UC120 IP PBX User Manual

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Preface

UC120-1V1S1O Universal Gateway


User Manual V1.0

Shenzhen Dinstar Co., Ltd.


User Manual V2.1
Address: Floor 18, Building 7A, Vanke Cloud City Phase 1, Xingke 1st Street, Xili Sub-district, Nanshan District,
Shenzhen.

Postal Code: 518052

Telephone: +86 755 61919966

Fax: +86 755 2645 6659

Emails: [email protected], [email protected]

Website: www.dinstar.com

UC120-1V1S1O User Manual Copyright©2011-2019 Dinstar I


Preface

Preface

Welcome
Thanks for choosing the UC120-1V1S1O Universal Gateway! We hope you will make full
use of this rich-feature gateway. Contact us if you need any technical support:
86-755-26456110/112.

About This Manual


This manual provides information about the introduction of the gateway, and about how to
install, configure or use the gateway. Please read this document carefully before install the
gateway.

Intended Audience
This manual is aimed primarily at the following people:

⚫ Users
⚫ Engineers who install, configure and maintain the gateway.

Revision Record
Document Name Document Version Firmware Version
UC120-1V1S1O Universal Gateway V1.0 (2019/11/25) 2.55.1.4
User Manual V1.0
Conventions
Gateway or device mentioned in this document refers to the UC120-1V1S1O gateway. Those
words in blue are the contents that users need to pay attention to.

UC120-1V1S1O User Manual Copyright©2011-2019 Dinstar II


Contents

Contents

1 Product Introduction ...................................................................................................................... 1


1.1 Overview ......................................................................................................................................................... 1
1.2 Application Scenario ....................................................................................................................................... 1
1.3 Product Appearance.......................................................................................................................................... 2
1.4 Description of Indicators ................................................................................................................................ 3
1.5 Features & Functions ...................................................................................................................................... 4
1.5.1 Key Features ........................................................................................................................................... 4
1.5.2 Physical Interfaces ................................................................................................................................. 4
1.5.3 Voice Capabilities ................................................................................................................................... 5
1.5.4 FXS .......................................................................................................................................................... 5
1.5.5 FXO.......................................................................................................................................................... 5
1.5.6 Software Features.................................................................................................................................... 6
1.5.7 Supplementary Services .......................................................................................................................... 6
1.5.8 Environmental ......................................................................................................................................... 6
1.5.9 Maintenance ............................................................................................................................................ 7
2 Quick Installation............................................................................................................................ 8
2.1 Installation Attentions ...................................................................................................................................... 8
2.2 Installation Steps .............................................................................................................................................. 8
2.3 Network Connection ........................................................................................................................................ 9
2.3.1 Network Connection Diagram under Route Mode.................................................................................. 9
2.3.2 Network Connection Diagram under Bridge Mode ................................................................................ 9
2.4 Connect Gateway to Network ........................................................................................................................ 10
2.4.1 Connect Gateway to Network via Network Port ................................................................................... 10
2.4.2 Connect Gateway to Network via WiFi ................................................................................................ 10
2.4.3 Preparations for Login .......................................................................................................................... 11
2.4.4 Log In Web Interfacer ........................................................................................................................... 11
3 Basic Operation ............................................................................................................................. 12
3.1 Methods to Number Dialing ........................................................................................................................... 12
3.2 Call Holding ................................................................................................................................................... 12
3.3 Call Waiting.................................................................................................................................................... 12
3.4 Call Transfer ................................................................................................................................................... 12

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Contents

3.4.1 Blind Transfer ....................................................................................................................................... 12


3.4.2 Attended Transfer .................................................................................................................................. 13
3.5 Three-way Conference ................................................................................................................................... 14
3.6 Switching Bwtween Two Calls ...................................................................................................................... 14
3.7 Description of Feature Code .......................................................................................................................... 15
3.8 Send or Receive Fax....................................................................................................................................... 16
3.8.1 Fax Mode Supported ............................................................................................................................. 16
3.8.2 Explanation of T.38 and Pass-through .................................................................................................. 16
3.9 Function of RST Button ................................................................................................................................. 17
3.10 Query IP Address and Restore Default Setting............................................................................................. 18
4 Configuration Wizard ................................................................................................................... 19
4.1 Configuration Wizard ..................................................................................................................................... 19
4.1.1 UC120 Regarded as Terminal and Registered to SIP Server................................................................. 19
4.1.2 Other SIP Clients registered to UC120 ................................................................................................. 20
4.1.3 UC120 Connected to PBX through Trunking ....................................................................................... 20
4.1.4 UC120 Serving as VPN Client .............................................................................................................. 20
5 Configurations on Web Interface ................................................................................................ 21
5.1 Introduction to Web Interface ......................................................................................................................... 21
5.2 Status .............................................................................................................................................................. 22
5.2.1 Overview ............................................................................................................................................... 22
5.2.2 SIP......................................................................................................................................................... 23
5.2.3 PSTN..................................................................................................................................................... 23
5.2.4 DHCP Client List .................................................................................................................................. 25
5.2.5 Fail2ban ................................................................................................................................................ 25
5.2.6 VPN ...................................................................................................................................................... 25
5.2.7 Current Call........................................................................................................................................... 26
5.2.8 Parking Lot............................................................................................................................................ 26
5.2.9 CDRs ..................................................................................................................................................... 26
5.2.10 Service ................................................................................................................................................ 27
5.2.11 Performance ........................................................................................................................................ 27
5.2.12 About................................................................................................................................................... 28
5.3 System ............................................................................................................................................................ 29
5.3.1 Setting ................................................................................................................................................... 29
5.3.2 User Manager ........................................................................................................................................ 30
5.3.3 Provision ............................................................................................................................................... 32
5.3.4 Operation Log ....................................................................................................................................... 33
5.3.5 Service Log ........................................................................................................................................... 34

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5.3.6 Config Changes Log ............................................................................................................................. 34


5.3.7 Backup/Restore/Upgrade ...................................................................................................................... 34
5.3.8 Voice ..................................................................................................................................................... 36
5.3.9 Command Line...................................................................................................................................... 36
5.3.10 Diagnostics.......................................................................................................................................... 36
5.3.11 Cloud Service ...................................................................................................................................... 37
5.3.12 API ...................................................................................................................................................... 41
5.3.13 Event Report ....................................................................................................................................... 41
5.3.14 Schedul Task ....................................................................................................................................... 42
5.3.15 Email ................................................................................................................................................... 43
5.3.16 FTP Server .......................................................................................................................................... 44
5.3.17 Disk Manager ...................................................................................................................................... 44
5.3.18 Reboot ................................................................................................................................................. 45
5.4 Network .......................................................................................................................................................... 45
5.4.1 Setting ................................................................................................................................................... 45
5.4.2 VoLTE ................................................................................................................................................... 48
5.4.3 Uplink Config ....................................................................................................................................... 49
5.4.4 Access Control ...................................................................................................................................... 51
5.4.5 Firewall ................................................................................................................................................. 52
5.4.6 DHCP Server......................................................................................................................................... 53
5.4.7 Port Mapping ........................................................................................................................................ 54
5.4.8 DMZ Setting ......................................................................................................................................... 56
5.4.9 Diagnostics............................................................................................................................................ 56
5.4.10 VPN..................................................................................................................................................... 57
5.4.11 Static Route ......................................................................................................................................... 62
5.4.12 Hosts ................................................................................................................................................... 63
5.4.13 Fail2ban .............................................................................................................................................. 63
5.5 Profile ............................................................................................................................................................. 65
5.5.1 SIP......................................................................................................................................................... 65
5.5.2 FXS/FXO .............................................................................................................................................. 68
5.5.3 Codec .................................................................................................................................................... 74
5.5.4 Number ................................................................................................................................................. 75
5.5.5 SIM Number Learning .......................................................................................................................... 77
5.5.6 Time ...................................................................................................................................................... 78
5.5.7 Manipulation ......................................................................................................................................... 79
5.5.8 Speed Dial ............................................................................................................................................. 80
5.5.9 Dialplan ................................................................................................................................................. 81
5.5.10 AutoCLIP ............................................................................................................................................ 83

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Contents

5.5.11 Recording ............................................................................................................................................ 84


5.5.12 Voice Mail ........................................................................................................................................... 86
5.6 Extension ........................................................................................................................................................ 87
5.6.1 SIP......................................................................................................................................................... 87
5.6.2 FXS ....................................................................................................................................................... 89
5.6.3 Ring Group............................................................................................................................................ 92
5.6.4 Paging Group ........................................................................................................................................ 93
5.6.5 Call Queue ............................................................................................................................................ 94
5.7 Trunk .............................................................................................................................................................. 97
5.7.1 SIP......................................................................................................................................................... 97
5.7.2 FXO ...................................................................................................................................................... 99
5.7.3 VoLTE Trunk ....................................................................................................................................... 102
5.8 Call Control .................................................................................................................................................. 105
5.8.1 Setting ................................................................................................................................................. 106
5.8.2 Route Group ........................................................................................................................................ 107
5.8.3 Route ................................................................................................................................................... 108
5.8.4 Feature Code ....................................................................................................................................... 110
5.8.5 IVR...................................................................................................................................................... 111
5.8.6 SMS Route .......................................................................................................................................... 112
5.8.7 SMS .................................................................................................................................................... 113
5.8.8 USSD .................................................................................................................................................. 114
5.8.9 Diagnostics.......................................................................................................................................... 115
6 Glossary........................................................................................................................................ 117

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1 Product Introduction

1 Product Introduction
1.1 Overview
The UC120-1V1S1O gateway is a multi-functional and all-in-one gateway, which integrates
voice service (VoLTE, VoIP and PSTN) and data service (LTE 4G/WCDMA 3G). It provides
three interfaces (including LTE, FXS and FXO), offering seamless connectivity to VoIP
Network, PLMN and PSTN.

Based on SIP, UC120-1V1S1O not only can interact with IPPBX, softswitch and SIP-based
network platforms, but also supports types of WCDMA/LTE frequency ranges, thus meeting
the worldwide requirements about the mobile network.

Besides, the gateway has built-in WiFi and high-speed data handling capacity, allowing users
to enjoy high-speed internet surfing through WiFi or LAN ports.

UC120-1V1S1O is ideally suitable for personal use. Meanwhile, it is perfect for small and
micro enterprises, offering high-speed internet access, good voice service and message
service.

1.2 Application Scenario


The application scenario of UC120-1V1S1O universal gateway is shown as follows:

Figure 1-1 Application Scenario of UC120-1V1S1O

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1 Product Introduction

1.3 Product Appearance


Front View:

Back View:

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1 Product Introduction

1.4 Description of Indicators


Indicator Definition Status Description
There is no power supply or power
Off
PWR Power Indicator supply is abnormal.
On The UC120 device is powered on.
Slow The device is initialized successfully
Flashing and is running normally
RUN Running Indicator On The device is being initialized.

Off The device is not running normally.


Fast
WiFi is in normal running.
Flashing
WiFi WiFi Indicator Off WiFi is disabled or WiFi is faulty

On The WiFi module malfunctions.


Slow The FXS port is initialized
Flashing successfully and is in idle status
FXS FXS In-use Indicator The FXS port is in off-hook (in-use)
On
status.
Off The FXS port is faulty
Fast The FXO port is connected with
Flashing PSTN line and is in idle status
The FXO port has yet to be
Slow
connected with PSTN line, but is in
FXO FXO In-use Indicator Flashing
normal status.
The FXO port is currently occupied
On
by a call.
Off The FXO port is faulty.
Network does not work or network
Off cable is not connected to the
Network Connection
WAN/LAN WAN/LAN port..
Indicator
Fast
Network is successfully connected.
Flashing
(1) LTE indicator
(2) Strong signaling indicator
(3) Weak signaling indicator

When these three indicators are all dull (off), it


means that the VoLTE module does not exist.

When the LTE indicator fast flashes and at the


SIM LTE 4G Indicator
meantime, both the strong signaling indicator and
the weak signaling indicator are dull, it means the
VoLTE module malfunctions.

When the LTE indicator is on, and at the


meantime, both the strong signaling indicator and
the weak signaling indicator are dull, it means that
the VoLTE module works normally but no SIM is

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1 Product Introduction

inserted.

When the LTE indicator is on, and at the


meantime, the strong signaling indicator is dull
and the weak signaling indicator fast flashes, it
means that the VoLTE module works normally
but the SIM card is faulty.

When the LTE indicator is on, and at the


meantime, the strong signaling indicator is dull
and the weak signaling indicator slow flashes, it
means that both the VoLTE module and the SIM
card work normally and the SIM card is being
registered/dialed.

When the LTE indicator is on, and at the


meantime, the strong signaling indicator is dull
and the weak signaling indicator is on, it means
that both the VoLTE module and the SIM card
work normally and the SIM card can be
registered/dialed properly.

When these three indicators are all on, it means


that both the VoLTE module and the SIM card
work normally and the SIM card can be
registered/dialed properly.

1.5 Features & Functions


1.5.1 Key Features
⚫ FXS/FXO/LTE interface on a single gateway
⚫ Send/receive calls from LTE and from PSTN/PLMN via FXO
⚫ Flexible dial plan and routing strategies based on time, number and source IP etc.
⚫ IVR Customization
⚫ Support high-speed NAT forwarding and WiFi hotspot
⚫ Serve as VPN client
⚫ Support voice mail and call recording
⚫ Built-in SIP server, support up to 60SIP extensions and 15 concurrent calls
⚫ User-friendly web interface, multiple management ways

1.5.2 Physical Interfaces


⚫ FXS Port: 1
⚫ FXO Port: 1
⚫ USB port: 1

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1 Product Introduction

⚫ SIM Slot: 1
⚫ SD Slot: 1
⚫ Network Port:1 WAN Port & 1 LAN Ports(10/100 Base-T RJ45)

⚫ WiFi:2.4Ghz 802.11n

1.5.3 Voice Capabilities


⚫ VoIP Protocols: SIP over UDP/TCP/TLS, SDP, RTP/SRTP
⚫ Codecs: G.711a/μ law, G.723.1, G.729A/B, G722
⚫ Silence Suppression
⚫ Comfort Noise Generator(CNG)
⚫ Voice Activity Detection(VAD)
⚫ Echo Cancellation: G.168 with up to 128ms
⚫ Dynamic Jitter Buffer
⚫ Adjustable Gain Control
⚫ Automatic Gain Control (AGC)
⚫ Call Progress Tones: Dial Tone, Ring Back Tone, Busy Tone
⚫ FAX: T.38 and Pass-through
⚫ NAT Traversal: STUN/UPnP
⚫ DTMF: RFC2833/Signal/Inband

1.5.4 FXS
⚫ FXS Connector: RJ11
⚫ Caller ID: Bellcore Type 1&2, ETSI, BT, NTT and DTMF
⚫ Answer and Disconnect Signaling: Answer, Disconnect, Busy Tone
⚫ Polarity Reversal
⚫ Hook Flash

1.5.5 FXO
⚫ FXO Connector: RJ11
⚫ Caller ID: FSK and DTMF
⚫ Polarity Reversal
⚫ Answer Delay
⚫ Busy Tone Detection
⚫ No Current Detection

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1 Product Introduction

1.5.6 Software Features


⚫ Ring Group
⚫ Routing Groups
⚫ Caller/Called Number Manipulation
⚫ Routing Based on Time Period
⚫ Routing Based on Caller/Called Number Prefix
⚫ Routing Based on Source Trunks
⚫ Dial Rules
⚫ Failover Routing
⚫ FXO Impedance Auto Match
⚫ IVR Customization
⚫ Auto Attendant Function
⚫ CDRs

1.5.7 Supplementary Services


⚫ Call Forwarding (Unconditional/Busy/No Reply)
⚫ Call Waiting and Call Holding
⚫ Call Transfer (Blind & Attended)
⚫ Call Queuing
⚫ Intra-group Pick-up
⚫ Auto-answer
⚫ Hotline
⚫ No Disturbing
⚫ Voice Mail
⚫ Three-way Conversation

1.5.8 Environmental
⚫ Power Supply: 12VDC, 2A
⚫ Power Consumption: 18W
⚫ Operating Temperature: 0 ℃ ~ 45 ℃
Storage Temperature: -20 ℃~80 ℃
⚫ Humidity: 10%-90% (Non-Condensing)
⚫ Dimensions:260×180×35mm(W/D/H)
⚫ Weight:1.0kg

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1 Product Introduction

1.5.9 Maintenance
⚫ Web GUI for Configuration
⚫ Telnet Management
⚫ Configuration Restore & Backup
⚫ Multiple Languages
⚫ Firmware Upgrade: support HTTP/HTTPS/TFTP/FTP
⚫ Auto Provision
⚫ CDR Query and Export
⚫ Syslog Query and Export
⚫ Network Tools: Ping, Traceroute and Nslookup
⚫ Flow Statistics: TCP, UDP, RTP
⚫ Network Capture

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2 Quick Installation

2 Quick Installation

2.1 Installation Attentions


To avoid unexpected accident or device damage, please read the following instructions before
you install the UC120-1V1S1O gateway.

⚫ The adapter of the gateway accepts DC input voltage of 12V 2A. Please ensure stable
and safe power supply;
⚫ To reduce the interference to telephone calls, please separate power cables from
telephone lines;
⚫ To guarantee stable running of the gateway, please make sure that there is enough network
bandwidth;
⚫ For better heat dissipation, please place the gateway on a flat surface and do not pile up
with other devices;
⚫ If WiFi is enabled, please ensure the WiFi antennas are well connected with the gateway
⚫ If you want the gateway to communicate with the LTE network, please insert an SIM card.

2.2 Installation Steps


⚫ Connect WiFi antenna to the WiFi antenna interface of the gateway, and then connect the
LTE antennas to WiFi antenna interfaces.

⚫ Connect the power adapter to the power jack;


⚫ Connect telephone line to the FXS port and connect PSTN line to the FXO port;

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2 Quick Installation

⚫ Connect network cable to the LAN port(s) and WAN port (please refer to 2.3 Network
Connection);
⚫ Insert a SIM card to the SIM slot, with the chip facing down.

2.3 Network Connection


UC120-1V1S1O works in two network modes: route mode and bridge mode. When it is under
the route mode, the IP address of WAN port must be different from the IP address of LAN
port. But when it is under the bridge mode, the IP address of WAN port and that of LAN port
are the same.

2.3.1 Network Connection Diagram under Route Mode


Under the route mode, the default IP address of WAN port is a DHCP IP address, while the
default IP address of the LAN port is a static IP address, namely 192.168.11.1.

Figure 2-1 Network Connection Diagram under Route Mode

Note: The IP address of LAN port of the gateway and the IP address of PC must be at the
same network segment, while that of WAN port is at a different network segment.

2.3.2 Network Connection Diagram under Bridge Mode


Under the Bridge mode, the IP address of WAN port is the same with that of LAN port.
Generally, when the gateway works under the bridge mode, the IP address of the gateway has
been modified. In the following diagram, it is assumed that the IP address has been modified
into 172.16.80.1.

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2 Quick Installation

Figure 2-2 Network Connection Diagram under Bridge Mode

Note: The IP address of PC and that of WAN port of the UC120-1V1S1O gateway are at the
same network segment.

2.4 Connect Gateway to Network


The above network diagrams show how to connect the gateway to network through network
ports. In fact, the gateway can also be connected to network through WiFi.

2.4.1 Connect Gateway to Network via Network Port


Please connect the UC120-1V1S1O gateway to network according to the network diagrams in
Section 2.3 Network Connection. Then connect a telephone to the FXS port. Dial *158# to
query the IP address of LAN port. Modify the IP address of PC to make it at the same
network segment of LAN port of the gateway.

You are also allowed to log in the gateway by using the WAN PORT, but you need to

2.4.2 Connect Gateway to Network via WiFi


Connect power source to the gateway, and then use a laptop to search the SSID of the gateway.
The default SSID is domain_the last six characters of the Mac address, for example, if the
mac address is F8-A0-3D-59-09-A3, the default SSID is domain_5909a3.

By default, there is no password for WiFi, and the built-in DHCP server is turned on.

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2 Quick Installation

2.4.3 Preparations for Login


Modify the IP address of the PC to make it at the same network segment with the
UC120-1V1S1O gateway, since the default IP address of LAN port of the gateway is
192.168.11.1.

Check the connectivity between the PC and the UC120-1V1S1O. Click Start → Run of PC
and enter cmd to execute ‘ping 192.168.11.1’ to check whether the IP address of LAN port
runs normally.

2.4.4 Log In Web Interfacer


Open a web browser and enter the IP address of LAN port (the default IP is 192.168.11.1).
Then the login GUI will be displayed.

You also can enter the IP address of WAN port, but it’s required to modify the IP address of
PC to make it at the same network segment with that of WAN port.

It is suggested that you should modify the username and password for security consideration.

Figure 2-3 Login GUI of UC120-1V1S1O

By default, the username is admin, while the password is admin@123#. After entering
username and password, click Login to enter into the web interface.

Under some circumstances, login of the Web will be limited:

⚫ For three consecutive login failures, you need to slide to validate your user account;
⚫ Failing to log in the Web for ten times consecutively, the IP address of the UC120 device
will be put into the blacklist, and you need to reset a new IP address for the device;

⚫ Successful login or device restart will wipe out login failure records.

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3 Basic Operation

3 Basic Operation

3.1 Methods to Number Dialing


There are two methods to dial telephone number or extension number:

⚫ Dial the called number and wait for 4 seconds for dialing timeout, or dial the called
number directly (the system will judge whether the dialing is completed according to
Digitmap and Regular Expression dialplans).

⚫ Dial the called number and press #.

3.2 Call Holding


If a calling party places a call to a called party which is otherwise engaged, and the called
party has the call holding feature enabled, the called party is able to switch to the new
incoming call while keeping the current call holding on by pressing the flash button or the
flash hook.

When the called party presses the flash button or the flash hook once again, he or she will
switch back to the first call.

3.3 Call Waiting


If a calling party places a call to a called party which is otherwise engaged, and the called
party has the call waiting feature enabled, the calling party will hear a IVR voice ‘Please hold
on, the subscriber you dialed is busy’ and the called party will hear three beeps.

By pressing the flash button or the flash hook, the called party is able to switch between the
new incoming call and the current call.

3.4 Call Transfer


3.4.1 Blind Transfer
Blind transfer is a call transfer in which the transferring party connects the call to a third party
without notifying the third party.

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3 Basic Operation

Example: A gives a call to B and B wants to blindly transfer the call to C. Operation
instructions are as follows:

1. A dials the extension number of B;

2. The extension of B rings, and B picks up the phone. Then A and B go into conversation;

3. B presses *1 to trigger blind transfer (at the same time, A can hear the waiting tone). Then
B dials the extension number of C ( end up with # or wait for 4 seconds);

4. The extension of C rings, B hangs up the phone and C picks up the phone. Then C and A
goes into conversation.

Note:

⚫ On the ‘Call Control →Feature Code’ page, feature code service should be ‘On’.
⚫ If B hears continuous busy tones after he dials the extension number of C, it means the call
has timed out.

3.4.2 Attended Transfer


Attended transfer is a call transfer in which the transferring party connects the call to a third
party after he confirms that the third party agrees to answer the call.

Example: A gives a call to B and B wants to attended transfer the call to C. Operation
instructions are as follows:

1. A dials the extension number of B;

2. The extension of B rings, and B picks up the phone. Then A and B go into conversation;

3. B presses *2 to trigger attended transfer (at the same time, A can hear a waiting tone). Then
B dials the extension number of C;

4. Then one of the following situations will happen:

a. If the extension of C cannot be reached because the dialing/call has timed out, C rejects
the call or C is busy, B will automatically switch to the conversation with A.

b. The extension of C rings (at the same time, B can hear a ringback tone). If B hangs up
the phone at this moment, A will continue to hear the waiting tone. Then if A also hangs up
the phone, the extension of C will continue to ring. If C picks up the phone at this moment,
the call will end directly.

c. The extension of C rings and then C picks up the phone. C and B go into conversation,
and A will continue to hear a waiting tone. If it’s B that hangs up the phone at this moment, C
and A go into conversation. If it’s C that hangs up the phone, B and A go into conversation.

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3 Basic Operation

3.5 Three-way Conference


When the FXS port of UC120 is the caller:
Step1. A dials the number of B and B picks up the phone, and then A and B go into
conversation;
Step2. A presses the flash hook, and then dial the number of C after hearing the dialing tone.
Step3. C pick up the phone, and A and C go into conversation and meanwhile the call between
A and B is kept holding.
Step4. Then, if A presses the flash hook and dials 1, the conversation will switch back to A
and B; if A presses the flash hook and dial 2 , the conversation will switch to A and C; if A
presses the flash hook and dial 3, the conversation will switch to A , B and C (three-party
conversation).
When the FXS port of UC120 is the callee:
Step1. B places a call to A, and A picks up the phone after the phone rings. And then C also
gives a call to A (at the same time, A can hear a waiting tone).
Step2. If A presses the flash hook, A and C go into conversation and meanwhile the call
between A and B is kept holding.
After that, if A dials 1, the conversation will switch back to A and B; if A dial 2 , the
conversation will switch to A and C; if A dials 3, the conversation will switch to A , B and C
(three-party conversation).
Step2 (optional). When C is calling A and B hands up the phone during the process, A and C
will automatically go into conversation.

3.6 Switching Bwtween Two Calls


When the FXS port of UC120 is the caller:
Step1. A dials the number of B and B picks up the phone, and then A and B go into
conversation;
Step2. A presses the flash hook, and then dial the number of C after hearing the dialing tone.
Step3. C pick up the phone, and A and C go into conversation and meanwhile the call between
A and B is kept holding.
Step4. If A presses the flash hook again, and the call will be switched back to A and B. If A
presses the flash hook once more, the call will be switched to A and C.

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3 Basic Operation

When the FXS port of UC120 is the callee:


Step1. B places a call to A, and A picks up the phone after the phone rings. And then C also
gives a call to A (at the same time, A can hear a waiting tone).
Step2. If A presses the flash hook, A and C go into conversation and meanwhile the call
between A and B is kept holding.
After that, if A presses the flash hook again, and the call will be switched back to A and B. If
A presses the flash hook once more, the call will be switched to A and C.

3.7 Description of Feature Code


UC120-1V1S1O provides convenient telephone functions. Connect a telephone to the FXS
port and dial a specific feature code, and you can query corresponding information.

Code Corresponding Function

*159 Dial *159 to inquiry WAN IP

*158 Dial *158 to inquiry LAN IP


*114 Dial *114 to inquiry phone number

*157* Dial *157*0 to set route mode; dial *157*1 to set bride mode

Dial *150*1 to set IP address as static IP address; dial *150*2 to set IP


*150*
address as DHCP IP address
*152* Dial *152* to set IPv4 address, for example:
Dial *152*192*168*1*10# to set IPv4 address as 192.168.1.10
*156* Dial *156* to set IPv4 gateway, for example:
Dial *156*192*168*1*1# to set IPv4 gateway as 192.168.1.1

*153* Dial *153* to set IPv4 netmask, for example:


Dial *153*255*255*0*0*# to set IPv4 netmask as 255.255.0.0
*111 Dial *111 to restart the UC120 device

*51 Dial *51 to enable the call waiting service


*50 Dial *50 to disable the call waiting service
*1 Dial *1 to trigger blind transfer, for example:
Dial *18000, and you can blind transfer to the extension number 8000

*2 Dial *2 to trigger attended transfer, for example:


Dial *28000#, and you can attended transfer to the extension number
8000

*72* Enable unconditional call forwarding service. Example: Dial *72*8000,

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and calls will be unconditionally forwarded to extension number 8000

*73 Disable unconditional call forwarding service


*90* Enable the ‘call forwarding on busy’ service. Example: Dial *90*8000,
and calls will be forwarded to extension number 8000 when the called
number is busy

*91 Disable the ‘call forwarding on busy’ service

*92* Enable the ‘call forwarding on no reply’ service. Example: Dial


*92*8000, and calls will be forwarded to extension number 8000 when
there is no reply from the called number
*93 Disable the ‘call forwarding on no reply’ service

*78 Enable the ‘No Disturbing’ service


*79 Disable the ‘No Disturbing’ service

** Pick up the ringing extension which is in the same ringgroup. Example:


Dial**8000, and you can take the incoming call of extension number
8000
*160* Dial *160*1# to allow HTTP WAN access, Dial *160*0# to deny
HTTP WAN access

Note:
A voice prompt indicating successful configuration will be given after each configuration
procedure. Please do not hang up until hearing this voice prompt.

3.8 Send or Receive Fax


3.8.1 Fax Mode Supported
⚫ T.38(IP-based)
⚫ T.30(Pass-Through)

3.8.2 Explanation of T.38 and Pass-through


T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time.
Under the T.38 mode, analog fax signal is converted into digital signal and fax signal tone is
restored according to the signal of peer device. Under the T.38 mode, fax traffic is carried in
T.38 packages.

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Pass-through: Under the pass-through mode, fax signal is not converted and fax traffic is
carried in RTP packets. It uses the G.711 A or G711U codec in order to reduce the damage to
fax signal.

3.9 Function of RST Button


Press the RST button for different time length, and the UC120-1V1S1O device will execute
different function:
1. On the condition that the device is running normally, press the RST button for 3 to 6
seconds, the login password of the device will be restored to the factory default, and the
network mode will become the route mode, with WAN address obtained through DHCP and
LAN IP address defaulted as 192.16.11.1. At the meanwhile, he access ports of Http, Https,
Telnet and SSH are restored to the default settings.

Figure 3-1 Default settings of Http, Https, Telnet and SSH

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2. On the condition that the device is running normally, press the RST button for 6 to 12
seconds, and all configurations are restored to the default settings.

3. On the condition that the device is powered off, press the RST button and the WPS button,
and connect the UC120 gateway with power source. After about 30 seconds, the device will
wipe out all configurations, rebuild a file system and then re-load a firmware version (this
method is used in case of version fault).

3.10 Query IP Address and Restore Default Setting


After connecting a telephone to the FXS port, you can dial *158 to query the IP address of
LAN port and dial *159 to query the IP address of WAN port.

If you want to restore UC120-1V1S1O to default settings, you can press the RST button for 6
to 12 seconds or you can configure it on the Web interface.

On the Web interface, click System → Backup/Restore/Upgrade and then select the parts
(system, network or service) that need to be restored to default settings. Click Reset and then
restart the device, and the selected parts will be restored to default settings.

Figure 3-2 Reset to Defaults

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4 Configuration Wizard

4.1 Configuration Wizard


The following are the common ways to configure the UC120-1V1S1O gateway.

4.1.1 UC120 Regarded as Terminal and Registered to SIP Server


1. UC120-1V1S1O Registered to SIP Server

Add an SIP Trunk to SIP Server

Enable Register for the SIP Trunk

Add Inbound and Outbound Routes

2. FXS/FXO Port Registered to SIP Server

Add an SIP Trunk to SIP Server

Enable Register for FXS/FXO Port and


Designate the Port to the SIP Trunk

Note: Although ‘Register’ has been enabled for FXS/FXO port, calls through FXS/FXO port
will take inbound and outbound routes as first priority. For outgoing calls, if outbound route
cannot be matched, then the registered SIP trunk will be selected. For incoming calls, if
inbound route cannot be matched, then the registered FXS/FXO port will be selected.

Generally, local extension number is taken as first priority for call routing selection, followed
by DID, route and then registered port.

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4.1.2 Other SIP Clients registered to UC120


Under this mode, UC120-1V1S1O is regarded as an SIP Server. Create an extension account
first on the Extension →SIP interface, and configure listening port on the Profile → SIP
interface. Then, configure the IP address, extension account and listening port of
UC120-1V1S1O on SIP client.

4.1.3 UC120 Connected to PBX through Trunking

Add an SIP Trunk to PBX

Add Inbound and Outbound Routes

4.1.4 UC120 Serving as VPN Client

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5 Configurations on Web Interface

5.1 Introduction to Web Interface


Modify the IP address of PC to make it at the same network segment with that of LAN port of
the UC120-1V1S1O gateway (the default IP of LAN port is 192.168.11.1).

Open a web browser on the PC and then enter the IP address of LAN port. Click Login, and
the login GUI is displayed. Both the default username and password are admin.

The displayed login GUI is shown as follows:

Figure 5-1 Introduction to login GUI

Table 5-1 Introduction of Web Interface

Index Item Description

The name of the gateway; it can be edited on the System →


1 UC120
Setting interface

2 Menu Bar The menu bar of UC120-1V1S1O

All changes to the configuration of the gateway need to be


Unsaved
3 saved. Click Apply to enter into the page to save the changes;
Changes
click Revert to return to original configuration.
Detailed
4 The detailed configuration interface or display interface
Interface
Auto Refresh The button can be enabled or disabled. If it is enabled, the
5
Button information on the Status → Overview/SIP/PSTN/Current

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Call interfaces will be refreshed automatically

The role of the current user logging into the Web. And the
6 User Role “exit” sign will pop up when the mouse moves over there. You
can log out of the web from there

5.2 Status
The ‘Status’ meun mainly displays all kinds of status information. It includes the following
sub-menus: Overview, SIP, PSTN, DHCP Client List, VPN, Parking Lot, Current Call, CDRs,
Service, performance and About.

5.2.1 Overview
Log in the Web interface of UC120-1V1S1O, click Status → Overview, and the following
interface will be displayed. On the interface, divice model, firmware version as well as
information about performance, WAN network, VoLTE network, LAN network, WiFi and
DHCP server are shown.

Figure 5-2 Overview

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5.2.2 SIP
Click Status → SIP, and the following interface will be displayed. On the interface,
information of SIP profile, SIP Trunk and SIP extension is shown.

Figure 5-3 Status of SIP Profile, SIP Trunk and SIP Extension

Table 5-2 Explanation of SIP Parameters

Belong To Parameter Explanation


Name The name of the SIP profile
Listening Address The current listening address and port of SIP
Green color means normal running, while red color
Profile means listening address and port of SIP is
State unavailable.

There are two states :Running and Down

If heartbeat is enabled, option message will be sent


Heartbeat
to peer device (the peer device is reachable)
Green color means available, while red color
SIP Trunk means abnormal, unavailable or prohibited.
Status
There are five statuses: Running, Reged/Up,
Noreg/Up, Trying-Down, Fail-Wait
Profile The profile that is used by the SIP trunk
Profile The profile that is used by the SIP extension
SIP Extension SIP extension is registered or not.
Status
There are two statuses: Registered. Unregistered

5.2.3 PSTN
On the Status → PSTN interface, information of FXS and FXO is shown. Green color means
available or registered, while red color means abnormal, unregistered or prohibited.

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Figure 5-4 Status of FXS and FXO

If ‘SIP Register Status’ is ‘Registered’, it means FXS and FXO have been registered to SIP
server on the Trunk → SIP/FXO interface respectively. FXS can also be registered to SIP
server on the Extension → FXS interface.

Table 5-2 Status Explanation of FXS and FXO

Belong To Parameter Explanation


There are two module statuses: Ready and Config
Module Status
Failed
Parameter Status There are two parameter statuses: OK and error
FXS
There are two SIP register statuses: Registered and
SIP Register Status
Unregistered
Hook State There are two hook states: Onhook and Offhook

There are two module statuses: Ready and Config


Module Status
Failed
Parameter Status There are two parameter statuses: OK and error
FXO There are two SIP register statuses: Registered and
SIP Register Status
Unregistered
Hook State There are two hook states: Onhook and Offhook
Line State There are two hook states: Online and Offline
The state of the VoLTE channel. If the VoLTE SIM
Channel State card is successfully registered, it means the channel
state is OK.
Phone Number The number of the VoLTE SIM card
VoLTE SIP Register Status The register status of the VoLTE trunk in use
Carrier The carrier of the VoLTE SIM card
Signal The signal strength of the VoLTE SIM card
Talking State Whether the SIM card is on call or not

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5.2.4 DHCP Client List


UC120-1V1S1O has a built-in DHCP server. When the DHCP server is enabled, it can assign
IP addresses to the clients connected to it.

On the Status →DHCP Client List interface, information of DHCP clients connected to the
UC120-1V1S1O gateway, such as client name, Mac address and IP address, is shown.

Figure 5-5 DHCP Client List

5.2.5 Fail2ban
On the Status →Fail2ban interface, you can see currently-banned IP addresses t and historic
banned IP addresses. You can also unban those IP addressed that have been blocked before.

Fail2ban is a log-parsing application that monitors system logs for symptoms of an automated
attack on your device. When an attempted compromise is located, using the defined
parameters, Fail2ban will add a new rule to block the IP address of the attacker, either for a set
amount of time or permanently. Fail2ban can also alert you through email that an attack is
occurring.

Figure 5-6 Banned IP Addresses

For the explanation of parameters related to fail2ban, please refer to the “Network
->Fail2ban” section.

5.2.6 VPN
On the Status → VPN interface, the online records and historical records of UC120-1V1S1O
as a L2TP client, a PPTP client and an OpenVPN client are displayed.

Meanwhile, the UC120-1V1S1O gateway can also serve as a VPN server, such as L2TP
server, PPTP server and OpenVPN server. Related online records and historical records are
shown on the Status → VPN ->OpenVPN Server or Status -> VPN -> L2TP/PPTP Server
Access List interface.

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Figure 5-7 VPN Connection Records

5.2.7 Current Call


On Status →Current Call interface, the source, destination, calling number, called number,
start time, answer time, state and duration of the current real-time call are shown. If there is
no current call, no information will be shown

Figure 5-8 Current Call Information

5.2.8 Parking Lot


You can use the parking feature to park a call, and then retrieve the call either from your
phone or another phone. After you park a call, the call is placed on hold, you can continue the
conversation after retrieving it.

On the Status -> Parking Lot interface, the numbers that are parked and the parking duration
are shown.

Figure 5-9 Call Parking Status

5.2.9 CDRs
Click Status → CDRs, and you can set query criteria to query the CDRs (Call Detailed
Records) that you want on the displayed interface. Meanwhile, you are allowed to clear CDRs
or export CDRs through clicking the Empty or Export button. The maximum number of
CDRs that can be saved is 5000.

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CDRs cannot be saved on the Status → CDRs interface unless the CDRs function has been
enabled on the System → Setting interface.

Figure 5-10 CDRs

5.2.10 Service
Click Status → Service, and the service status of UC120-1V1S1O is displayed. This function
is enabled by default. The Web, SSH and Telnet service can be disabled and their ports can be
modified on the Network → Access Control interface. If no running status is shown, it
means exception has occurred on UC120.

Besides, if syslog is disabled on the System → Setting interface, the logs cannot be uploaded
to the server, but log service is still running.

Figure 5-11 Service Status

5.2.11 Performance
On the Status ->Performance Interface, you can see the peformance statistics of the system.

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Figure 5-12 System Performance

5.2.12 About
On the Status → About page, the device model, device SN, hardware ID, MAC address,
IMEI, VoLTE module, boot image, root image, WIFI driver , firmware Version of the
UC120-1V1S1O device are displayed.

Figure 5-13 Device and Company Information

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5.3 System
Configurations for hostname, timezone, NTP, login username & password, other user name,
provision, TR069, operation log, service log, upgrade/backup/restore, IVR upload, Command
Line, cloud server and device reboot can be carried out in the System section.

5.3.1 Setting
On the System → Setting interface, you can modify the device name, set a new timezone,
synchronize local time and enable CDRs, Syslog as well as built-in NTP server.

Figure 5-12 Basic Setting

Figure 5-4 Explanation of Basic Setting Parameters

Parameter Explanation
Hostname The name of the gateway. After it is configured, the name will be
displayed on the left of the menu bar.

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Timezone You can choose a time zone you want. The default value is UTC
(Universal Time Coordinated)
Local Time The current time based on current time zone. It is synchronized with
NTP.
CDRs If it is enabled, CDRs will be saved automatically. 5000 CDRs call be
saved at most and they can be queried on the Status → CDRs
interface. If it is disabled, CDRs will not be saved
Service Log Level There are eight levels, including Debug, Info, Notify, Warning, Error,
Critical, Alert and Emergency
Enable Syslog Whether to enable syslog
Time If NTP server is enabled, the UC120-1V1S1O can be synchronized
Synchronization with the world standard time. Meanwhile, you’re able to add or
reduce NTP servers. Please consult local telecom operators or surf the
internet for the address of NTP servers.

Delete a NTP Server

Add a NTP Server

5.3.2 User Manager


Click System → User Manager, and you can modify the username name and password for
logging in the UC120-1V1S1O gateway. Factory defaults for username name and password
are admin and admin@123# respectively, so it is advised to modify them for security
consideration.
The abovementioned username and password are also used to log in Web Interface, Telnet and
SSH.
The super administrator of the device can add different users to the device and assign different
roles for them, like observer, operator and administrator. Different roles can be allocated with
different permissions to the functions.

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Figure 5-14 Modify Username ,Password and Manager Users

Figure 5-15 Add New User

Figure 5-5 Explanation of Provision Parameters

Parameter Explanation
Name The name of the new user. After it is established, the name and the
password will be used to log into the web page of the device.
User Group You can choose a role for the new user, such as administrator,
operator and observer. The default value is administrator.

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New Password Setting the login password for the new user. The password needs to
consist of 8 to 32 characters.
Expiration The expiry date when the user cannot log in the device any more.
Status Choose enable or disable.
Web Access The permissions to view status, system, network, profile, extension,
Permission trunk and call control.

5.3.3 Provision
Provision is used to make UC120-1V1S1O automatically upgrade with the latest firmware
stored on an http server, an ftp server or a tftp server.

As for how to configure UC120-1V1S1O and http/ftp/tftp server for Provision, please make
reference to the instruction guide of Provision.

Select the checkbox on the right of Enable, and you will see the following interface:

Figure 5-16 Provision

Table 5-6 Explanation of Provision Parameters

Parameter Explanation
Periodic Check Whether to enable periodic check. If it is enabled, the gateway will
automatically check whether the firmware version stored on the URL is

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updated.
Check Interval The interval to check whether the firmware version stored on the URL
is updated. If it is 3600s, the gateway will check every 3600s.
URL The URL of the http/ftp/tftp server:
For example:
ftp://172.16.77.200/home
tftp://172.16.77.200/provision.xml
http://test.domain.com/test
Username The login username of the http/ftp/tftp server
Password The login password of the http/ftp/tftp server

Note: Proxy Address, Proxy Username and Proxy Password are optional to be configured.

5.3.4 Operation Log


The logs tracing the operations carried out on the Web can be queried on the System →
Operation Log interface. You are allowed to set query criteria to query the logs that you want
and to export the logs through clicking the Export button at the top-right corner.

Figure 5-17 Operation Logs

Note: Operation logs are generally used to locate faults by device manufacturer.

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5.3.5 Service Log


Service logs (the running logs of UC120-1V1S1O) can be exported on the System → Service
Log interface. Those logs are used for analyzing where a problem has occurred on the
gateway.

Figure 5-18 Service Log

5.3.6 Config Changes Log


On the System →Config Changes Log interface, the configurations changed by
administrator on the Web of the gateway are recorded.

Figure 5-19 Config Changes Log

5.3.7 Backup/Restore/Upgrade
On the System → Backup/Restore/Upgrade interface, you can back up or restore
configuration data, and can upgrade UC120 to a new version. But you need to restart the
device for the change to take effect after executing restore or upgrade.

Figure 5-19 Backup/Resotre/Upgrade

Figure 5-20 Upgrade the Device

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Note: the file you choose to be upgraded on the above interface is a local file, while the
version file upgraded through the Provision function is a file from http/ftp/tftp server.

Figure 5-21 Back up configurations

Figure 5-22 Upgrade VoLTE module

Table 5-7 Explanation of Backup/Resotre/Upgrade Button

You can download the configuration data to be backed up. Select any of the
Download checkboxes on the left of System, Network and Service, and then click
Download
Select any of the checkboxes on the left of System, Network and Service,
Reset and then click Reset, and configurations related to the selected part will be
restored to factory defaults.

Restore Choose a backup file, and then click Restore.

Choose a file to be upgraded (which is provided by Shenzhen Dinstar Co.,


Upgrade Ltd.), and then click Upgrade.

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5.3.8 Voice
On the System → Voice interface, you can upload an IVR file according to your needs. At
present, only wav audio file is allowed. The format of the uploaded wav audio file must be:
monaural, 8000hz, 16bit, and size of no more than 1M.

Figure 5-23 Upload IVR File

5.3.9 Command Line


On the System → Command Line interface, some commonly-used command lines can be
directly selected in the draw-down box, and therefore user has no need to enter command
lines on Telnet. In this way, the efficiency of problem diagnostics is greatly improved.

Commonly-used command lines include fxo config, fxo status, fxs config, fxs status, gsm
status, gsm bcch, gsm oper, sip status, sip profile and so on.

Figure 5-24 Command Line

5.3.10 Diagnostics
Use a telephone line to connect the FXS port and the FXO port, and then insert an SIM card.
On the System → Diagnostics interface, select a module (FXS/FXO or LTE ) that you want
to diagnose. Click Start, and the gateway will begin to diagnose the selected module(s).

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If the progress bar of diagnostics is green, it means the module that is diagnosed works well;
if the progress bar is red, it means the module that is diagnosed is faulty.

Figure 5-25 Diagnostics

5.3.11 Cloud Service


Cloud service is mainly used to centrally manage all kinds of devices. Through cloud service,
you can query the status of a device, upgrade devices at batch, log in or configure a device
remotely. The UC120-1V1S1O gateway provides Cloud service. Enter the IP address, service
port and password of the Cloud server, and then the gateway will connect to the cloud server.

Figure 5-26-1 Cloud Server

NATS Server:

UC120-1V1S1O can work as a NATS client to send messages to a NATS server, and then the
NATS server will open related ports to facilitate the connection with those clients or servers of
users.

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How NATS works:

1) NATS network topology

2) Operating mode

Both NATS client and management server need to connect to the NATS server;

The management server subscribes to from the NATS server, and the NATS client is
responsible for publishing messages. After receiving the subject published by the client, the
NATS server forwards the message to the management server. The flows of subscribing to
subject message: management server -> NATS server. The flows of reporting a message:
NATS client -> NATS server -> management server;

Configuration steps:

1) Configure NATS server on the web interface (UC120-1V1S1O acts as a client):

On the "System -> Cloud Server" interface, enable the NATS server and configure other
parameters, such as IP address, port (when TLS is disabled, the port defaults to 4222, you can
choose not to fill in), user name and password (can be empty), enable heartbeat etc.

Figure 5.26-2 NATS server configuration

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(3) Configure the same NATS server information on the management server;

(4) Enter the command line on the management server and enter the command to subscribe to
the subject; command format: nats_client subscribe xxxxxx (example: nats_client subscribe
*.server.register)

The following are the three subjects that the management server must subscribe to:

Register subject(used to receive the registration information of the NATS client): nats-client
subscribe *.server register

Event subject (used to receive event messages from NATS clients): nats_client subscribe
*.server.event

Heartbeat subject (used to receive heartbeat messages from NATS clients): nats_client
subscribe *.server.heartbeat

After subscribing , you can use the nats-client status command to view the
Note: registration status and the subjects that have been subscribed. The command
format for unsubscribing subjects is: nats-client unsubscribe *.server.register.

(5) Enable NATS report on the "System -> Event Reporting" interface of UC120-1V1S1O.

Figure 5.26-3 Enable NATS report

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To make NATS report work normally, the following three points need to be met:

1) The NATS client (UC120-1V1S1O) has enabled the NATS report;

2) The connection between the NATS client and the NATS server has been established;;

3) The NATS client has been registered to the management server.

You can view the status of the NATS client by entering "nats-client status" on
Note: the "System -> Command Line" interface. If the status displays "NATS Status:
OK", it means that the NATS client and NATS server are successfully
connected.

Figure 5.26-4 Enter the command to view the status of the NATS client

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5.3.12 API
UC120 provides API (Application Programming Interface) to interwork with other devices or
platforms. This function enables you to centrally manage devices through command lines.

5.3.13 Event Report


UC120-1V1S1O allows the following events to be reported through NATS or URL: device
startup, call status, registering or unregistering of SIP extensions, vailability or unavailability
of SIP trunks, off-hook or on-hook of FXS phone, FXO status and update of CDR
information.

For event report through NATS, please refer to the configuration steps of NATS in the Could
Server section.

For event report through URL, please see the following example:

1. Select the event that is to be repported and the way to report the event (URL);

2. Input the URL.

Format: http://ip:port/event?key1=$value1&key2=$value2

http://172.18.111.65:8080/sip?sn=$sn&mac=$username&key=$sip_status

Event refers to startup, callstatus, sip, siptrunk, fxs, fxo, gsm, volte, vpn and cdr, while value
refers to the parameter that needs to be reported. Key can be defined by yourself, but it’s
generally the same with value.

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Figure 5-27 Input URL

3. Use a softphone to register to an extension of UC120-1V1S1O, and then the registering of


unregistering of the softphone will be reported to UC120 thorugh URL.

4. On the System→Event Report→Log interface, you can view the report information.

5.3.14 Schedul Task


On the System → Schedule Task interface, you can set a scheduled time to reboot the UC120
device, record backup, access SIM internet, and back up CDRs, logs or configurations.

Figure 5-28 Configure scheduled task

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5.3.15 Email
On the System → Email interface, you can configure a email client on UC120, which can be
used to send or reveice emails. The email client can also used to test connection. But on top of
that, SMTP, IMAP and POP 3 services need to be enabled for the email client.

When the email client is used with SMS routing, email and SMS are bound, which brings
great convenience, for example, you can receive an email , although someone is sending you
an SMS message. Meanwhile, logs will be generated can be viewed on the System → Email
→ Log interface.

Figure 5-29 Configure Email Client

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Username Enter the address of email client


password The password or authorization code of the email client
Server Address The Address of the SMTP server, supported by the email
client
Protocol Choose IMAP or POP3. When POPS is selected, TLS
port is 995 by default.
Message Query Interval (min) The time interval to check whether there is a new email.
Message Valid Time Range Only those emails received during this time range are
addressed.
Number of Message Per The maximum number of emails that are received at one
Receive time. If the number exceeds, they will be received in
batches.

5.3.16 FTP Server


On the System → FTP Server interface, you can enable the FTP server function of UC120
and configure related parameters such as username, password and access permissions. You
can connect FTP clients to this FTP server and then access those files (like recording files and
system logs) that are open on the UC120 device through the 21 port.

Figure 5-30 Configure FTP Server

5.3.17 Disk Manager


On the System →Disk Manager interface, you can see the memory usage of USB and SD
card. USB memory are divided into three categories, including voicemail(40%), recording

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(50%) and Others(10%). You can also redevide the proportion of each category, disconnect
the USB or execute formatting on this interface.

Figure 5-31 Disk Manager

Note: UC120 only supports USB of FAT and EXT4.

5.3.18 Reboot
On the System → Reboot interface, you can click Perform Reboot to reboot the
UC120-1V1S1O gateway. After the device is rebooted, those configurations that have been
saved will remain unchanged.

Figure 5-32 Reboot Device

5.4 Network
UC120-1V1S1O works in two modes: route mode and bridge mode. When it is under the
route mode, the IP of WAN must be different from the IP of LAN. But when it is under the
bridge mode, the IP of WAN and the IP of LAN are the same.

5.4.1 Setting
On the Network → Setting interface, you can set the IP address of WAN port and LAN port,
and can turn on WiFi.

Under the route mode, the default IP address of WAN port is a DHCP IP address, while the
default IP address of the LAN port is 192.168.11.1.

In fact, there are three kinds of IP addresses for selection for WAN port and LAN port,
including Static IP address, DHCP and PPPOE.

DHCP: Obtain IP address automatically.

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UC120-1V1S1O is regarded as a DHCP client, which sends a broadcast request and looks for
a DHCP server to answer. Then the DHCP server automatically assigns an IP address to the
UC120-1V1S1O from a defined range of numbers.
Figure 5-33 Default IP Address under Route Mode

Figure 5-34 Set WAN IP as DHCP IP

Note: When WAN IP is set as DHCP IP, please ensure that there is DHCP server working
normally in the network.

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Static IP Address:
Static IP address is a semi-permanent IP address and remains associated with a single
computer over an extended period of time. This differs from a dynamic IP address, which is
assigned ad hoc at the start of each session, normally changing from one session to the next.

If you choose static IP address, you need to fill in the following information:
⚫ IP Address: the IP address of the WAN port of the UC120-1V1S1O;
⚫ Netmask: the netmask of the router connected the UC120-1V1S1O;
⚫ Default Gateway: the IP address of the router connected the UC120-1V1S1O;
⚫ Use custom DNS server: the IP address of the DNS server

Figure 5-35 Set WAN IP as Static Address

PPPoE:

PPPoE is an acronym for point-to-point protocol over Ethernet, which relies on two widely
accepted standards: PPP and Ethernet. PPPoE is a specification for connecting the users on an
Ethernet to the Internet through a common broadband medium, such as a single DSL line,
wireless device or cable modem. PPPOE IP address refers to IP address assigned through the
PPPoE mode.

If you choose PPPoE, you need to fill in to fill in the following information:
⚫ Username: the account name of PPPoE
⚫ Password: the password of PPPoE
⚫ Server Name: the name of the server where PPPoE is placed

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Figure 5-36 Set WAN IP as PPPoE IP

5.4.2 VoLTE
On the Network → VoLTE page , you can enable the VoLTE service, and fill in information
of APN,username,password, mode, PIN code, dial number and so on.

Figure 5-32 VoLTE Config

Table 5-10 Explanation of Parameters for VoLTE Config

Current Mode 4G
Status The LTE module is enabled or not.

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There are two statuses: Enabled. Disabled


APN The name of access point, you can select 3GNET. Cmnet or custom.
Username The name of VoLTE module, used to access network
Password The password of VoLTE module, used to access network
Mode The mobile network has 2g & 3g, 4g, or automatic configuration.
PIN Code The SIM card's personal identification number.
Dial Number The LTE network's dialing number.
Service The LTE service type is UMTS.

5.4.3 Uplink Config


On the Network → Uplink Config page, you can configure an uplink strategy based on the
priority between WAN and VoLTE. The "WAN Master, VoLTE Slave" means that the data
from the UC120-1V1S1O will be transmitted through the WAN port, and when the WAN fails,
the data will be transmitted through LTE, while the "VoLTE Master, WAN Slave" is the
opposite.

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Figure 5-37 Uplink Configuration

Table 5-11 Explanation of Parameters for Uplink Config

Uplink Strategy WAN Master,LTE Slave or LTE Master,WAN Slave


A WAN (or LTE) tracking IP address (which can be more than
one), when this address is not available, it is thought that the
Track IP network of WAN (or LTE) is not available, switching to LTE
(or WAN).
If this IP is 0.0.0.0.0, it means the current configured DNS
address and gateway is being tracked by PING .
Ping Count The times of checking IP addresses by PING.

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If the Ping does not respond during the timeout period, it is


thought as a connection failure. For example, the timeout period
Timeout(s)
is set as 2 seconds, which means if there is no response after
the IP is checked by Ping for 2 seconds, it is a connection failure
Interval(s) The interval for the device to check IP address by PING
The number of consecutive failures (by PING), if this value is
Count of Down
reached, it means the network connection fails
The number of consecutive successes (by PING), if this value is
Count of Up
reached, it means the network is available

5.4.4 Access Control


The access ports of Web, Telnet and SSH, as well as relevant on-off controls, can be
configured on the Network → Access Control interface.

Figure 5-38 Access Control

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5.4.5 Firewall
If the UC120-1V1S1O works under the route mode, you can choose to enable the firewall and
set filter rules to accept or reject certain destination IP addresses.

Configuration Procedures:
1. Select On in the drop-down box on the right of Filter Rules Control
2. Select filter action, accept or reject;
3. Click the New button;
4. Fill in information of filter rule;
5. Click the Save button to save the configuration.
Figure 5-39 Firewall

Note:
: Edit information for the corresponding filter rule.

: Delete the corresponding filter rule.


/*: Information of Source or Destination is not completely filled in.

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Figure 5-40 Create Filter Rule

Table 5-12 Explanation of Parameters for Filter Rule

LAN IP The IP address that you want UC120 to accept or reject. It is the IP
address of a host from local-area network; it can also be a string of
IP addresses, for example, 172.16.11.1/15.
LAN Port The port of LAN host which the accepted or rejected IP address
belongs to
LAN MAC The Mac of the LAN host twhich the accepted or rejected IP
address belongs to
WAN IP The IP address that you want UC120 to accept or reject. It is the IP
address of a host from wide-area network; it can also be a string of
IP addresses, for example, 152.16.11.11/19.
WAN Port The port of WAN host which the accepted or rejected IP address
belongs to
Action Choose accept ot reject

5.4.6 DHCP Server


If there is a need, you can choose to enable the built-in DHCP server of UC120-1V1S1O to
assign IP addresses to PC or other clients that are in the same local-area network with UC120.
Under this condition, the UC120-1V1S1O gateway works like a router.

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Figure 5-41 Enable DHCP Server

Table 5-13 Explanation of Parameters for DHCP Server

Start Address The start IP address of the address pool to be assigned


End Address The end IP address of the address pool to be assigned

Lease Time The validity period of the assigned IP address

Gateway The gateway of the IP address pool to to be assigned, it is optional to


fill in
Master DNS The master DNS of the client whose IP address is assigned by the
built-in DHCP server; it is optional to fill in
Slave DNS The slave DNS of the client whose IP address is assigned by the
built-in DHCP server; it is optional to fill in

5.4.7 Port Mapping


When the UC120-1V1S1O works under the route mode, port mapping allows a client in the
wide-area network to visit a client in the local-area network.

Configuration Procedures:

1. Click Network → Port Mapping, and the following interface will be shown.

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Figure 5-42 Port Mapping

2. Click the New button.

3. Fill in information on the following interface.

Figure 5-43 Create New Port Mapping

Table 5-14 Explanation of Parameters for Port Mapping

Name The name of this port mapping


WAN Port The port of the client in the wide-area network, which is to visit
local-area network
Protocol Choose TCP, UDP or TCP/UDP
LAN IP The IP address of the to-be-visited client in local-area network
LAN Port The port of the to-be-visited client in local-area network (this port
cannot conflict with the port of UC120-1G1S1O)
Status Chose enable or disable.

4. Click the Save button to save the above configurations.

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5.4.8 DMZ Setting


When the UC120-1V1S1O gateway works under the route mode and the DMZ service is
enabled, the clients in the wide-area network are allowed to have direct access to the clients in
the DMZ (demilitarized zone).

Figure 5-44 Enable DMZ Service

5.4.9 Diagnostics
On the Network → Diagnostics interface, you can use three network utilities including Ping,
Traceroute and Nslookup to diagnose the network, and can capture data packages of the
available network ports.

Figure 5-45 Network Diagnostics

Ping is used to examine whether a network works normally through sending test packets and
calculating response time.

Instructions for using Ping:

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1. Enter the IP address or domain name of a network, a website or a device in the input box
of Ping, and then click Ping.

2. If related messages are received, it means the network works normally; otherwise, the
network is not connected or is connected faultily.

Traceroute is used to determine a route from one IP address to another.


Instruction for using Traceroute:
1. Enter the IP address or domain name of a destination device in the input box of Traceroute,
and then click Traceroute.
2. View the route information from the returned message.

Nslookup (Name Server Lookup) is a network command-line tool to obtain domain name of
internet or to diagnose the problems of DNS.
Instruction for using Nslookup:
1. Enter a domain name and then click Nslookup.
2. View the DNS information from the returned message.

Network Capture
On the following interface, you can capture data packages of the available network ports. You
can also set source IP, source port, destination IP or destination port to capture the packages
that you want.

There is a "and"/" or "logical type. The "and" relationship can only capture a one-way
message, or "or" relationship to fetch the interaction message between a particular IP.

Note: If there are multiple source or destination IP addresses, please use ‘|’ to separate them,
for example, 172.16.115.12|172.16.115.15.
After package capturing is completed, save the captured packages on a computer and then use
a tool to analyze them.

5.4.10 VPN
VPN(Virtual Private Network)is a network technology that creates a secure remote network
connection over a public network through encrypted tunnel and conversion of data’s
destination address. UC120-1V1S1O can serve as a VPN client to connect with VPN server.

UC120-1V1S1O supports the following VPN protocols:

Layer 2 Tunneling Protocol (L2TP) is a protocol used to package data of PPP link layer and
transmit the data between two sites over the Internet through a tunnel.

Point-To-Point Tunneling Protocol (PPTP) is another tunneling protocol used to connect a


remote client to a private server over the Internet. PPTP is an enhanced security protocol

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which supports VPN. And its security can be enhanced through PAP (Password
Authentication Protocol) and EAP (Extensible Authentication Protocol).

OpenVPN is a kind of VPN based on the application layer of OpenSSL. It allows VPN
clients to use a sahred key, certificates or username/password to authenticate themselves.

UC120-1V1S1O works as a VPN client

Figure 5-46 UC120 Works as PPTP Client

Table 5-17 Explanation of Parameters for PPTP Client

Status Whether to enable the PPTP client function (UC120-1V1S1O works as


PPTP client)
Default Route Whether to enable default route;

If default route is enabled, data are transmitted between PPTP client


and PPTP server through VPN route; if it is not enabled, data are
transmitted between PPTP client and PPTP server through network’s
outbound route.

Data Encryption Whether to encrypt data during data transmission

Server Address The IP address of the PPTP server that assigns account to PPTP client
Username The username of the account assigned by PPTP server to PPTP client
Password The password of the account assigned by PPTP server to PPTP client

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L2TP:UC120-1V1S1O works as a L2TP client and is connected to L2TP server.

Figure 5-47 UC120 Works as L2TP Client

Table 5-18 Explanation of Parameters for L2TP Client

Status Whether to enable the L2TP client function (UC120-1V1S1O works as


L2TP client)
Default Route Whether to enable default route;

If default route is enabled, data are transmitted between L2TP client


and L2TP server through VPN route; if it is not enabled, data are
transmitted between L2TP client and L2TP server through network’s
outbound route.

Data Encryption Whether to encrypt data during data transmission

Server Address The IP address of the L2TP server that assigns account to L2TP client
Username The username of the account assigned by L2TP server to L2TP client
Password The password of the account assigned by L2TP server to L2TP client

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OpenVPN:Allow a single point of VPN to use certificate to authenticate itself.

Figure 5-48 UC120 works as a OpenVPN client

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UC120-1V1S1O works as a VPN Server

Figure 5-49 UC120 works as a OpenVPN Server

Figure 5-50 UC120 works as a L2TP Server

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Figure 5-51 UC120 works as a PPTP Server

5.4.11 Static Route


On the Network → Static Route interface, you can configure static routes for the network.

Figure 5-52 Create New Static Route

Table 5-16 Explanation of Parameters for Static Route

Name The name of the static route

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Target IP The destination IP address of the static route


Netmask The netmask of the static route, default: 255.255.255.0
Gateway The IP address of the outbound gateway of the static route
Interface The outbound interface of the static route, namely WAN port or LAN port
Status The static route is enabled or disabled

5.4.12 Hosts
On the Network → Hosts interface, you can add a host file. After enabling the hosts file, you
can visit the corresponding host by inputting the alias or domain name of the host. The format
of the hosts file is as follows: IP address host alias/domain name.

The hosts file contains the mapping relationship between IP address and host name/alias
/domain name. And the mapping relationship allows quick and convenient access to the host.

Figure 5-53 Enable Hosts File

5.4.13 Fail2ban
Fail2ban is used to scan system logs and update firewall rules to reject the IP addresses that
show malicious signs (for example, too many login failures) for a specified amount of time.

On the Network → Fail2ban interface, you can configure rules for Fail2ban. For UC120,
Fail2 ban is generally targeted SSH and SIP.

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Figure 5-54 Configure Fail2ban Rules

SSH/SIP

Ban Duration (Second) The time period during which the IP addresses that conform to the
banning rule or are in the backlist are prohibited.
Range: 60-315360000 seconds

Max Retry Durations The time period during which the maximum retries have been
(second) executed and then the corresponding IP address will be banned.
For example, if this parameter is set as 60 seconds and the maximum
number of retries is set as 10, an IP address will be banned in case
that it has tried 10 times during 60 seconds.
Range: 5-3600

Max Retry The maximum number of retries during a specific time.


For example, if this parameter is set as 10 and the max retry duration
is set as 60 seconds, an IP address will be banned in case that it has
tried 10 times during 60 seconds.
Range: 5-3600

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White List Those IP addresses that are in the white list will not be banned by
Fail2ban.

Black List Those IP addresses that are in the black list will not be banned by
Fail2ban.

Note: If an IP address does not receive any response after it has sent out SSH/SIP attempts,
and the network is reachable, you can go to the Status → Fail2ban interface to check
whether the IP address is banned or not.

5.5 Profile
The Profile menu includes the following sub-menus: SIP, FXS/FXO, Codec, Number, Time,
Manipulation and Dialplan.

5.5.1 SIP
On the Profile → SIP interface, you can set SIP information such as listening port, which will
be used in extension and trunk. Multiple SIP profiles can be configured for one
UC120-1V1S1O device, so you can choose different SIP profiles according to different needs.

Figure 5-55 Configure SIP Profile

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Table 5-19 Explanation of Parameters for SIP Profile

Name The name of the SIP profile

The local listening interface of this SIP profile. It can be WAN


port, LAN port, Open VPN, L2TP and PPTP.
Local Listening Interface
If the SIP profile is used by a SIP trunk, the interface filled in
here is the listening port for the SIP trunk.

The local listening port of this SIP profile. If the SIP profile is
Local Listening Port used by a SIP trunk, the port filled in here is the listening port
for the SIP trunk.

Starting NAT can speak on different networks, including


NAT
four: uPNP/NAT-PMP、IP Address、Stun、DDNS

If the parameter is set as 50 seconds, it means that the call


Progress Timeour(s)
will be considered as timeout in case that no one answers the

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call during 50 seconds.

DTMF is short for Dual Tone Multi Frequency


DTMF Type There are three DTMF modes, including SIP Info, INBAND,
RFC2833

RFC2833-PT RFC2833 payload coding

By default, this parameter is off.


Process DTMF as
Hold/Unhold When it is set as on, DTMF will be addressed as call
hold/unhold.

PRACK Provisional Response ACKnowledgement

Session Expires: The validity period of a SIP session. When


a SIP session times out, an invite message needs to be sent to
refresh the session, otherwise, the session ends;
Session Timer It is 1800 seconds by default
Min Session Expires: the minimum validity period to respond
to a SIP session.
Session Refresh Method: re-INVITE or UPDATE

From: User Part : to obtain the caller number from the user
part contained in the ‘From’ field.
From: Display Name: to obtain the caller number from the
display name contained in the ‘From’ field.
Caller Number Source
To: User Part: to obtain the caller number from the user part
contained in the ‘To’ field.
Contact: User Part: to obtain the caller number from the user
part contained in the ‘Contact’ field.

From: User Part : to obtain the called number from the user
part contained in the ‘From’ field.
From: Display Name: to obtain the called number from the
display name contained in the ‘From’ field.
Called Number Source
To: User Part: to obtain the called number from the user part
contained in the ‘To’ field.
Contact: User Part: to obtain the called number from the user
part contained in the ‘Contact’ field.

To take the remote device or the local device as priority for


inbound codec negotiation
Inbound Codec Assume local device supports PCMA, PCMU, G.729 and
Negotiation Priority G.723, while the remote device supports G.723 and G.729

If remote device is taken as codec negotiation priority, G.723


will be the codec mode, since the remote device supports

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G.723 and G.729 and G.723 is prior to G.729

Inbound Codec Profile The codec supported by SIP for inbound calls

Outbound Codec Profile The codec supported by SIP for outbound calls

Bypass Media(SIP to
Whether to allow SIP to communicate with the server directly
SIP)

Detect Extension is Whether to detect the SIP extension using this SIP profile is
Online online or not

If this function is enabled, incoming calls from unknown


sources are allowed. Unknown sources are those IP addresses
Allow Unknown Call
that do not fall into the source range configured for SIP trunks
or SIP extensions
The source of inbound calls, which is allowed. It can be an IP
address or a network segment. If it is a network segment, the
format is 172.16.0.0/16 or 172.16.0.0/255.255.0.0, which
Inbound Source Filter means calls from the network segment of 172.16 is allowed to
come in.

0.0.0.0 means calls of any source is allowed to come in

QoS
Whether to enable QoS. QoS is a technology used to solve
network delay or congestion
User Agent Then content of the ‘user agent’ field in SIP packets
Encryption Whether to encrypt this SIP profile
Timer T1 The value of timer T1 in SIP protocol. Default value is 500ms
Timer T2 The value of timer T2 in SIP protocol. Default value is 4000ms
TimerT4 The value of timer T4 in SIP protocol. Default value is 5000ms

Timer T1X64 (ms) The value of timer T1X64 in SIP protocol. Default value is
32000ms

5.5.2 FXS/FXO
On the Profile → FXS/FXO interface, you can configure the driving parameters of FXS port
and FXO port, including tone standard, dial timeout, ring timeout, hook-flash detection,
DTMF parameters, CID-related parameters, impedance, dialplan and so on.

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Figure 5-56 FXS/FXO Profile

Click , and corresponding configuration interface will pop up.

Figure 5-57 Configure FXS Parameters

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Table 5-20 Explanation of FXS Parameters

Name The name of this FXS profile

The national standard of dialing tone, busy tone and ring tone;
Tone Group
default value is China
The timeout value for dialing a digit of a telephone number;
Digit Timeout (s) When the time of dialing a digit exceeds this value, the system will
think the dialing has completed; Default value is 4 seconds
The timeout value for dialing the first telephone number after
Dial Timeout (s)
off-hook; Default value is 10 seconds

The timeout value for the ringing of the analog phones of the FXS
Ring Timeout (s)
port when there are incoming calls

No Answer Timeout The timeout value for ending a call which goes out through the FXS
(s) port, when nobody answers the call.

Whether to enable flash-hook detection; If flash detection is not


Flash Detection enabled, the press on flash-hook will be ignored and won’t be
processed.
Min Time: when flash-hook detection is enabled, if the time of the
press on the flash-hook is less than this minimum time, the press
will be ignored and won’t be processed.
Min Time(ms)/
Max Time(ms)
Max Time: when flash-hook detection is enabled, if the time of the
press on the flash-hook is longer than this maximum time, the
phone will be hanged up.

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The minimum interval between the sending of two DTMF tone


DTMF Send
Interval(ms) DTMF: Dual Tone Multi Frequency
DTMF Gain Signal gain of DTMF
DTMF Duration
The minimum duration of a DTMF tone
(ms)
DTMF Detect
The threshold for the device to detect DTMF
Threshold
The terminator for ending DTMF detection. It means when the
DTMF Teminator terminator is detected, the system will think the dialing is
completed and begin to process call.
Send DTMF
Whether to send DTMF terminator
Terminator
The modes of sending CID to the called phone when there are
CID Send Mode incoming calls, including FSK and DTMF;
FSK: Frequency-shift keying; CID: Caller ID
The message modes to display caller information, including SDMF
Message Mode
and MDMF
The message formats to display caller information, including
Message Format
Display Name and CID, Only display Name, Only display CID
Send CID Before If it is enabled, the CID will be shown before ringing; otherwise,
Ring CID will be displayed after ringing
Send CID After If it is enabled, the CID will be shown after ringing; otherwise, CID
Ring(ms) will be displayed before ringing
Delay Timeout After
The maximum interval between ringing and displaying of CID
Ring (ms)

Impedance The impedance (SLIC) matched with analog phones

REN(Ringer The equivalent number of ringing phones. It is used to determine


Equivalency how many devices can be connected by telephone lines.
Number) Range:1to 4
If polarity reverse is on, call tolls will be calculated based on the
changes in voltage. If polarity reverse is off, you need to set the
Polarity Reverse
time for offhook detect and call tolls will be calculated starting
from the set time.
Send Flash Hook via
SIP INFO
If this parameter is on, signal of flash-hook is sent via SIP INFO

Offhook Current Detect This current threshold is used to determine whether a phone is
Threshold offhook.

Onhook Current Detect This current threshold is used to determine whether a phone is
Threshold onhook.

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The rules for dialing. The UC120-1G1S1O device supports regular


Dialplan
expression. Please make reference to Profile → Dianplan section.

Figure 5-58 Configure FXO Parameters

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Table 5-21 Explanation of FXO Parameters

Name The name of this FXO profile

The national standard of dialing tone, busy tone and ring tone;
Tone Group
default value is China
The timeout value for dialing a digit of a telephone number;
Digit Timeout (s) When the time of dialing a digit exceeds this value, the system
will think the dialing has completed; Default value is 4 seconds
The timeout value for dialing the first telephone number after
Dial Timeout (s)
off-hook; Default value is 10 seconds

The timeout value for the ringing of the analog phones of the FXS
Ring Timeout (s)
port when there are incoming calls

No Answer Timeout The timeout value for ending a call which goes out through the
(s) FXS port, when nobody answers the call.

Whether to enable ‘detect polarity reverse’. If ‘detect polarity


reverse’ is on, call tolls will be calculated based on the changes in
Detect Polarity
voltage. If ‘detect polarity reverse’ is off, you need to set the time
Reverse
for offhook delay and call tolls will be calculated starting from the
set time.
Detect before ring: the CID will be shown before ringing;
otherwise, CID will be displayed after ringing;
Detect Caller ID Detect after ring: the CID will be shown after ringing; otherwise,
CID will be displayed before ringing
Off: the CID will not be shown

DTMF Detect
The timeout value to detect CID (in DTMF format)
Timeout(s)

Dial Delay(ms) The delay time of dialing. Default value is 400ms

The minimum interval between the sending of two DTMF tone


DTMF Send
Interval(ms) DTMF: Dual Tone Multi Frequency

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DTMF Gain Signal gain of DTMF

DTMF Duration (ms) The minimum duration of a DTMF tone


DTMF Detect
The threshold for the device to detect DTMF
Threshold
The terminator for ending DTMF detection. It means when the
DTMF Teminator terminator is detected, the system will think the dialing is
completed and begin to process call.
Send DTMF
Whether to send DTMF terminator
Terminator

Detect Tone counts Set the number of busy notes to check

Detect Tone Delta Set the error size to check the busy tone

Intermittent Ratio The intermittent ratio to detect busy tone

The rules for dialing. The UC120-1G1S1O device supports


Dialplan regular expression. Please make reference to Profile → Dianplan
section.

5.5.3 Codec
UC120-1V1S1O supports four codec modes, including G729, G723, PCMU and PCMA. You
can adjust the priority of these four modes according to you needs.

Figure 5-59 Add or Delect Codec Profile

: Edit codec profile.

: Delete the corresponding codec profile or a codec mode.

: Create a new codec profile.

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5.5.4 Number
On the Profile →Number interface, you can set a prefix for calling numbers or called
numbers. When the prefix of a calling number or a called number matches the set prefix, the
call will be passed to choose a route.

Figure 5-60 Number Profile

: Edit number profile.

: Delete the corresponding number profile

Click , and you will see the following interface:

Figure 5-61 Create Number Profile

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Table 5-22 Explanation of Number Parameters

Name The name of the number profile


Prefix of Caller The prefix of the calling number. It supports multiple prefixes,
Number multiple rules for "or" relationships .It supports regular expression
Prefix of Called The prefix of the called number. It supports regular expression.It
Number Supports multiple prefixes, multiple rules for "or" relationships.
The length of the calling number or called number. For example, :
Length 4|6|7 means the calling number or called number must be 4 digits, 6
digits or 7 digits except the prefix

Regex (Regular Expression) Syntax

Table 5-3 Explanation of frequently-used metacharacters in Regex

^ Matches the starting position in a number string. For example, ^134


matches the numbers starting with 134
$ Matches the ending position of a string. For example, 2$ matches the
numbers ending with 2.
| Separates alternate possibilities. For example, 2|3|4 means 2,3or 4.
\ Marks the next character as a special character, a literal, a backreference,
or an octal escape
[] Matches a single character that is contained within the bracket. For
example, [123] matches 1, 2, or 3. [0-9] matches any digit from "0" to
"9".
[^ ] Matches any one character except those enclosed in [ ]. For example, [^9]
matches any character except 9.
. Matches any single character except the newline character. For example,
3.4 matches 314, 324, 334, 344.

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? Indicates there is zero or one of the preceding element. For


example, colou?r matches both color and colour
* Indicates there is zero or more of the preceding element. For
example, ab*c matches ac, abc, abbc, abbbc, and so on.
+ Indicates there is one or more of the preceding element. For
example, ab+c matches abc, abbc, abbbc, and so on, but not ac
\d Mark any digit, equal to [0-9 ]
\D Mark any character that is not a digit, equal to [^0-9 ]
\s Mark any blank character such as a space or a tab.
\S Mark any character that is not a blank character

Examples:
^0755 Matches the phone numbers with starting digits of 0755.

^0755|^8899|^0110 Matches the phone numbers with starting digits of 0755, 8899
or 0110.

^[1][358][0-9]{9}$ Matches the phone numbers with the first digit as 1, the second
digit as 3, 5 or 8, the left nine digits as any of 0 to 9.

Note: the matching of number prefix also supports some digits that are not conform to the
format of regular expression. For example, 0755 matches the numbers starting with 0755, and
0755|8899|0110 matches the numbers starting with 0755, 8899 or 0110.

5.5.5 SIM Number Learning


On the Profile → SIM Number interface, you can configure a number learning rule, which is
used to send a text message (SMS ) to a destination number and then the destination number
will reply a message to the UC120 device. UC120 will match the key words of the reply
message to detect the number of the SIM card.

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Figure 5-62 SIM number learning rule

Index The index of the SIM number learning rule, used to identify the rule
Name The name of the SIM number learning rule, used to identify the rule
Type UC120 only supports SMS currently.
The destination number that receives the text message. You can
Destination Number
also enter a phone number to have a test.
The text content that is sent to the destination number. It is based on
Send Text
the region where the SIM card belongs to.
Check SMS From The UC120 device will not match the key words of the reply
Number message unless this number is matched.
Keywords The keywords used to detect the SIM number.

5.5.6 Time
On the Profile → Time interface, you can set a time period for calls to choose routes. If the
local time when a call is initiated falls into the set time period, the call will be passed to
choose the corresponding route.

Click the New button, and you will see the following interface:

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Figure 5-63 CreateTime Profile

Table 5-23 Explanation of Time Parameters

Name The name of the number profile


Configure the starting date and ending date of a period
Date Period : Add a date period
: Delete a date period
Weekdate Choose a weekdate

Time Period Choose the starting time and ending time of a day

5.5.7 Manipulation
Number manipulation refers to the change of a called number or a caller number during
calling process when the called number or the caller number matches the preset rules.

Click the New button, and you will see the following interface:

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Figure 5-64 Create Manipulation Profile

Table 5-24 Explanation of Manipulation Parameters

Name The name of this manipulation profile


The number of digits that are deleted from the left of the caller
Delete Prefix Count
number or calling number
The number of digits that are deleted from the right of the caller
Delete Suffix Count
number or calling number
Add Prefix The prefix added to the caller number or the calling number

Add Suffix The suffix added to the caller number or the calling number
The number which replace the caller number or the calling
Replace by
number
If the checkbox on the right of Caller is selected, it means the
caller number will be manipulated; if the checkbox on the right
of Called is selected, it means the called number will be
manipulated.

Note: During number manipulation, deletion rules are carried out first, followed by adding
rules. If ‘Replace by’ has been set, deletion rules and adding rules are invalid.

5.5.8 Speed Dial


On the Profile → Speed Dial interface, you can set one-digit or two-digit peed dial numbers
for FXS/SIP calls. For example, if the short number (speed dial number ) is set as 1, the long

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number is set as 8000, and this speed dial profile is applied to an FXS/SIP extension, the
FXS/SIP extension only needs to dial 1 and the call will be directed to the extension number
of 8000.

Figure 5-65 Add Speed Dial Profile

5.5.9 Dialplan
Dialplan is used for number dialing of calls through FXS and FXO ports. It supports Regular
Expression (Regex) and DigitMap.

Figure 5-66 Add Dialplan

Regex (Regular Expression) Syntax

^ Matches the starting position in a number string. For example, ^134


matches the numbers starting with 134
$ Matches the ending position of a string. For example, 2$ matches the

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numbers ending with 2.


| Separates alternate possibilities. For example, 2|3|4 means 2,3or 4.
\ Marks the next character as a special character, a literal, a backreference,
or an octal escape
[] Matches a single character that is contained within the bracket. For
example, [123] matches 1, 2, or 3. [0-9] matches any digit from "0" to
"9".
[^ ] Matches any one character except those enclosed in [ ]. For example, [^9]
matches any character except 9.
. Matches any single character except the newline character. For example,
3.4 matches 314, 324, 334, 344.
? Indicates there is zero or one of the preceding element. For
example, colou?r matches both color and colour
* Indicates there is zero or more of the preceding element. For
example, ab*c matches ac, abc, abbc, abbbc, and so on.
+ Indicates there is one or more of the preceding element. For
example, ab+c matches abc, abbc, abbbc, and so on, but not ac
/d Mark any digit, equal to [0-9 ]
/D Mark any character that is not a digit, equal to [^0-9 ]
/s Mark any blank character such as a space or a tab.
/S Mark any character that is not a blank character

Examples of Regex Syntax:


^0755 Matches the phone numbers with starting digits of 0755.

^0755|^8899|^0110 Matches the phone numbers with starting digits of 0755, 8899
or 0110.

^[1][358][0-9]{9}$ Matches the phone numbers with the first digit as 1, the second
digit as 3, 5 or 8, the left nine digits as any of 0 to 9.

DigitMap Syntax:

Digit 0-9
Supported T Timer
Objects A digit, a timer, or one of the symbols of
DTMF
A, B, C, D, #, or *
One or more DTMF symbols enclosed in the [], but
Range []
only one DTMF symbol can be selected
One or more expressions enclosed the
Range ()
(), but only one can be selected

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Separator | Separate expressions or DTMF symbols.


Two digits separated by hyphen (-) which matches a
Subrange -
ny digit between and including the two digits.
Wildcard x Matches any digit of 0 to 9
Modifiers . Matches 0 or more times of the preceding element

Modifiers ? Matches 0 or 1 times of the preceding element

Examples of DigitMap Syntax

Matches the phone numbers with stating digits as


(13 | 15 | 18)xxxxxxxxx
13, 15 or 18 and the left nine digits as any of 0 to 9
Matches the phone numbers starting with any digit
of 2 to 8 and the left six digits as any of 0 to 9; or
[2-8] xxxxxx | 13xxxxxxxxx
matches the phone numbers starting with 13 and the
left nine digits as any of 0 to 9

5.5.10 AutoCLIP
AutoCLIP is mainly used to SIP trunks, FXO trunks ans VoLTE trunks and it helps record the
outgoing and incoming calls of a trunk.

Figure 5-67 Configure AutoCLIP Rule

Index The index of AutoCLIP profile

Name The name of AutoCLIP profile

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You can choose missed calls or all calls. If missed calls is selected, uc120
Record Strategy will record the missed calls of the trunk. If all calls are selected, all the
calls going through the trunk will be recorded
The validity period of a record. For example, if this parameter is set as 2
hours, the record will be valid in 2 hours since the record is generated.
Record Expire (hour) During the validity period, if there is coming call for the extension number
contained in the record, the call will directly led to the extension without
routing.
By default, this parameter is disabled.
Delete Used Record If this parameter is selected, those records that have been used to match
extension number or trunk will be deleted.
By default, this parameter is enabled.
Match Outgoing Trunk If this parameter is enabled, those calls going through the trunks in the
record can coming in without routing.

5.5.11 Recording
On the Profile→ Recording interface, you can choose SD card or Udisk as master/slave
storage location.

How to Record Calls:

Configure a recording profile (or choose one of the two default recording profiles), and then
add it to a SIP/FXS route. When there are calls going through the route and match the
recording profile, the calls will be recorded.

Figure 5-68 Add recording profile

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Figure 5-69 View Recording List

The index of the recording profile


Index
Range: 1-32

Name The name of the recording profile, used to identify the recording profile

Auto Recording after Answer: start recording after the callee pick up the
phone.
Ban Recording: ether caller or callee enables his function, and then the call
Strategy
in both directions will not be recorded.
Manual Recording after Answer: press *3 to start recording after the callee
answers the call.
Inbound & Outbound: If this recording profile is added to FXS/SIP
extension, both inbound and outbound calls will be recorded.
Inbound: If this recording profile is added to FXS/SIP extension, only
inbound calls will be recorded.
Recording Direction
Outbound: If this recording profile is added to FXS/SIP extension, only
outbound calls will be recorded.
Note: If this recording profile is added to routing, this parameter is invalid
and all calls going through the routing will be recorded.
If the actual recording time is shorter than this value, the recording file will
Min Duration
not be saved.

Silence Detect Select on or off.

If the time of initial silence is shorter than this timeout value and there is
Initial Silence Timeout(s) voice afterwards, the recording will not stop.
If the time of initial silence is longer than this timeout value, and there is

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voice afterwards, the recording will stop when the recording time reaches
the preset value.
If the time of final silence is shorter than this timeout value and there is
voice afterwards, the recording will not stop.
If the time of final silence is longer than this timeout value, and there is
Final Silence Timeout(s)
voice afterwards, the recording will stop before the call ends.
Note: The uc120 will not execute final silence detection unless the initial
silence is shorter than its timeout value.

Silence Detect Threshold The threshold for silence detection.

You can click Recording List to view the recording files which show the caller/called number,
recording duration and so on. You can also play, download or delete the recording files on this
interface.

5.5.12 Voice Mail


On the Profile→ Voice Mail interface, you can configure the location, number and duration
of a voice mail.

How to use voice mail:

Go to the Extension → SIP or Extension → FXS interface, enable the voice mail function,
and then calls that times out will enter into voice mail.

Figure 5-70 Configure Voicemail Profile

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Master/Slave Storage
Select SD card or Udisk
Location
If this maximum number of messages is reached, a prompt voice “the mail
Max Message Per User
box is full” will be played.
Maximum of Login If this maximum number of attempts (by dialing *170*2 to log in the
Attempts voice mailbox) is reached, the call will hang up.
When a call enters into the voice mailbox and the caller dial inexistent
Maximum of Operation
DTMF repeatedly, the caller will be forced to log out the voice mailbox
Failure
after the repetition times exceed this value.
Min Message Time
The minimum duration of a voice mail
(second)
Max Message Time
The maximum duration of a voice mail.
(second)
If this parameter is on, new messages will be played automatically.
Auto Play New Message If it is off, a prompt voice “please dial 1 to listen to new message” will be
given.
If this parameter is on, the caller number will be played together with
Play CID Number
messages.

Play from Latest Message If this parameter is on, the latest messages will be played first.

When to play message date. You can choose ‘Before Playing Message’,
Play Message Date
‘After Playing Message’ and ‘Never’.

You can click Message List to view the voicemail files which show the caller/called number,
message duration and so on. You can also play, download or delete the message files on this
interface.

5.6 Extension
5.6.1 SIP
On the Extension → SIP interface, you can configure the SIP accounts registered in the
UC120-1V1S1O by SIP clients (hereby UC120-1V1S1O is regarded as a SIP server).

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Figure 5-71 Configure SIP Extension

Table 5-25 Explanation of Parameters for SIP Extension

Name The name of this SIP extension

Extension The SIP account of the extension registered in UC120 by a SIP client

Password The password of the SIP account registered in UC120 by a SIP client

Direct Inward Dialing; if the called number is same with DID, the call
DID will be directly forwarded to the extension, rather than choosing a
route. Users can set multiple DID.
If ‘Any’ is chosen, all SIP clients are allowed to register the SIP
Register Source
account of this extension; if ‘Specified’ is chosen, only the SIP client

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with the specified IP address or network segment is allowed to register


the SIP account of this extension.
For example, 172.16.0.0/16 means the register source is 172.16
If a calling party places a call to a called party which is
Call Waiting otherwise engaged, and the called party has the call waiting feature
enabled, the calling party will hear an IVR voice.
If ‘Do Not Disturb’ feature is enabled, calls cannot reach the called
Do Not Disturb
party.
If ‘Call Forward Unconditional’ feature is enabled, all coming calls
Call Forward
will be forwarded to a preset number.
Unconditional

Call Forward When the SIP extension is not registered, you can transfer all the calls
Unregister to the set number
If ‘Call Forward Busy’ feature is enabled, new coming call will be
Call Forward Busy
forwarded when the corresponding local port is busy.

Call Forward No If ‘Call Forward No Reply’ feature is enabled, calls will be forwarded
Reply when nobody answer the calls during a specified period.

If NAT is enabled, the IP address of SIP extension in LAN will be


NAT
turned into the outbound IP address of public network, thus making

NAT traversal possible

Call In Filter When you breathe in to SIP, you match the relevant filter conditions

Call Out Filter When the SIP is called out,The filter conditions are matched

Call In Black List The rules in the list will not take effect

Call Out White List The rules in the list take effect

SIP Profile The SIP profile that is selected for the extension

If it is enabled, this SIP extension is registered to UC120-1G1S1O;


Status
Otherwise the SIP extension is not registered

5.6.2 FXS
On the Extension → FXS interface, you can configure the parameters of the FXS extension.

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Figure 5-72 Configure Parameters of FXS Extension

Table 5-26 Explanation of Parameters for FXS Extension

Extension The extension account of FXS port, which is used to register


Direct Inward Dialing; if the called number is same with DID, the call
DID will be directly forwarded to the extension, rather than choosing a
route.

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If it is enabled, the FXS extension account will be registered to the SIP


Register to SIP
trunk that has been set.
Server
Default is off.
The address and port of the master SIP server;
Master Server it is generally the IP address of a SIP trunk. Please make reference to
Trunk → SIP section

Slave Server The address and port of the slave SIP server

Username User name when the FXS port account is registered

The username of this FXS extension account, which is used during


Auth Username register authentication

The password of this FXS extension account, which is used during


Password
register authentication
Specify
Transport
Whether to specify transport protocol on register URL.
Protocol on
Register URL
The validity period after the FXO trunk is registered successfully.
When the time expires, the FXO trunk will send register request to the
Expire Seconds
server.
Default value is 1800s
When the FXO trunk fails to be registered, the interval to send register
Retry Seconds request;
Default value is 60s
Hot line If hotline is enabled, calls will directly go to the hotline number

Number Holine number

Delay The delay time for a call to be send out after dialing is completed
If a calling party places a call to a called party which is
Call Waiting otherwise engaged, and the called party has the call waiting feature
enabled, the calling party will hear an IVR voice.
If ‘Do Not Disturb’ feature is enabled, calls cannot reach the called
Do Not Disturb
party.
If ‘Call Forward Unconditional’ feature is enabled, all coming calls
Call Forward
will be forwarded to a preset number.
Unconditional

Call Forward If ‘Call Forward Busy’ feature is enabled, new coming call will be
Busy forwarded when the corresponding local port is busy.

Call Forward No If ‘Call Forward No Reply’ feature is enabled, calls will be forwarded
Reply when nobody answer the calls during a specified period.

Input Gain The receiving gain of the FXS port

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Output Gain The sending gain of the FXS port

Work Mode The working mode of the FXS port, including Voice and POS
When a call is given to the FXS port of UC120-1V1S1O, the call will
Call In Filter
not be connected to the FXO port if it is in the blacklist
When a call goes out from the FXS port of UC120-1V1S1O, the call
Call Out Filter
cannot go out if it is in the blacklist
Call In Black
Calls from the number profiles in the blacklist will be blocked
List
Call Out White
Calls from the number profiles in the whitelist will be not blocked
List
FXS Profile The FXS profile that is selected for this FXS extension
If it is on, this FXS extension can be used, otherwise, the FXS
Status
extension is unavailable.

5.6.3 Ring Group


On the Extension → Ring Group interface, you can group FXS extension and SIP
extension(s) together and set strategy for choosing the FXS extension and which SIP
extension to ring under a ring group. The ring group function is widely used in call centers.

Figure 5-73 Configure Ring Group

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Table 5-27 Explanation of Parameters for Ring Group

Name The name of this ring group


Select the FXS extension and an SIP extension or several SIP
extensions;
Members Select
: Add an extension to the ring group

: Delete an extension from the ring group


The strategies for choosing which SIP extension to ring, including
Strategy Sequence (Ascending), Sequence (Cyclic Ascending), Simultaneous
and Random
Ring Group
The number of the ring group; it is generally the same with DID.
Number

DID Same with Ring Group Number; it is optional to fill in


Ring Time
The duration of ring when there is a coming call. Range: 5s to 60s
(5-60s)

Note: If ring group function has been set, the call forwarding function is unavailable.

5.6.4 Paging Group


On the Extension → Paging Group interface, you can group SIP extensions into a paging
group and then if there calls given from FXS/FXO/SIP to the paging group, the calls will be
led to one extension of the paging group according to the preset strategy.

Figure 5-74 Configure Paging Group

Table 5-28 Explanation of Parameters for Paging Group

Name The name of this paging group


The number of the paging group.
Paging Group
Number When there calls given from FXS/FXO/SIP to this number, the calls
will be led to one extension of the paging group according to the preset

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strategy.

Include one-way paging and two-way intercom.


one-way paging:members of the paging group only can listen to the
Strategy voice of presenter and cannot answer the call.
two-way intercom:members of the paging group can have
conversation with the presenter, but members cannot talk to each other.
Select the SIP extensions that are added into the paging group. An SIP
extension cannot exist in two paging groups at the same time.
Members Select
Click to add an SIP extension to the paging group;

Click to delete an SIP extension from the paging group.

5.6.5 Call Queue


On the "Extension → Call Queue" interface, the user can add the local extension to a queue.
When calling into the call queue, the system will transfer the call to the queue member/agent
to answer the call according to the strategy.

For example, when a large number of customers call in at the same time, and the customer
service staff is limited, queue the incoming and play a voice waiting tone or custom music file.
At the same time, the agent can answer the call according to the preset call queue strategy.

Figure 5-75 New Call Queue

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Table 5-29 Explanation of Call Queue Parameters

Name The name of the call queue.


Calls into the queue, the agents ring according to the strategy.
Strategy Simultaneous: The agents ring together.

Sequential Mode: When there is no incoming call, a new user calls in,

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each time it will ring sequentially from the first agent).

Random: one is randomly selected for ringing.

Memory rotation mode : When there is no incoming call , a new user


calls in, and the ringing starts from the next agent who hangs up last
before.

Max idle time : Idle time, namely the time from the end of the agent's
last call to the present; ringing in the order from longest to shortest
time.

Min talk time :The ringing starts from the least to the most according
to the times of calls.

Call Queue
The number of the call queue can be called into the queue.
Number
Agent Wrap The interval time between the next ringing after the agent call hangs
Time(5s-300s) up.
Agent Ring
If the ringing exceeds the time, it will call to the next agent.
Time(5s-300s)
Menu Tone The first menu tone the remote end hears when calling in.
The remote end waits for the agent to answer the waiting tone after
Waiting Music
calling in.
The longest time the caller waits. The caller will exit after this time. 0
means no limit, but it should be noted that this time is not necessary.
Max Wait
For example, an agent is ringing and the caller has reached the timeout.
Time(0s-300s)
The caller will wait until the agent answers or hang up after the
timeout.
Call Forward
If the caller times out, other actions can be configured.
Timeout
If there is no agent in the queue, it will exit the queue (if there is
Leave When
ON-Break, it is still an agent), and the call transfer will be performed
Queue Empty
when the queue is empty.
Max Queue How many users are waiting, those connected are not counted, 0 means
Length no limit, hang up if the maximum number of queues is exceeded.
If the times that the agent does not answer is exceeded, it will enter
Max No Answer On-Break state, in this state, it will not be ringing again until the agent
answer.
Enable Position Timely notify the user of the waiting position in the queue, the first one
Announcement does not notify.

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Figure 5-76 Dynamic Agent Login Setting

Table 5-30 Explanation of Dynamic Agent Login Setting Parameters

Login Suffix Extensions dial “Call Queue Number” + login suffix, log into the
specified queue, and register as an available member of the queue.

Logout Suffix Extensions dial “Call Queue Number” + logout suffix to exit from the
specified queue and stop receiving calls assigned to the queue.

5.7 Trunk
5.7.1 SIP
SIP trunk can realize the connection between UC120-1V1S1O and IPPBX or SIP servers.
Figure 5-75 Configure SIP Trunk

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Table 5-31 Explanation of Parameters for SIP Trunk

Name The name of the SIP trunk

Address The IP address or domain name of the peer SIP devices or servers
The SIP listening port of the peer SIP devices or servers; 5060 is the
Port
default port
If outbound proxy is used, enter the IP address or domain name of the
Outbound Proxy
proxy server
Port If outbound proxy is used, enter the listening port of the proxy server

Transport Transport protocol: TCP or UDP

Register If it is on, the SIP trunk will send register request to the peer device

Username The username of this SIP trunk, it is generally a phone number

Auth Username The username used for register authentication by this SIP trunk

Password The password used for register authentication by this SIP trunk
From Header Choose the registered username or the true caller ID for the ‘from
Username header’ of the invite message when a call goes out.

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The validity period after the SIP trunk is registered successfully. When
Expire Seconds the time expires, the SIP trunk will send register request to the server.
Default value is 1800s
When the SIP trunk fails to be registered, the interval to send register
Retry Seconds request;
Default value is 60s
If heartbeat in on, heartbeat (options) messageswill be sent to examine
the connection with servers;
Heartbeat
The default value is ‘Off’

Heartbeat Period The interval of sending heartbeat (options) messages


he SIP profile of the SIP Trunk; make reference to Profile → SIP
SIP Profile
section
If it is enabled, it means the SIP Trunk can be used; otherwise, the SIP
Status
trunk is unavailable

Note:

If UC120-1V1S1O is regarded as a terminal and intends to register to a server, you need to


configure a SIP trunk connecting UC120-1V1S1O and the server, and then enable register for
the SIP trunk.

If the FXS f UC120-1V1S1O intends to register to a server, you need to configure a SIP trunk
connecting UC120-1V1S1O and the server, then enable register for the port and designate the
SIP trunk to it.

5.7.2 FXO
FXO Trunk interconnects the PSTN with UC120-1V1S1O. Calls from the PSTN can come
into the gateway and calls can go out from the gateway to search telephone numbers under the
PSTN.

Different from the FXO ports of other gateways, the FXO port of UC120-1V1S1O only
allows one-time dialing, which means called numbers needs to be dialed directly for calls that
go out from the FXO port.

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Figure 5-76 Configure FXO Trunk

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Table 5-32 Explanation of Parameters for FXO Trunk

Port The FXO portl number

Extension The extension account of the FXO port, which is used to register
Autodial Number The autodial number of the FXO port when there are incoming calls

If it is enabled, the FXO trunk will be registered to the SIP trunk


Register to SIP Server that has been set.
Default is off.
The address and port of the master SIP server;
Master Server it is generally the IP address of a SIP trunk. Please make
reference to Trunk → SIP section
Slave Server The address and port of the slave SIP server

Username of the FXO port account, used for the authentication


Username of registration

The username of this FXO trunk, which is used during register


Auth Username authentication

The password of this FXO trunk, which is used during register


Password
authentication
Choose the registered username or the true caller ID for the
From Header Username
‘from header’ of the invite message when a call goes out.
Specify Transport
Protocol on Register Whether to specify transport protocol on register URL.
URL
The validity period after the FXO trunk is registered
successfully. When the time expires, the FXO trunk will send
Expire Seconds
register request to the server.
Default value is 1800s
When the FXO trunk fails to be registered, the interval to send
Retry Seconds register request;
Default value is 60s
The format to display caller information, including:
Caller ID/Caller ID
Display
Display Name/ Caller ID
Name/Username
Extension/ Caller ID
Format
Caller ID/ Extension
Anonymous
Display Name /
Set the caller's caller id format when the main number is not
Username Format when
detected
CID unavailable
Input Gain The receiving gain of the FXO port

Output Gain The sending gain of the FXO port

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Impedance The impedance (SLIC) matched with phones

FXO Profile The FXS profile that is selected for this FXS extension
If it is on, this FXO trunk can be used, otherwise, the FXO trunk
Status
is unavailable.

FXO Automatch Impedance:

Click the Detection button, and the UC120-1V1S1O gateway will automatically detect the
most-matched impedance.

Figure 5-77 FXO Automatch Impedance

5.7.3 VoLTE Trunk


VoLTE trunk helps interconnect the LTE network with the IP network. The VoLTE function
allows calls or SMS from the IP network to be transmitted to mobile network, and packs voice
or SMS from mobile network into IP packages and send them to IP network.

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Figure 5-80 Configure VoLTE Trunk

The extension account of the VoLTE trunk, which is used to


Extension
register
The autodial number of the The address and port of the master
SIP server;
Autodial Number
it is generally the IP address of a SIP trunk. Please make
reference to Trunk → SIP section when there are incoming calls
If it is enabled, the GSM trunk will be registered to the SIP trunk
Register to SIP Server that has been set.
Default is off.

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The master SIP server to which the VoLTE trunk is registered.


Enter the address and port of the master SIP server;
Master Server
it is generally the IP address of a SIP trunk. Please make
reference to Trunk → SIP section
The slave SIP server to which the VoLTE trunk is registered.
Slave Server
Enter the address and port of the slave SIP server
The username of this VoLTE trunk used for registration; it is
Username
generally a phone number
The username of this VoLTE trunk, which is used during register
Auth Username
authentication
The password of this VoLTE trunk, which is used during register
Password
authentication
Choose the registered username or the true caller ID for the
From Header Username
‘from header’ of the invite message when a call goes out.
Specify Transport
Protocol on Register Whether to specify transport protocol on register URL.
URL
The validity period after the VoLTE trunk is registered
successfully. When the time expires, the GSM trunk will send
Expire Seconds
register request to the server.
Default value is 1800s
When the VoLTE trunk fails to be registered, the interval to send
Retry Seconds register request;
Default value is 60s
The format to display caller information, including:
Caller ID/Caller ID
Display
Display Name/ Caller ID
Name/Username
Extension/ Caller ID
Format
Caller ID/ Extension
Anonymous
Display
Name/Username The format to display caller information when CID is
Format when CID unavailable
unavailable
Click the Refresh button, and the gateway will automatically to
Carrier
identify the carrier of the inserted SIM card.
Reactive when register
Whether to reactivate the VoLTE trunk when register fails
fail
ucs2 or 7bit,
SMS Encoding 7-bit is used to send original ASCII, while UCS2 is used to send
various languages
Generally, the GSM module can automatically detect the SMS
SMS Center Number
center number. This parameter will be used when the LTE

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module cannot detect the SMS center number.

CLIR Whether to enable Calling Line Identification Restriction

PIN code is personal identification code. When SIM card is


PIN Code locked, you can modify the PIN code to prevent the SIM infor
from being stolen
DSP Input Gain Volume control of DSP input
DSP Output Gain Volume control of DSP output
Module Speaker Gain Volume control of module speaker
SIM Number Learning Choose a number learning profile for the SIM card. The SIM
Profile number will be displayed on the Status →PSTN→VOLTE page
If it is on, this VoLTE trunk can be used, otherwise, the VoLTE
Status
trunk is unavailable

5.8 Call Control


This section is to configure routes or route groups for incoming and outgoing calls through
UC120-1V1S1O, as well as IVR, SMS, USSD and so on.

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5.8.1 Setting
Figure 5-81 Basic Setting of Call Control

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Table 5-33 Explanation of Parameters for Call Control


Disconnect call when no RTP If it is enabled, and no RTP packets are received within
packet the preset time, calls will be disconnected

Packet Loss Concealment (PLC) Whether to enable the ‘Packet Loss Concealment’
function

Echo Canceller Tail Length (ms) Default value: 64ms

DTMF Min Detect Interval (ms) The minimum time for DTMF detection

Echo Gain Default value: -4dB

RTP Start Port The start port of RTP packets

RTP End Port The end port of RTP packets

Local extension call If it is enabled, calls between local extensions do not


need routes.

Fax Mode T38 or T30(Pass-through)

Tone Detection by Local If it is enabled, UC120-1V1S1O will detect fax tones


automatically during a call and the call will be switched
into fax mode after a fax tone is detected.

SDP Param ‘a=X-fax’ Attribute parameter ‘a=X-fax’ is carried in SDP

SDP Param ‘a=fax’ Attribute parameter ‘a=fax’ is carried in SDP

SDP Param ‘a=X-modem’ Attribute parameter ‘a=X-modem’ is carried in SDP

5.8.2 Route Group


On the Call Control →Route Group interface, you can group SIP trunks, SIP extensions,
FXS extension and FXO trunk together according to your needs and set strategy for choosing
which trunk or extension as the destination route under a route group.

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Figure 5-78 Create Route Group

Table 5-34 Explanation of Parameters for Route Group


Name The name of the route group

Members Select Select FXS extension, SIP extension, SIP trunk, FXO trunk or GSM
trunk

Strategy The strategies for choosing which route under the route group as the
destination route, including Sequence (Ascending), Sequence (Cyclic
Ascending), Simultaneous and Random

5.8.3 Route
On the Call Control → Route interface, you can configure routes for incoming calls and
outgoing calls.

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Figure 5-79 Create a Route

Table 5-35 Explanation of Route Parameters


Priority The priority for choosing the route; the higher value, the lower
priority

Name The name of the route

Condition The condition under which the route will be used

Source The source of the call; it can be the FXS extension, SIP
extension, FXO trunk ,GSM trunk, a customized source or any

Number Profile The profile of the caller number and the called number; please
make reference to the Profile → Number section.The default
value is ‘Off’

Note: it cannot be simultaneously used with the following


parameters of ‘caller number prefix’ and ‘called number prefix’

Caller Number Prefix The prefix of caller number; it supports regular expression

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Called Number Prefix The prefix of called number; it supports regular expression

Time Profile The profile of time during which the route can be used; make
reference to the Profile → Time section

Action Include manipulating number and sending call to destination

Manipulation If it is on, the caller number or called number of the route will
be manipulated; make reference to the Profile→ Manipulation
section

Destination The destination of the route

Failover Action The processing when a call through this route fails

5.8.4 Feature Code


UC120-1V1S1O provides convenient telephone functions. Connect a telephone to the FXS
port and dial a specific feature code, and you can query conrresponding information.

The following is the corresponding function of each feature code:

Figure 5-80 Feature Code

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5.8.5 IVR
On the Call Control → IVR interface, you can carry out specific configurations for the IVR
which has been uploaded from the System → Voice interface.

Figure 5-81 IVR Setting

Table 5-36 Explanation of IVR Parameters


Status If it is disabled, the IVR cannot be seen in the destination of route.

Timeout If it is set as ‘10’, it means if no DTMF tone is received during 10 seconds,


the IVR will be played repeatedly or the call will be hanged up. The default
value is 10 seconds.

Enable Direct Whether to allow direct dialing of extensions during the playing of IVR
Extension

Repeat Loops If it is set as ‘3’, the call will be hanged up after the IVR has been repeated
for three times during timeout.

Menu DTMF: It can be 0-9 quick-dial numbers, *, #, others or timeout.


Destination: the destination of the IVR; it can be an extension or a trunk.
For example, if DTMF is configured as 1,2,3 and others, and the telephone
key that is pressed is not 1, 2 or 3, the IVR will choose the destination of
‘others’.
When the the playing of the IVR times out, and user does not press any
telephone key, the IVR will choose the destination of ‘timeout’.
When the destination is a trunk, user does not need to pre-configure the
called number, and the system will prompt the user to dial the called
number.

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5.8.6 SMS Route


UC120-1V1S1O allows SMS to be sent between SIP clients, and meanwhile allow SMS to be
sent between IP network/GSM network and GSM network. On the Call Control→ SMS
Route interface, you can establish route for these SMS.

For example, you can download a softphone on a PC which is connected to UC120-1V1S1O,


and type the content of the SMS through the softphone. Then configure a SMS route on the
Call Call Control→ SMS Route interface. The source of the SMS route is the number of the
softphone.

Figure 5-82 Create SMS Route

Table 5-37 Explanation of SMS Route Parameters


Priority The priority for the SMS route; the higher value, the lower priority

Name The name of the SMS route

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Source The source of the SMS route. It can be a trunk or an extension.It also can
be a LTE SMS and USSD.

Src Number Prefix the source number to support regular expressions


Prefix

Content Has Match key words in text message content


the Words

Action The text message action can choose whether to forward or reply

Destination The destination of the SMS route. It can be a trunk or an extension.

Dest Number The source of the destination number.There are two sources:custom and
Src get from content.

Add Prefix in The prefix of the SMS content. It is generally ‘none’, which means there
Content is no prefix to be matched.

Add Suffix in The suffix of the SMS content. It is generally ‘none’, which means there
Content is no suffix to be matched.

5.8.7 SMS
If an SIM card has been inserted into the SIM slot, you can send or receive SMS on the SMS
interface. The length of a SMS can not be more than 170 characters.

Figure 5-83 Send and Receive SMS

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Notice

If the SIM fails, you will get a notice:SIM1: The module is not recognized successfully.
Please check whether the module is supported or installed !

Send Message
Enter contents into the box on the left, and then input the number of recipient . Click Send in
the last.
Note: If there are mutilple recipients , use | to separate them, for example,
13151103146|18954405566.

Receive Message
All SMS received by UC120 are displayed on the Receive List.

Read Message
Click on the Receive List to read SMS contents.

Reply Message
Click , and then enter SMS contents in the box on the left. Click Send in the last.

Delete Message
Click to delete an SMS.

Note: Group sending of SMS is not allowed.

5.8.8 USSD
USSD (Unstructured Supplementary Service Data) is a global system service for mobile
(GSM) communication technology that is used to send text between a mobile phone and an
application program in the network.

USSD is similar to Short Messaging Service (SMS), but, unlike SMS, USSD transactions
occur during the session only. With SMS, messages can be sent to a mobile phone and stored
for several days if the phone is not activated or within range.

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Figure 5-84 USSD

Table 5-38 Explanation of USSD Parameters

USSD Code USSD code is a special number starting with * or #, followed by 2~3
digits, then ending with #;
Service provider feedbacks a service menu to user according to this
USSD code.
Encode Supports auto, 7-bit and UCS2;
7-bit is used to send original ASCII, while UCS2 is used to send any
languages
USSD Message The Content of the USSD message

5.8.9 Diagnostics
In case that call cannot be connected or voice has quality problem, you can enter into the Call
Control →Diagnostics interface to collect fault-related information and then send it to
technical support to locate fault.

Operation Procedures:

1. Select the module that needd to be traced. For example, if a call from SIP to FXS has
voice problem, you can select SIP message, FXS/FXO and Voice, and then click the Start
button.

2. Give a call, and come back to the Call Control →Diagnostics interface after the call ends.
Then click Stop and download the tracing file.

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3. In order to locate faluts more quickly, you sometimes need to enter into the System
→Service Log interface, click export, and then send this exported file and the tracing file to
technical support,

Figure 5-85 Call Tracing for Diagnostics

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6 Glossary

6 Glossary

Glossary Description

ARP Address Resolution Protocol

CID Caller Identity

DNS Domain Name System

DDNS Dynamic Domain Name Service

DHCP Dynamic Host Configuration Protocol

DMZ Demilitarized Zone

DND Do NOT Disturb

DTMF DTMF:Dual Tone Multi Frequency

FTP File Transfer Protocol

HTTP Hypertext Transfer Protocol

LAN Local Area Network

L2TP Layer 2 Tunneling Protocol

PPTP Point-to-Point Tunneling Protocol

MAC Address Media Access Control Adress

NAT Network Address Translation

Ping Packet Internet Grope

SIP Session Initiation Protocol

TCP Transmission Control Protocol

UDP User Datagram Protocol

RTP Real Time Protocol

PPPOE Point-to-point Protocol over Ethernet

QoS Quality of Service

UPnP Universial Plug and Play

VLAN Virtual Local Area Network

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Glossary Description

NTP Network Time Protocol

STUN Simple Traversal of UDP over NAT

PSTN Public Switched Telephone Network

WLAN Wireless Local Area Network

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