 please leave your message after the tone
29 您拨打的用户已欠费停机 sorry, the number you dialed is out of service
30 正在为您转接，请稍后 Now transfer your call, please waiting
Appendix 4 Information Name for SS7
Signaling
ACB Access barred signal
ACC Automatic congestion control information message
ACM Address complete message (note)
Remark: there contains six signaling in ACM
ADC Address-complete, charge
ADN Address-complete, no charge
ADX Address-complete, coin box
AFC Address-complete, charge subscriber free
AFN Address-complete, no charge, subscriber free
AFX Address-complete, coin box, subscriber free
ADI Address incomplete signal
ANC Answer signal, charge
ANN Answer signal, no charge
ANU Answer signal, unqualified
BLA Blocking-acknowledgment signal
BLO Blocking signal

BSM Backward set-up message
CBK Clear-back signal
CCF Continuity-failure signal
CCL Calling party clear signal
CCM Circuit supervision message
CCR Continuity-check-request signal
CFL Call-failure signal
CGC Circuit-group-congestion signal
CHG Charging message
CLF Clear-forward signal
CNM Circuit network management message group
COT Continuity signal
CSM Call supervision message
DPN Digital path not provided signal
EUM Extended unsuccessful backward set-up information message
FAM Forward address message
FOT Forward-transfer signal
FSM Forward set-up message
GRA Circuit group reset-acknowledgment message
GRM Circuit group supervision messages
GRQ General request massage
GRS Circuit group reset message
GSM General forward set-up information message
HBA Hardware failure oriented group blocking-acknowledgment message
HGB Hardware failure oriented group blocking message
HGU Hardware failure oriented group unblocking message
HUA Hardware failure oriented group unblocking-acknowledgment message
IAI Initial address message with additional information
IAM Initial address message
LOS Line-out-of-service signal
MAL Malicious call identification signal
MBA Maintenance oriented group blocking-acknowledgment message
MGB Maintenance oriented group unblocking message

MGU Maintenance oriented group unblocking message
MPM Meter Pulse Message
MUA Maintenance oriented group unblocking-acknowledgment message
NAM National area message
NCB National call supervision message
NNC National-network-congestion signal
NSB National successful backward set-up message
NUB National unsuccessful backward set-up massage
OPR Operator signal
RAN Re-answer signal
RLG Release-guard signal
RSC Reset-circuit signal
SAM Subsequent address message
SAO Subsequent address message with one signal
SBA Software generated group blocking-acknowledgment message
SBM Successful backward set-up information message
SEC Switching-equipment-congestion signal
SGB Software generated group blocking message
SGU Software generated group unblocking message
SLB Subscriber Local busy signal
SSB Subscriber-busy signal (electrical)
SST Send-special-information tone signal
STB Subscriber toll busy signal
SUA Software generated group unblocking-acknowledgment message
UBA Unblocking-acknowledgment message
UBM Unsuccessful backward set-up information message
UNN Unallocated-number signal

Appendix 5 SS7 link
1. One SS7 Signaling Link to Manage Multiple Devices

(or cards)
Take two devices as an example. (Devices=cards) Suppose that there are four E1 port with
one SS7 link from the other side, use two sets of MG900 to connect. (or two cards), each
MG900 is with 2 E1 ports.
MG900-1 with IP address: 192.168.16.253;
MG900-2 with IP address: 192.168.16.254;
Diagram of connection as follows:
Signal Point
1-2-4
CIC:0 CIC:1 CIC:2 CIC:3

SS7Link
PCM0 PCM1 PCM0 PCM1

Signal Point Signal Point
1-2-3 MG900-1 1-2-3 MG900-2
192.168.5.253 192.168.5.254
Ethernet
Configuration of SS7 link:
SS7 Link Configuration MG900-1 MG900-2
Device‟s IP 192.168.5.253 192.168.5.254
LinkType 3_NetServer 2_NetClient
LinkTimeSlot PCM0 16TimeSlot
SLC 0 0
Functional ISUP ISUP
OPC 1-2-3 1-2-3

DPC 1-2-4 1-2-4
SGW IP 0.0.0.0 192.168.5.253
SubMail 0 0
SS7Link from network by UDP √ √
PCM Configuration MG900-1(192.168.5.253)
PCM0 SS7 LinkID:0，CIC:0，Mask 0 TimeSlot
MG900-2(192.168.5.254)
MG900-1:
There is ss7 signaling link on the E1 port of MG900-1, set its Link Type as “Net server”;
(please pay attention to number of “Submail”, the port of the network connection is decided by
the number of “Submail”. If the number of “Submail” is 0 and Link ID is o, then the port of the
network connection is 7320. )
For network connection, there are two kinds: TCP and UDP. Please select UDP.

Link ID of two E1 ports on MG900-1 are all 0, CIC of the first E1 port is 0, CIC of the second E1
port is 1.
MG900-2:
There is no signaling link on the E1 port of MG900-2, there is voice channel only, it take link
from the MG900-1 via network, so the way of connection (Link Type) is “NetClient”. There is no
need to fill with Link Timeslot, keep it empty. SGW IP is the IP of MG900-1. Select “UDP”. Link
ID of two E1 ports on MG900-2 are all 0, CIC of the first E1 port is 2, CIC of the second E1 port
is 3.
The configuration sample of the above two MG900:
If there are 33 E1 ports, there should take 9 sets of MG900 with 4E1 ports.(Each MG900 is
with 4E1 ports)
The configuration of signaling link on E1 is same as the above example MG900-1. CIC of 4E1
ports are 0, 1, 2, 3.
The configuration of the rest MG900‟s is same as the above example MG900-2, only
difference is CIC, CIC is in ascending order.

2. Two SS7 signaling links to manage three devices (or

three E1/DSP cards)
For example, there are 5 E1 ports from operator (/the other side) with two SS7 signaling links.
Double links share loading. Use three E1/DSP cards to connect, such as card-1, card-2,
card-3. Connection diagram is as follows:
SP
1-2-4
CIC: 0 CIC:1
16 timeslots CIC:2 CIC:3 CIC:4
16 timeslots
with links with links
PCM0 PCM1 PCM0 PCM1 PCM0 PCM1

SP SP SP
1-2-3 1-2-3 1-2-3
MG900-1 MG900-2 MG900-3
192.168.5.251 192.168.5.252 192.168.5.253
Ethernet
To make double backup, please connect two SS7 links with 2pcs of MG900s/cards, the third
MG900/card should connect the sever side of the 2pcs of MG900s/cards at same time.
Configurations of 3pcso of MG90s/cards are as follows:
SS7 Link Configuration MG900-1/card-1 MG900-2/card-2 MG900-3/card-3
Device‟s IP 192.168.5.251 192.168.5.252 192.168.5.253
Link Type 3_NetServer 3_NetServer 2_NetClient
Link Timeslot PCM0 16TimeSlot PCM0 16TimeSlot
SLC 0 1
Functional ISUP ISUP ISUP
OPC 1-2-3 1-2-3 1-2-3
DPC 1-2-4 1-2-4 1-2-4
Sub Mail 1 2 1
Signaling gateway IP 192.168.5.252 192.168.5.251 192.168.5.251

Sub Mail of link parameter 2 2 1 2
Signaling Gateway IP2 0.0.0.0 0.0.0.0 192.168.5.252
SS7Link from network by UDP √ √ √
PCM Configuration MG900-1(192.168.5.251)
MG900-2(192.168.5.252)
MG900-3(192.168.5.253)
Connection way of MG900-1/card-1 is Net_sever, because it not only needs to provide link
service, but also needs to get link from MG900-2/card-2. Connection way of MG900-2/card-2
is same as MG900-1/card-1. MG900-3/card-3 as Net_client connects with MG900-1/card-1
and MG900-2/card-2 at same time, so that can make backup better.
Configuration files:
lsd_MG_1_251.ini lsd_MG_2_252.ini lsd_MG_3_253.ini
Check link status:
When configuration is correct and link is normal, use command “Ss7ShowStatus” to check link
status.
MG900-1（192.168.5.251）
-> Ss7ShowStatus
---------LinkID:0---------------
MailID:1,MailNum:1,ready:1
Mail[0]:8,[1]:0 (show link status is ok)
CtiMode:0,Monitor:0
Clt[0] peer=192.168.5.252:7322
Clt[1] peer=192.168.5.253:1024
cic map:
cic:1 = Clt[0]

cic:3 = Clt[0]
cic:4 = Clt[1]
(Above show CIC 1 and 3 is controlled by MG900-2/card-2, CIC 4 is connected into
MG900-3/card-3)
MailOk:1,netOk:1,listen:7321(listen port is 7321)
GwMail[0]:8,[1]:0,GwMailID:2
Clt conn:1,dst=192.168.5.252:7322(as Net_client that connected into M900-2/card-2)
MG900-2（192.168.5.252）
-> Ss7ShowStatus
---------LinkID:0---------------
Mail[0]:8,[1]:0(show link status is ok)
CtiMode:0,Monitor:0
Clt[0] peer=192.168.5.251:7321
Clt[1] peer=192.168.5.253:1025
cic map:
cic:0 = Clt[0]
cic:2 = Clt[0]
cic:4 = Clt[1]
(Above show CIC 0 and 2 is controlled by MG900-1/card-1, CIC 4 is connected into
MG900-3/card-3)
MailOk:1,netOk:1,listen:7322(listen port is 7322)
GwMail[0]:8,[1]:0,GwMailID:1
Clt conn:1,dst=192.168.5.251:7321(as Net_client that connected into M900-1/card-1)
MG900-3（192.168.5.253）
-> Ss7ShowStatus
---------LinkID:0---------------
Mail[0]:8,[1]:8(show link status is ok)
CtiMode:0,Monitor:0
MailOk:2,NetOk:2
conn:1,dst=192.168.5.251:7321,mail:1(as Net_client that connected into M900-1/card-1)
conn:1,dst=192.168.5.252:7322,mail:2(as Net_client that connected into M900-2/card-2)

Appendix 6 FAQ
1. Network Failure
Firstly, please check whether physical connection of network cable is normal, judge it via the
LED light of LAN port, if there is no light on the LED light of LAN, there is a problem in network
cable; if the LED light of LAN is green, so there is no problem in network cable.
If under the case that the network can not be adaptive, connection of point-point use cross
network cable, connection of HUB use straight network cable.
Under the case that the connection of network cable is no problem, please connect to console
port and check by command. Use command “ifconfig” to check whether configuration of IP is
correct or not; if not correct, please change IP.
Attention please: there is only one time that NC-MG930 can adapt the switch of network speed
10/100/1000M, if during the operation, there changed connection of network cable, it‟s better
to reboot device, so that NC-MG930 can re-adapt the network speed.
2. E1 Failure
Firstly, please check the status of PCM LED light on device. If it is red and twinkle, means
there is a problem of physical E1 connection, please check E1 PCM cable and transmission
cable; if it is red and light, means that the physical E1 connection is good, but there is a
problem of link, if it is off, means there is no problem of physical E1 connection and link.
Under case of remote connection, please use PcmMonitor to check. If there is a problem of
physical E1 connection, please check E1 PCM cable and transmission cable; there can use
way of loop to check line. Near side, remote side, loop test one by one. (Loop means the
connection of TX and RX in one same E1.
3. PRI Failure
Usually, Failure of PRI is related with configuration of two parameters: Net Mode (User side or
Net side) and enabling or disabling of CRC4. If you don‟t know how to configure them, please

try to set them one by one.
4. SS7 Failure
If SS7 cannot be up, there is only need to check the configuration of E1 time-slot that loaded
SS7 link. Firstly, please try a loop test on our device, if after the loop test, the link on our device
is good, secondly please ask the opposite side to check the configuration of link time-slot.
After link is up, if there is flash off or there can not normally call, perhaps there is a problem of
configuration of Signaling Point Code or Signaling Link Code. There can use mtp2tl 0, 0, 5 to
trace.
After link is up, if there is no voice of calls, perhaps there is a problem of configuration of CIC,
please check.
5. Failed to connect PcmMonitor
Look up the monitor port in configuration is correct or not, and check the firewall of computer.
6. No voice of SIP calls
Check whether the codec is same as the opposite side. Also check network environment,
whether there is IP conflict, or hardware firewall.
Appendix 7 Line Order of cables
1. Console cable
End of RJ45 crystal head, access device, and the other side of RS232 DB9 Female head, then
link to computer. The relationship is from console RJ45 to RS232.

2. Network cable
There are two kinds of network cable, one is Crossover cable, the other is straight-through. If
device connect to computer, please use Crossover cable. If connect to HUB, use
Straight-through cable.
Network cable at both end of RJ45 line sequence as shown:
3. E1 cable
E1 cable enclosed with device is from RJ45 to 2 BNC. PCM connector form RJ45 to BNC:
RJ45 1 2 4 5
Surface Core Surface Core

BNC
RX TX

Picture of E1 cable: (RJ45 to 2 BNC);
If you want to make an E1 cable from RJ45 to RJ45, please check the following line order of
RJ45 E1 cable for your reference.
4. Analog FXS/Extension cable
1. Line order of analog FXS cable from MG930‟s RJ45 backplane FXS is as follows: (. Each
RJ45 port from MG930 outputs 4 lines)

2. DB36 Backplane (Each DB36 port from MG930 outputs 16 lines)
Line order of DB36 cable:
Chromatograph of DB36 Analog Cable
DB36 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Pin 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34
Line
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Number
Primary
White White White White Red Red Red Red Black Black Black Black Yellow Yellow Yellow Yellow
Color
Salve
Blue Orange Green Brown Blue Orange Green Brown Blue Orange Green Brown Blue Orange Green Brown
Color
5. E&M cable
Signaling interface of E&M can be divided as: PBX side and TLE (Tie Line Equipment) side;
we define our board E&M as TLE side.
Our E&M supports three Types: Type I, Type II, and Type V. When the E&M signaling are I and
II, the connection device on two sides must be PBX and TLE. When he E&M signaling are V,
there is no request for the connection device on two sides, they can be same or not same; to
unify, when the E&M signaling are these three types, the connection device on two sides are
PBX and TLE.

Type of our E&M (Type I, or Type II, or Type V), is decided by the other end device. The
connected device must work under one same Type. If work mode of the other end device is
PBX, then work mode of our E&M is TLE; if work mode of the other end device is TLE, then
work mode of our E&M is PBX.
Work mode PBX or TLE, this is configured by coding switch on E&M board, the detailed
configuration are as follows:
Type Work Mode PBX TLE
Type I Coding order 4 and 5 Coding order 3 and 6
switch to ON switch to ON
Type II Coding order 2, 5 and 7 Coding order 1, 6 and 8
Type V Coding order 3, 5 and 7 Coding order 3, 5 and 7
Way to make 4-wires audio E&M cable
Firstly, find the definition of PIN of the other end device; then check the following picture A or
picture B, and make 4-wires audio E&M cable according to the following steps:
1. Connect SB of our E&M and the other end device (if no, don‟t connect).
2. Connect SG (GND) of our E&M and the other end device (if no, don‟t connect).
3. Connect “Send signaling M” of our E&M and “Receive signaling E” of the other end device.
4. Connect “Receive signaling E” of our E&M and “Send signaling M” of the other end device.
5. Connect “(Audio send Tip&Ring) T&R” of our E&M and “(Audio Receive Tip&Ring) T1&R1”
of the other end device (T--T1, R--R1)
6. Connect “(Audio Receive Tip&Ring) T1&R1” of our E&M and “(Audio send Tip&Ring) T&R”
of the other end device (T--T1, R--R1)
Picture A: Line order of 4-wires Audio E&M cable on MG930

RJ45 PIN Order DB36 PIN Definition of Description
(N=0,1,2,3) PIN
1 22+4N SB Signal of power

2 4+4N E Receive signaling
3 21+4N R1 Audio receive Ring
4 3+4N R Audio send Ring
5 20+4N T Audio send Tip
6 2+4N T1 Audio receive Tip
7 19+4N M Send signaling
8 1+4N SG Signal of ground connection
Picture B: Line order of 4-wires Audio E&M cable on MG320

RJ45 PIN Order Definition of PIN Description
8 SB Signal of power
7 E Receive signaling
6 R1 Audio receive Ring
5 R Audio send Ring
4 T Audio send Tip
3 T1 Audio receive Tip
2 M Send signaling
1 SG Signal of ground connection
Way to make 2-wires audio E&M cable
Firstly, find the definition of PIN of the other end device; Then check the following picture C or
picture D, and make 2-wires audio E&M cable according to the following steps:
1. Connect SB of our E&M and the other end device (if no, don‟t connect).
2. Connect SG (GND) of our E&M and the other end device (if no, don‟t connect).
3. Connect “Send signaling M” of our E&M and “Receive signaling E” of the other end device.
4. Connect “Receive signaling E” of our E&M and “Send signaling M” of the other end device.
5. Connect “(Audio send Tip&Ring) T&R” of our E&M and “(Audio Receive Tip&Ring) T&R” of
the other end device (T--T, R--R)

Picture C: Line order of 2-wires Audio E&M cable on MG930

RJ45 PIN Order DB36 PIN Definition of Description
(N=0,1,2,3) PIN
1 22+4N SB Signal of power
2 4+4N E Receive signaling
3 21+4N
4 3+4N R Audio Ring
5 20+4N T Audio Tip
6 2+4N
7 19+4N M Send signaling
8 1+4N SG Signal of ground connection
Picture D: Line order of 2-wires Audio E&M cable on MG320

RJ45 PIN Order Definition of Description
PIN
8 SB Signal of power
7 E Receive signaling
6
5 R Audio Ring
4 T Audio Tip
3
2 M Send signaling
1 SG Signal of ground connection

6. PTT cable
Definition of RJ45 connector as follows:
Line order of PTT board for MG930:

DB15 RJ45
PIN7 《————————》 PIN1 (PTTIN)
PIN13《————————》 PIN2 (GND)
PIN3 《————————》 PIN3 (Vout)
PIN5 《————————》 PIN4 (Vin)
PIN6 《————————》 PIN6 (PTTOUT)
PIN7 (Vout, does not through resistance to output
directly)
PIN Function Description for Function

Name
PIN 1 PTTIN External PTT (PTT board give intercom device high electrical
potential) 0V become 5V
PIN 2 GND Ground Connection
PIN 3 Vout Receive audio to output
PIN 4 Vin External MIC signals input.
PIN 6 PTTOUT or Carrier Detection (Intercom device give PTT board high
COR electrical potential) 0V become 5V
PIN7 Vout Receive audio to output (This PIN directly output and does not
through resistance.
Any Digital Mobile Radio connects with PTT from our company, there are 5 pins from
expandable port of Digital Mobile Radio, which define as following functions:
Function Description for Function
Vin External MIC signals input.
Vout Receiving audio output
PTTOUT or COR Carrier Detection (Intercom device give Niceuc PTT board high electrical potential)
0V become 5V
PTTIN External PTT (Niceuc PTT board give intercom device high electrical potential) 0V

become 5V
GND Ground Connection
(Remark: Intercom device=digital mobile Radio)
Examples:
Motorola GM338 Motorola M8220 MG930-PTT

PIN Function PIN Function Funtion PIN
2 VIN 11 VIN Vout PIN3
11 VOUT 14 VOUT Vin PIN4
8 PTTOUT 21 PTTOUT PTTIn PIN1
3 PTTIN 17 PTTIN PTTOut PIN6
7 GND 8 GND GND PIN2
Other digital mobile radio Connection MG930-PTT
PIN Function/definition PIN Function/definition PIN order of RJ45
connector
VIN ----------------- Vout PIN3
VOUT ---------------- Vin PIN4
PTTOUT ---------------- PTTIn PIN1
PTTIN ---------------- PTTOut PIN6
GND ---------------- GND PIN2
A.7.1 An example for Connection of PTT Card
The Vehicle Radio on other side is MD780 from Hytera.
A.7.1.1 Physical Connection
Make connection between PTT card and digital mobile radio or vehicle radio. Use cable to
connect with mainframe of digital mobile radio or vehicle radio, not hand microphone. (You
could find connectors on back of mainframe)

A.7.1.2 Line Order of Connection Cable
Brand Hytera MD780 Color of the Niceuc MG930 PTT

following example
Connector DB26 RJ45
PIN Actual name Function Function PIN

of Function when
connects with
Niceuc PTT
7 Tx Audio (as VIN) Green full line VOUT 3
8 Rx Audio (as VOUT) Green dotted line VIN 4
3 DB26_GPIO4 (as PTTOUT) Blue PTTIN 1
Full line
16 DB26_PTT_IN (as PTTIN) Orange PTTOUT 6
Full line
2 Ground (as GND) Blue dotted line GND 2

RJ45
DB26
1 2 3 4 5 6 7 8

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Manual of WEB Configuration

from NICEUC Company

Shenzhen NICEUC communication Tech Co., Ltd

Address: Room 1101-1102, Yizhe Building, Yuquan Rd, Nanshan District,

Shenzhen City, Guangdong, China (Postcode: 518052)

sales@niceuc.com supports@niceuc.com -2-

14) Conference ................................................................................................... - 43 -

sales@niceuc.com supports@niceuc.com -3-

6.6.3 Out Gain and In Gain of Wireless line ...................................................... - 105 -

sales@niceuc.com supports@niceuc.com -4-

6. No voice of SIP calls .................................................................................................... - 137 -

sales@niceuc.com supports@niceuc.com -5-

Chapter 1 Introduction of devices

number: NC-MG900 and NC-MG1000.

number: NC-MG160, NC-MG320, MG08W-64SIM, NC-MG320W, NC-MG640, NC-MG640W,

·IPPBX: IP Integrated Gateway, support E1/T1, FXO/FXS, E&M, GSM/CDMA/WCDMA and

Recording etc. Product model number: NC-MG320, NC-MG640, NC-MG930.

NC-AD300R, NC-AD300D, etc.

Chapter 2 Introduction of WEB Configuration

it‟s very easy and convenient.

“192.168.16.254”; the default port is 80 for web services.

Mozilla Firefox, etc. To be better in showing, please use Google Chrome.

sales@niceuc.com supports@niceuc.com -6-

2.1 Connect devices by Web browser

changed, default password is “nice” which can be changed by subscriber.

2.2 Introduction for interface of web configuration

whole frame of the interface are as follow:

1. Logo: company logo of NICEUC; or we can make customer‟s logo here.

there show English only (change by web resource database)

4. Operations for devices, some related operations.

6. Contents: for each sub key of each function.

sales@niceuc.com supports@niceuc.com -7-

Chapter 3 Operations for device

3.1 Look for configuration

load into RAM.

3.2 Save configuration

3.3 Set to Default

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3.4 Save to file

saving or opening the file, select “save” to save it.

Attention: before saving, please “Read ROM” or “Read RAM” in advance.

3.5 Load from file

successfully, there will show “loading from file is successful”.

Chapter 4 Look for status

4.1 Device Details

hardware, software, etc. For example:

App version 8.14,Made on Feb 13 2014, 17:58:32 Version of software and

sales@niceuc.com supports@niceuc.com -9-

soft info:931 Software type

sid:0x0,123456 Serial number of device

dev type:30 Hardware type

dev id:0x00 Hardware ID

dev version:0x33 Hardware version

AUTH Authorization information

OK,E1=4,USER=208,GSM=64,VOIP=128,REG=1024,SAPI=0,OID=42419. for the quantity of lines

SNMP for functions

Firmware version: v10_25_03_02 Version of MSP software

SPU version: C64V_7_9_2 SPU Version

Hardware type of MSP

MSP type(21): M82354,channels=64 and the quantity of

4.2 Hardware Resources

there will show the details of E1/T1 port.

lines, each type is with different color.

FXS(internal line/extension) in red, High-Resistance Recording in green, PTT(Push To