MG3000-R32 User Manual - Gohigh Data Networks Tech...

MG3000-R32 

User Manual 

GoHigh Data Networks Technology CO., LTD 

© Copyright, 2008 

i 

G3000-R32-User Manual

Preface 

Dear User: 

ii 

G3000-R32-User Manual 

Firstly, thank you for your selection of Gohigh’s MG3000-R32 Intelligent Routing Gateway! 

This manual will introduce the technical characteristics, system configuration, installation and 

operation of MG3000-R32, which will help you to understand the product in details, and use it 

more easily. 

Please read the manual carefully before you start to use it. 

The copyright of the manual belongs to Gohigh Company. Without Gohigh Company’s 

permission, any forms of copy of it are strictly prohibited. 

Gohigh Company owns the authority of final interpretation of the manual, and reserves the 

rights to revise the manual and improve the product. 

If the content of this manual changes, there will be no notification. 

If you have any doubts about the manual or suggestions, please contact Gohigh Company. 

Gohigh Data Networks Technology CO., LTD 

2008-04 

“瑞影”、“RealView”、“MG3000-R” are registered trademarks and product model, which belong 

to Gohigh Company, please using them within the legitimate scope. 

Other trademarks belong to corresponding registered companies separately. 

Declaration 

Please read the following statement before you officially use MG3000-R32 of Gohigh Company. 

Only after you finish reading and agree to the following terms, can you start to use it officially. If 

you have any questions, please contact the supplier or contact us directly. If you start to use 

MG3000-R32 without asking us, we will treat you as agreeing to the following provisions by 

default: 

1、Please pay attention: At any time, except the parameters which we remind you that you 

can modify, please do not modify any other parameters of MG3000-R32, because without 

professional guidance the modification may lead to abnormal status. 

2、When MG3000-R32 have any hardware failures, please contact our technical service 

center. MG3000-R32 can not support any non-original hardware changes because of its 

special designed integrated structure. And please do not disassemble the chassis and any 

hardware within it. 

3、Please keep the label of serial number attached in the back panel. Lacking of it may 

affect your legitimate right for services. 

4、When you meet any problems during using MG3000-R32, please read this manual 

carefully. And if you still can not solve them, please contact our technical service center. 

5、When using MG3000-R32 to cooperate with products of other brand, if you meet 

problems, please contact your corresponding provider first, or entrust them to contact our 

company, so that we can communicate easily, and solve your problem together. For example: 

when there are problems during cooperation with softswitch, or Trunk gateway etc., we 

particularly hope that you can handle it in this way. 

6、Because different vendors may have different comprehension of VOIP protocol and its

iii 


realization, we do not guarantee MG3000-R32 can be compatible to every vendor which is 

non-mainstream; And any requirement for software function modification, please promoted it 

through our company’s market system, the corresponding commercial actions will not be 

included in this manual. 

7、After product installed, please recycle the product package in reasonable way, avoiding 

the environment pollution. 

If the provisions above conflict commercial contract, please refer to the commercial contract. 

Contents 

This manual is applied to version MG3000-R32 V1.3.5 and the version after it, mainly including 

the following contents: 

First part: product introduction 

Include appearance, specification, technical characteristics and description. 

Second part: installation procedures 

Include installation notes, parts inventory, installation preparation and explanation. 

Third part: product management 

Introduce MG3000-R32 product management, include: 

Chapter 4, Instructions for keyboard inquiry, Chapter 5, Web configure description, appendix II 

Command Line Guide 

Fourth part: service function description 

Introduce the realization of all basic services and supplementary services, and the unique 

IP-To-PSTN, PSTN-To-IP function, Call Progress Tone, Number Match, Change Rule, 

Intelligent Route, Answer Supervision, and Disconnect Supervision etc... 

Fifth part: maintenance and trouble shooting 

Include version upgrade, normal trouble diagnosis and troubleshooting 

Sixth part normal problems 

Introduce how to handle normal problems. 

When you are reading this manual, please notice that it has 3 levels: 

1、About individual configuration items (Chapter 5); 

2、About services and functions (Chapter 6); 

3、About common problems (Chapter 8); 

In these three chapters, inevitably some contents may overlap; readers may use the links in the 

text to make reference. 

About 

If you want access technical support information online, please visit our company’s technical 

supporting website: http://gongcheng.gohigh.com.cn. You can contact our technical specialist 

online; and you can also call +8610 62303100, we will have specialist to support you. 

Inevitably there may have some inadequate things or omissions in the manual, we will be very 

appreciated if you can point them out and inform us.

Contents 


Preface ............................................................................................................................. ii 

Declaration ....................................................................................................................... ii 

Contents .......................................................................................................................... iii 

About ............................................................................................................................... iii 

1 Product introduction...............................................................................................................9 

1.1 Product Appearance..............................................................................................9 

1.2 Product specifications ......................................................................................... 11 

1.3 Technical Characteristics ....................................................................................12 

1.4 Product Description.............................................................................................13 

1.4.1 Only FXS sub-card ...............................................................................13 

1.4.2 Only 8FXO sub-card.............................................................................13 

1.4.3 Only 4S4O sub-card .............................................................................14 

1.4.4 Only 4S4L sub-card ..............................................................................14 

1.4.5 Mixed insertion of 8FXS and 8FXO......................................................15 

1.4.6 Mixed insertion of 8FXS and 4S4O......................................................16 

1.4.7 Mixed insertion of 8FXO and 4S4O......................................................16 

2 Installation steps..................................................................................................................18 

2.1 Installation notes .................................................................................................18 

2.1.1 Safety recommendations......................................................................18 

2.1.2 Limitation of signal transmission distance............................................18 

2.1.3 Environment requirement .....................................................................19 

2.1.4 Correct wiring solution ..........................................................................19 

2.2 Parts list...............................................................................................................19 

2.3 Install preparation................................................................................................20 

2.4 Installation approach ...........................................................................................21 

2.4.1 Installation.............................................................................................21 

2.4.2 Wiring....................................................................................................22 

2.4.3 Power on...............................................................................................22 

3 Equipment management description...................................................................................23 

3.1 Keyboard inquiry .................................................................................................23 

3.2 WEB ....................................................................................................................23 

3.3 Command line .....................................................................................................23 

3.4 Unified network management .............................................................................23 

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4 Instructions for keyboard inquiry .........................................................................................24 

5 Web configure description...................................................................................................25 

5.1 Log in...................................................................................................................26 

5.2 Quicksetup ..........................................................................................................26 

5.2.1 H.323 protocol ......................................................................................26 

5.2.2 SIP Protocol..........................................................................................30 

5.3 Basic Configure...................................................................................................34 

5.3.1 Protocol.................................................................................................34 

5.3.2 Port .......................................................................................................40 

5.3.3 Channel.................................................................................................42 

5.4 Advanced Configuration......................................................................................47 

5.4.1 FXS .......................................................................................................47 

5.4.2 FXO.......................................................................................................50 

5.4.3 Voice .....................................................................................................53 

5.4.4 SPS.......................................................................................................57 

5.4.5 Net Manage Setting..............................................................................59 

5.5 NBR Exchange....................................................................................................60 

5.5.1 Digit Map...............................................................................................60 

5.5.2 SpeedDial .............................................................................................62 

5.5.3 Change Rule.........................................................................................63 

5.5.4 DialedNBRChange ...............................................................................65 

5.5.5 Caller Change.......................................................................................68 

5.5.6 RegenNBRChange...............................................................................70 

5.6 Routing ................................................................................................................71 

5.6.1 RouteDirection......................................................................................71 

5.6.2 Peer to Peer Routing ............................................................................72 

5.6.3 ACL .......................................................................................................73 

5.6.4 PSTN User Management .....................................................................74 

5.7 Information ..........................................................................................................76 

5.7.1 Port Status ............................................................................................76 

5.7.2 System Info...........................................................................................77 

5.8 Device Management ...........................................................................................78 

5.8.1 Update ..................................................................................................78 

5.8.2 Password ..............................................................................................79 

5.8.3 Default setting.......................................................................................79 

5.8.4 Save......................................................................................................80 

5.8.5 Reboot ..................................................................................................81 

6 Service Function Description...............................................................................................82 

6.1 Basic Service.......................................................................................................82 

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6.1.1 IP Voice Service....................................................................................82 

6.1.2 IP FAX...................................................................................................86 

6.1.3 DTMF Secondary Dial ..........................................................................87 

6.2 SPS (Supplementary Service) ............................................................................87 

6.2.1 SPS through terminal ...........................................................................87 

6.2.2 SPS through Terminal cooperation with Server ...................................93 

6.3 FXO Mode ...........................................................................................................97 

6.3.1 IP-To-PSTN...........................................................................................97 

6.3.2 PSTN-To-IP.........................................................................................102 

6.3.3 PSTN Routing.....................................................................................106 

6.4 Call Progress Tone............................................................................................107 

6.4.1 Tones of different countries ................................................................107 

6.4.2 Self-defined tone format .....................................................................108 

6.4.3 Busy Tone Detection...........................................................................110 

6.4.4 Busy tone Auto Learning ....................................................................112 

6.5 Number Match...................................................................................................114 

6.5.1 Pattern Match Rule.............................................................................114 

6.5.2 Match rule ...........................................................................................115 

6.5.3 Application ..........................................................................................116 

6.6 PSTN Caller ID..................................................................................................119 

6.7 Change Rule .....................................................................................................119 

6.8 Intelligent Route ................................................................................................121 

6.8.1 Routing Direction ................................................................................122 

6.8.2 IP Fall Back.........................................................................................122 

6.8.3 Dial Special Number to Change Route ..............................................124 

6.9 DNS SRV Function............................................................................................125 

6.10 Answer Supervision ..........................................................................................126 

6.11 Disconnect Supervision.....................................................................................126 

6.11.1 Busy Tone Detection...........................................................................126 

6.11.2 Reversal Signal Detection ..................................................................126 

6.12 Early Media .......................................................................................................126 

6.13 Port Polling ........................................................................................................126 

6.14 Security Mechanism..........................................................................................127 

6.14.1 Encryption...........................................................................................127 

6.14.2 ACL .....................................................................................................127 

6.14.3 PSTN Incoming Call Authentication ...................................................128 

7 Maintenance and Trouble Shooting ..................................................................................128 

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7.1 Update...............................................................................................................128 

7.1.1 HTTP Update......................................................................................128 

7.1.2 TFTP Update ......................................................................................129 

7.1.3 Update procedures .............................................................................129 

7.2 Diagnosis...........................................................................................................130 

7.2.1 Port status check ................................................................................130 

7.2.2 Output Trace information....................................................................130 

7.2.3 IP Packets...........................................................................................130 

7.3 Debug Function.................................................................................................130 

7.3.1 WAN Port Mirror..................................................................................130 

7.3.2 Output TDM Data................................................................................131 

8 Normal Troubles ................................................................................................................132 

8.1 Can not establish a call due to configure error in number matching rules .......132 

8.2 PSTN Caller ID display error.............................................................................132 

8.3 If the callee number is #57#, how to configure it? ............................................132 

8.4 System can not start after version update ........................................................133 

8.5 Can not ring with multiple phones in parallel connection..................................133 

8.6 Hear noise in ringback tone ..............................................................................133 

8.7 Siemens Phone CID displays incorrectly..........................................................133 

8.8 IP route error, can not switch to PSTN..............................................................134 

8.9 Establish a call in One-stage mode, but the call is interrupted very soon........134 

9 Appendix I VOIP Technical Terms .....................................................................................136 

Caller ID .......................................................................................................................136 

FSK...............................................................................................................................136 

DTMF ...........................................................................................................................137 

Silence Compression and CNG...................................................................................137 

iLBC..............................................................................................................................137 

RTP Redundancy.........................................................................................................138 

T.38 Fax........................................................................................................................138 

Bypass Fax...................................................................................................................138 

Answer Supervision .....................................................................................................139 

Disconnect Supervision................................................................................................139 

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FXS、FXO、E&M Interface.........................................................................................140 

The difference between FXO and Lifeline ...................................................................140 

Diagram & Table Contents 

Table 1 MG3000-R32 LED DESCRIPTION ...............................................................9 

Table 2 MG3000-R32 FRONT PANEL INTERFACE DESCRIPTION ......................10 

Table 3 MG3000-R32 BACK PANEL DESCRIPTION ..............................................10 

Table 4 MG3000-R32 PRODUCT SPECIFICATIONS ............................................. 11 

Table 5 MG3000-R32 TECHNICAL CHARACTERISTICS.......................................12 

Table 6 ONLY 8FXO SUB-CARDS FUNCTION DESCRIPTION .............................13 

Table 7 ONLY 4S4O SUB-CARD FUNCTION DESCRIPTION................................14 

Table 8 ONLY 4S4L SUBCARD FUNCTION DESCRIPTION..................................14 

Table 9 MIXED INSERTION OF 8FXS AND 8FXO FUNCTION DESCRIPTION ....15 

Table 10 MIXED INSERTION OF 8FXS & 4S4O FUNCTION DESCRIPTION..16 

Table 11 MIXED INSERTION OF 8FXO & 4S4O FUNCTION DESCRIPTION..16 

Table 12 MG3000-R32 ENVIRONMENTAL REQUIREMENT ............................19 

Table 13 MG3000-R32 PARTS LIST...................................................................20 

Table 14 H.323 QUICKSETUP CONFIGURATION DESCRIPTION ..................27 

Table 15 SIP QUICK SETUP PAGE PARAMETERS DESCRIPTION................32 

Table 16 H.323 PARAMETERS DESCRIPTION.................................................35 

Table 17 SIP PARAMETERS DESCRIPTION.....................................................38 

Table 18 CHANNEL CONFIGURATIOIN DESCRIPTION...................................44 

Table 19 FXS PARAMETERS DESCRIPTION ...................................................48 

Table 20 FXO FUNCTION DESCRIPTION.........................................................51 

Table 21 VOICE PARAMETERS DESCRIPTION...............................................55 

Table 22 SPS PARAMETERS DESCRIPTION ...................................................57 

Table 23 NMS PARAMETERS DESCRIPTION ..................................................60 

Table 24 VOICE CODER TYPE OF MG3000-R32 .............................................84 

Table 25 PT VALUE OF ALL KINDS OF VOICE CODERS ................................84 

Table 26 PT VALUE OF ALL KINDS OF VIDEO CODER...................................85 

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Table 27 SPS OF DIFFERENT PROTOCOLS ...................................................87 

Table 28 MG3000-R32 TONES FORMAT.........................................................107 

Table 29 SELF-DEFINED TONE FORMAT DESCRIPTION.............................109 

Table 30 PATTERN MATCH RULE ...................................................................114 

Table 31 COMMON NUMBER RULE................................................................115 

Table 32 ROUTING DIRECTION TABLE..........................................................117 

Table 33 CALLOUT NUMBER CHANGE EXAMPLE........................................117 

Table 34 NUMBER CHANGE RULE DESCRIPTION .......................................120 

Table 35 NUMBER CHANGE EXAMPLES .......................................................121 

Table 36 VERSION NAME RULES...................................................................128 

Table 37 DTMF COMBINATIONS .....................................................................137 

Figure 1 MG3000-R32 Appearance......................................................................9 

Figure 2 MG3000-R32 front panel ........................................................................9 

Figure 3 MG3000-R32(220VAC)back panel ..................................................10 

Figure 4 MG3000-R32(48VDC)back panel....................................................10 

Figure 5 Console cable diagram.........................................................................20 

Figure 6 FXS subscriber line diagram ................................................................21 

Figure 7 FXO subscriber line diagram................................................................21 

Figure 8 Serial port parameters configuration ....................................................23 

Figure 9 Web page framework............................................................................25 

Figure 10 Web Login interface..............................................................................26 

Figure 11 H.323 QuickSetup Page.......................................................................27 

Figure 12 SIP protocol quicksetup page...............................................................31 

Figure 13 H.323 Configurations Diagram .............................................................35 

Figure 14 SIP protocol parameters settings .........................................................38 

Figure 15 H.323 Port Configuration Diagram .......................................................41 

Figure 16 SIP Protocol Port Settings ....................................................................42 

Figure 17 Channel settings...................................................................................43 

Figure 18 FXS Parameters Settings.....................................................................48 

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Figure 19 FXO Parameters settings .....................................................................50 

Figure 20 Set Forward Number ............................................................................51 

Figure 21 Voice Parameters settings....................................................................54 

Figure 22 SPS Parameters Settings.....................................................................57 

Figure 23 Net Manage Setting Diagram ...............................................................60 

Figure 24 Digit Map setting Diagram ....................................................................60 

Figure 25 Adding Dial Plan Diagram..................................................................61 

Figure 26 Set “#” as Number End Flag Diagram..................................................62 

Figure 27 Speed Dialing Diagram.........................................................................62 

Figure 28 Add speed dial rule Diagram ................................................................63 

Figure 29 Number Change Rule Table .................................................................64 

Figure 30 Add Number Change Rule Diagram.....................................................65 

Figure 31 Callout Number Change Table .............................................................65 

Figure 32 Callout Number Change Type ..............................................................66 

Figure 33 Callout Number Change Operation Diagram .......................................67 

Figure 34 Caller Number Change Rule ................................................................68 

Figure 35 Regeneration Number Change Diagram..............................................70 

Figure 36 Route Direction Table ...........................................................................71 

Figure 37 Peer to Peer Routing ............................................................................72 

Figure 38 Access Control Configuration Diagram ................................................74 

Figure 39 FXO Inbound Limit Mode Diagram.......................................................75 

Figure 40 White List Diagram ...............................................................................75 

Figure 41 Port Status Diagram..............................................................................76 

Figure 42 System Information Diagram................................................................77 

Figure 43 System Time Configuration Diagram....................................................77 

Figure 44 NTP setting diagram.............................................................................77 

Figure 45 Update Page.........................................................................................78 

Figure 46 Password change Diagram ..................................................................79 

Figure 47 Default setting Diagram ........................................................................79 

Figure 48 SIP Protocol save Diagram ..................................................................80 

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Figure 49 Configuration File Download Diagram..................................................80 

Figure 50 H.323 Save Configuration Diagram......................................................81 

Figure 51 Reboot Diagram....................................................................................81 

Figure 52 Echo caused by combined transmission imbalance ............................82 

Figure 53 Echo caused by resistance mismatch..................................................83 

Figure 54 Echo Distribution in VoIP network ........................................................83 

Figure 55 Configure *57* Routing direction..........................................................94 

Figure 56 Disable “Dial Special Number to Change Route”.................................95 

Figure 57 Disable “Use # as number end flag......................................................95 

Figure 58 #57# dial plan configure .......................................................................96 

Figure 59 #57# routing direction configure ...........................................................96 

Figure 60 One-stage flow......................................................................................98 

Figure 61 Answer supervision diagram ................................................................99 

Figure 62 CRBT Flow Diagram...........................................................................101 

Figure 63 Two-stage Flow Diagram....................................................................102 

Figure 64 Normal Mode Flow Diagram...............................................................103 

Figure 65 Auto Mode Flow Diagram...................................................................104 

Figure 66 Dual Mode Flow Diagram...................................................................105 

Figure 67 PSTN Routing Flow Diagram .............................................................106 

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1 Product introduction 

1.1 Product Appearance 

Figure 1 MG3000-R32 Appearance 


MG3000-R32 adopts cold-rolled galvanized sheet metal frame structure, 1U high, 19-inch 

wide, dark-green. 

Figure 2 MG3000-R32 front panel 

On MG3000-R32 front panel, there are indicator LED, reset button, LAN, WAN, 

CONSOLE. (Figure 2) 

Indicator LED instructs the running status of the system, and the port status (Table 1). The 

LEDs are arrayed from left to right when you face the front panel. 

Press Reset button shortly, system will reboot; if press the button for more than 3 

seconds, system will restore factory settings. 

Table 1 MG3000-R32 LED DESCRIPTION 

LED Color Status Description 

PWR Green 

RUN Green 

WAN A/L Green 

WAN SPD Green 

LAN A/L Green 

On Power on 

Off Power off 

Twinkle Running 

On System launching 

Twinkle quickly WAN running well 

Off WAN disconnected or in trouble 

On Speed 100M 

Off Speed 10M 

Twinkle quickly LAN running well 

Off LAN disconnected or in trouble 

LAN SPD Green On Speed 100M 

9

Port status Green 

Off Speed 10M 

On Port pick up 

Off Port hang up 

Table 2 MG3000-R32 FRONT PANEL INTERFACE DESCRIPTION 

Interface Type Specification 

RST Reboot system/restore factory settings 


WAN Ethernet port RJ45, 10/100Base-T, adaptive 

LAN Ethernet port RJ45, 10/100Base-T, adaptive 

CONSOLE Serial port RS-232 

Figure 3 MG3000-R32(220VAC)back panel 

Figure 4 MG3000-R32(48VDC)back panel 

MG3000-R32 back panel has sub-card interface and power interface. 

MG3000-R32 support 2 power input: 220VAC、48VDC, the corresponding interfaces 

are different. 220AC interface adopts common three-phase power interface (Figure 3); 

48VDC interface should adopt the power lines attached in the package to connect the 

-48V power supply. 48V power interface has 1:1 backup with a switch (Figure 4). 

Sub-card has 3 types: FXS, FXO, and Lifeline: 

� FXS connects user telephone or FXO port of PBX, providing remote feed 

� FXO connects subscribe lines of PSTN, need remote feed 

� Lifeline connects subscribe lines of PSTN, need remote feed 

For details of the difference, please refer to corresponding description in Appendix I. 

When you face back panel, sub-card and interface are arrayed from left to right. In 

4S4O sub-card, first 4 ports are FXS ports, and last 4 ports are FXO ports. While in 4S4L 

sub-card, first 4 ports are FXS ports, and last 4 ports are Lifeline ports. 

Table 3 MG3000-R32 BACK PANEL DESCRIPTION 

10

Interface Type specification 

8S 4*FXS+4*FXS 1*RJ45+1*RJ45 

8O 8*FXO+4*FXO 1*RJ45+1*RJ45 

4S4O 4*FXS+4*FXO 1*RJ45+1*RJ45 

4S4L 4*FXS+4*Lifeline 1*RJ45+1*RJ45 

Power interface(220VAC) 220VAC, 50~60Hz 

Power interface(48VDC) 48VDC 

1.2 Product specifications 

Table 4 MG3000-R32 PRODUCT SPECIFICATIONS 

Items Description 

System memory 64M 

System flash 8 M 

FXS feed voltage when 

hang up the phone 

FXS feed voltage when 

pick up the phone 

-48V 

-24V 

FXS ringing voltage 64V(valid value) 

FXS ringer equivalent 

value 


short distance Loop (300 meters): 3, long distance loop (1,500 

m): 5 

FXS loop current Not less than 18 mA 

FXS loop resistance Allow user’s loop resistance arrives 2100 ohm (including phone 

impedance) 

FXO loop current Not less than 10mA 

Surge voltage 

/ tolerant voltage 

Subscribe line meet level 2 surge (instant high voltage during voltage 

fluctuation) protection requirements, can bear 1500 volts (10/1000uS) 

surge voltage. 

The longest line length Not more than 3 km 

Off hook detection Loop detection start 

Dial mode DTMF 

Power supply voltage(AC) Normal 220VAC,range: 100VAC-240VAC 

Power supply voltage(DC) Normal 48VDC,range: 36VDC-72VDC 

current Maximum: 0.5A 

Power consumption Maximum: 70W 

Environmental temperature 

during runtime 

Environmental temperature 

when not running 

-10~+55℃ 

-25~+70℃ 

Humidity during runtime 0~90% (non-solidification) 

Physical dimensions L 482mm×W261mm×H43.5mm 

Weight 3.3kg 

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1.3 Technical Characteristics 

Table 5 MG3000-R32 TECHNICAL CHARACTERISTICS 

Items Description 

POTS interface 

Network interface 

Voice characteristics 

Fax 

IP protocol 

SIP function 

Supplementary services 

DTMF transmission 


FXS: connect user telephone, provide subscribe line detection, feed, 

ringing current, support DTMF function. 

FXO: support analog subscribe line access, realize analog on hook 

and off hook, ringing current detection, CID transmission, Line status 

detection, reversal polarity detection, lifeline etc. 

Lifeline:Connect analog subscribe line, provide lifeline function. 

One uplink Ethernet interface, RJ-45, 10/100Base-T, adaptive. 

One downlink Ethernet interface, RJ-45, 10/100Base-T, adaptive, 

provide mirroring function for uplink port. 

G.711A/G.711U/G.729/G.7231/G.726/iLBC 

VAD/ Mute compression 

CNG 

PCM gain adjust 

Echo cancellation:G.168,128ms 

T.38 

T.30 

H.323:ITU-T H.323 

SIP:RFC3261 

Support both register and P2P 

Support the 

between primary and backup proxy 

DNS SRV:RFC2782 

NAT 

Caller ID 

Call hold(SIP) 

Call transfer(SIP) 

Call forward all(SIP) 

Call forward busy(SIP) 

Call forward no answer(SIP) 

Call forward selective(SIP) 

Caller ID hide(SIP) 

Anonymous call block(SIP) 

Hotline 

No disturb(SIP) 

Transparent transport 

RFC2833/RFC4733 

Signal (SIP INFO/DTMF over H.245) 

Call process voice China, Taiwan, Japan, Korea, USA, Germany, India, Russia, France, 

England, Italy, Brazil, Spain 

12

Self-defined parameters 

Parameters auto-learning 

CID 方式方式 

FSK(Bellcore Type 1) CID、DTMF-based CID 

Reversal polarity 

Fault recovery 

Management /maintenance 

Send 

Detection 

Fall back for IP route failure 

Lifeline for power failure 


RS-232 serial interface: configure, maintenance and diagnosis 

telnet 

web 

1.4 Product Description 

Unified network management 

MG3000-R32 provides 3 types of concurrent calls: 32, 16, and 8;4 types of sub-cards: 

8FXS、8FXO、4FXS4FXO、4FXS4Lifeline, and supports mixed insertion of different types 

of cards. 

1.4.1 Only FXS sub-card 

If only has 8FXS sub-cards, MG3000-R32 can be configured into 8-32 FXS ports. 

1.4.2 Only 8FXO sub-card 

If only has 8FXO sub-cards, MG3000-R32 can be configured into 8-32FXO ports. 

Table 6 ONLY 8FXO SUB-CARDS FUNCTION DESCRIPTION 

FXO function Support or not Description 

Normal mode √ 

Auto mode √ 

Dual mode X Only when all sub-cards are 4S4O or 4S4L, can 

the product support dual mode 

PSTN route X Only provided to FXS 

Fall back for IP route failure X Only provided to FXS 

Switch route by dialing special 

key 

FXO as remote IP access 

interface(One-stage mode) 


interface(Two-stage mode) 

FXO reversal polarity detection √ 

FXO CID signal detection √ 

FXO busy tone detection √ 

FXO external line detection √ 

FXO inbound authentication √ 

X Only provided to FXS 

√ Poll one free port from all FXO ports with same 

number to accept a call 

√ Poll one free port from all FXO ports with same 

number to accept a call 

13


Lifeline for power failure X Only 4S4O and 4S4L sub-cards support 

Note: the polling method refers to 6.13, the same below. 

1.4.3 Only 4S4O sub-card 

When product only has 4S4O sub-card, it can be configured into dual mode or normal 

mode or auto mode 

Table 7 ONLY 4S4O SUB-CARD FUNCTION DESCRIPTION 

FXO Function Support or not Description 


Auto mode √ 

Dual mode √ 

PSTN route √ In normal mode and auto mode, poll in complete 

system and find one free FXO port to call out; 

In dual mode, call out through fixed FXO port in 

one sub-card: 1-5、2-6、3-7、4-8 

Fall back for IP failure √ Under normal mode and auto mode, poll a free 

Switch route by dialing 

special key 





FXO reversal polarity 

detection 





FXO port to call out; 

Under dual mode, use one FXO port of the fixed 

subcard: 1-5, 2-6, 3-7, 4-8. 

√ Default is IP route, switch to PSTN by press 

special key; 

Default is PSTN route, switch to IP by pressing 

special key; ( take effect in 4S4O dual mode) 

√ Support in normal mode and auto mode; 

No t support in dual mode 

√ Poll all FXO ports with same number to find one 

√ 

free port to accept call; 

No t support in dual mode 

Lifeline for power failure √ FXS port use the fixed FXO port on the same 

1.4.4 Only 4S4L sub-card 

sub-card : 1-5、2-6、3-7、4-8 

4S4L sub-card has no FXO port, does not support mixed insertion, and only can be 

configured into dual mode. 

Table 8 ONLY 4S4L SUBCARD FUNCTION DESCRIPTION 

14


Normal mode X 

Auto mode X 

Dual mode √ 


PSTN route √ Switch to PSTN route by pressing special key, and 

Fall back for IP route failure X 


special key 





Lifeline reversal polarity 

signal detection 

dial PSTN number 

√ Only support “default is IP route, switch to PSTN by 

X 

X 

dialing special key” 

√ Actually L port transfer reversal signal to FXS port 

transparently 

Lifeline CID signal detection √ Actually L port transfer CID signal to FXS 

Lifeline for power failure √ 

transparently 

1.4.5 Mixed insertion of 8FXS and 8FXO 

Mixed insertion means, for 8FXS and 8FXO sub-cards, they can be used together, and 

any quantity or slot is ok. 

Table 9 MIXED INSERTION OF 8FXS AND 8FXO FUNCTION DESCRIPTION 



Auto mode √ 

Dual mode X Support only when subcards are all 4S4O or all 

4S4L 

PSTN route √ Poll all system to find one free FXO to call out 

Fall back for IP route failure √ Poll all system to find one free FXO to call out 


special key 





FXO reversal polarity signal 

detection 





√ Only support default is IP route, switch to PSTN by 

dialing special key. 

√ Poll all FXO ports with same number to choose one 

free port to accept the call 


√ 


15


Lifeline for power failure X Only 4S4O and 4S4L sub-cards support 

1.4.6 Mixed insertion of 8FXS and 4S4O 

Mixed insertion means, for 8FXS and 4S4O sub-cards, they can be used together, and 


Table 10 MIXED INSERTION OF 8FXS & 4S4O FUNCTION DESCRIPTION 



Auto mode √ 

Dual mode X Support only when subcards are all 4S4O or all 

4S4L 


Fall back for IP route failure √ Poll all system to find one free FXO to call out 


special key 





FXO reversal polarity signal 

detection 





√ Only support default is IP route, switch to PSTN by 

dialing special key.” 




√ 


Lifeline from power failure √ Only 4S4O sub-card support, FXS port of 8 FXS 

1.4.7 Mixed insertion of 8FXO and 4S4O 

sub-card can not support 

Mixed insertion means, for 8FXO and 4S4O sub-cards, they can be used together, and 


Table 11 MIXED INSERTION OF 8FXO & 4S4O FUNCTION DESCRIPTION 



Auto mode √ 

Dual mode X Support only when subcards are all 4S4O or all 4S4L 


Fall back for IP route 

failure 

Switch route by 

dialing special key 

√ Poll all system to find one free FXO to call out 

√ Only support “default is IP route, switch to PSTN by 

dialing special key.” 

16

FXO as remote IP 

access interface 

(One-stage mode) 

FXO as remote IP 

access interface 

(Two-stage mode) 

FXO reversal polarity 

signal detection 

FXO CID signal 

detection 

FXO busy tone 

detection 

FXO external line 

detection 

FXO inbound 

authentication 

Lifeline for power 

failure 





√ 

√ 

√ 

√ 

√ 


√ Only 4S4O can support 

17

2 Installation steps 

2.1 Installation notes 


In this section, some important issues will be described in details, including: MG3000-R32 

accessories, notes before installing the product, and environmental requirements. 

Notes: 

� This section must be read carefully before installing MG3000-R32 

� Only specially trained personnel can replace the internal cards in 

MG3000-R32 

2.1.1 Safety recommendations 

During the installation process, for the safety of operator as well as MG3000-R32 

equipment, please implement the following principles: 

� Cut off 220V AC or 48V power supply in electrical installation; 

� Read all notes and warning labels on package and equipment before you open 

any component; 

� Inspect the ground wire connection is right or not, and operate if the connection 

is correct. (Inspection steps: cut off power supply, connect equipment power line, 

and test the resistance between cabinet protection ground and shell of the 

product by multimeter. If the result is 0 ohm, the ground wire has been connected 

to the product. If the testing condition is not ready, please guarantee the power 

socket has three-input, not 2 line in the use of 220V AC); 

� Pay attention to electrostatic protection: as air is dry, human body often carries 

static electricity, which will cause the breakdown of semiconductor devices. So 

anti-static bracelets must be worn during installation. Touch grounding objects 

(such as heating etc.) is also an effective way to eliminate static electricity. In 

order to keep the safety of the personnel, the resistance of anti-static bracelets 

which connect hand must be tested, and the result should be in the range of 

1M-10M Ohms. 

2.1.2 Limitation of signal transmission distance 

Signal transmission distance is determined by signal types, transmission media as well as 

signal rate. Therefore the limitation of the transmission distance should be considered 

during product connection. 

In the distance limitation this section mentioned, signal can be transmitted well. If the 

18

distance exceeds the limitation, the signal quality may decline. 


The following are different transmission distance limitations, please refer to them when 

you are wiring. 

� 100Base:category 5, super category 5 or category 6 UTP/STP, maximum 

transmission distance is 100 meters; 

� According to IEEE standard, maximum transmission distance can reach 15m 

when serial line EIA/TIA-232 signal baud rate is 115200 bps; 

� Maximum transmission distance of subscriber line is 3000 meters 

2.1.3 Environment requirement 

Under normal circumstances, MG3000-R32 shall run in dry, clean, well-ventilated 

environment. 

While in long term, the environmental requirements are as follows: 

Table 12 MG3000-R32 ENVIRONMENTAL REQUIREMENT 

Requirements Minimum Maximum 

Running environmental temperature -10℃ 55℃ 

Humidity 10% 90% 

Power(220VAC input) 100VAC 240VAC 

Power (48VDC input) 36VDC 72VDC 

Storage temperature -25℃ 75℃ 

Storage humidity 5% 95% 

2.1.4 Correct wiring solution 

Good wiring is very important for the normal running and safety of integrated access 

equipment. When you install a new system, you should consider many factors such as 

wiring solution basing on signal transmission distance limitations, electromagnetic 

interference, as well as connectors compatibility etc. Reduce distortions, pressure or 

mutual entanglement is the basic requirement for wiring. 

2.2 Parts list 

The following table shows the MG3000-R32 parts list, and the actual packing list please 

refers to the factory list. 

19

Table 13 MG3000-R32 PARTS LIST 


Item Part description Specification Units Quantity Configuration 

1 MG3000-R32 Set 1 

2 Power line 3 lines,1.8 m,GB 

3 Power line 

4 Power line 

3 lines,1.8 米,Euro 

standard 

-48VDC , 1.5 m , 

PHOENIX plug 

pics 

Choose 

1 of 3 

5 Console cable RJ45-DB9,3 米 Pic 1 

6 Ethernet cable RJ45,1.5 米 Pic 1 

7 FXS subscribe line RJ45-4*RJ11 

2.5 米 

8 FXO subscriber line 

9 User manual 

RJ45- bare wire 

2.5 m 

Include cd, certification, 

warranty card 

Choose according to the 

power supply. If dual 

-48VDC power supply 

are needed, choose 2 

pics -48VDC power 

lines. 

Pic Sum of One pic every 4 FXS 

Pic 

sub 

Set 1 

cards ×2 One pic every 4 FXO 

10 Frame setscrew 4 pics in 1 set Set 0~1 Optional 

2.3 Install preparation 

According to table 13, please check whether the necessary environmental requirement 

and parts are ready. If there are no console cable and subscriber lines on the site, you can 

make it referring to the map below: 

For console cable, one end is DB-9 connector; the other end is RJ-45 connector: 

Figure 5 Console cable diagram 

For subscriber line, one end is RJ-45 connector, the other end are 4 RJ-11 connectors or 

bared wires. Middle cable are made by 4 pairs user lines (network cable): RJ45’s 1 pin 

and 2 pin correspond to the first telephone port ( RJ11 ); 3 pin and 4 pin correspond to the 

second telephone port; 5 pin and 6 pin correspond to the third telephone port ( RJ11 ); 7 

20

pin and 8 pin correspond to the fourth telephone port ( RJ11 ) 

2.4 Installation approach 

2.4.1 Installation 

Figure 6 FXS subscriber line diagram 

Figure 7 FXO subscriber line diagram 


MG3000-R32 equipment is 1U high, and can be put on the table, or in standard19 inch 

rack fixed by enamel. 

21

2.4.2 Wiring 

1) Subscriber line 


� Plug subscriber line’s RJ-45 connector into MG3000-R32 FXS port, and connect 

another end to normal telephone. 

2) Central office cable 

� Plug subscriber’s line’s RJ-45 into MG3000-R32 FXO port, and connect another 

end to digital switch or PBX; 

� Connect RJ-45 of Ethernet cable to WAN port, and another end to network 

switch. 

3) Console cable 

� Connect RJ-45 of console cable to the serial port of product, and another DB-9 

4) Power line 

end to the serial port of computer. 

� MG3000-R32 ( 220VAC ) use three-phase power line in package; 

� MG3000-R32 (48VDC) use 48V power line in package, and please notice (facing 

Notice 

back panel) that when you connect to power, they are -48VDC, RTN, and GND 

from left to right. 

2.4.3 Power on 

� Try to avoid hot plug or unplug the console cable, otherwise it may 

cause serial port of computer burnout! 

Before power on, please check again according to installation instructions and steps to 

ensure accuracy. And switch 220V or 48V power on. 

22

3 Equipment management description 

3.1 Keyboard inquiry 


MG3000-R32 provides an auxiliary management method for user: user can set some 

important parameters of the system through telephone keyboard. When you forget the IP 

address of the product, or when it is inconvenient to use web or command line to inquiry 

port number, you can get that information through dialing the keyboard. Please find 

specific description in Chapter 4. 

3.2 WEB 

Web is the main product management approach for MG3000-R32. You can access 

product, inquire parameters, and configure through IE browser. For more detailed 

instructions, please refer to Chapter 5. 

3.3 Command line 

Figure 8 Serial port parameters configuration 

User can also manage the product through command line. Command line includes 2 ways: 

serial console, telnet console. 

Serial port parameters please refer to Figure 8,baud rate: 115200。 

telnet port:100。 

Serial console and telnet console have the same username/password as Web. 

3.4 Unified network management 

MG3000-R32 can cooperate with the unified network management platform of our 

company. User can unify the operation, management, upgrade, batch configure, output 

trace etc. Please refer to the manual of our unified network management. 

23

4 Instructions for keyboard inquiry 


For your convenience, MG3000-R32 provides the function of inquiring the parameters and 

port number through keyboard. 

For the product with FXS port, you can inquire through the telephone directly connected to 

FXS port. 

For all FXO port products, you can connect FXO to PSTN line, and use another PSTN 

telephone to dial the PSTN number which connects to FXO port. Then you can do the 

inquiry. (Only in normal mode) 

Keyboard operation instruction: 

� Inquire network parameters: off hook, dial “ ***# ”, then you will hear IP address, 

subnet mask, gateway address, end with busy tone to prompt user to hang up; 

� Inquire port number: off hook and dial “**4# “, you will hear the port number, end 

with busy tone to prompt user to hang up. 

To broadcast network parameters and port number, you need download audio files first. 

Normally, the factory settings should include Chinese voice packet. 

Notice 

� If need English voice packet, please contact technical engineer of our 

company. 

24

5 Web configure description 

Figure 9 Web page framework 

MG3000-R32’s web page consists of 3 parts: 


� Title block: at the top of web page; the left is the Logo of manufacturer; in the 

center is the picture for product name; If it is neutral version, manufacturer Logo 

and the product name picture does not show here; 

� Menu bar:Locate in the left side of web page; list the function menu of the 

system, which includes fast configuration, basic configuration, advanced 

configuration, number transform, route, inquiry, device configuration. And each 

function menu still consists of many configuration pages. 

� Main interface: at the right of web page, is the major display and operate area. 

25

5.1 Log in 

Figure 10 Web Login interface 


Input device’s IP address in browser address bar, such as http://192.216.224.167, user 

will be prompted to log in. 

MG3000-R32 factory default IP address is 192.168.9.237. 

Notice � If you forget the device’s Ip address, you can dial ***# to get it. 

Recommend to use Microsoft IE 6.0, resolution of 1024*768. 

System default user: username: admin (password: admin). 

Notice 

5.2 Quicksetup 

5.2.1 H.323 protocol 

� User should keep the login password safely. If you forget the 

password which you have modified, please press the reset button for 

a long time to restore the factory setting. 

26

Figure 11 H.323 QuickSetup Page 


H.323 protocol QuickSetup page provides some key parameters of the system, which 

includes Network Configuration, H.323 protocol configuration, Encryption Configuration, 

DSP Configuration, Fax Configuration, DTMF Configuration, and Phone Number and 

Route etc... 

After user finishes the QuickSetup configuration, basic H.323 service can be realized. 

Detailed description please refers to Table 14. 

Table 14 H.323 QUICKSETUP CONFIGURATION DESCRIPTION 

27


Parameters Description Default 

Network Configuration 

IP Mode - static IP: input IP address, NetMask, gateway address 

- DHCP: obtain dynamic IP address through DHCP 

IP Address Set system static IP address. The format is 

XXX.XXX.XXX.XXX. Such as 192.168.9.237 

NetMask Set system static NetMask corresponding to IP address and 

the format is XXX.XXX.XXX.XXX. Such as: 255.255.255.0 

Default Gateway Set system Gateway address. The format is 

XXX.XXX.XXX.XXX. Such as: 192.168.9.1 

Primary DNS Server Set IP address of primary DNS server. The format is 

Secondary DNS 

Server 

Protocol Configuration 

Protocol Type Set protocol type: 

Working with 

Gatekeeper 

XXX.XXX.XXX.XXX, such as: 202.106.0.20 

Set IP address of secondary DNS server. The format is 

XXX.XXX.XXX.XXX, such as 0.0.0.0 

- H.323 

- SIP 

Set H.323 working mode: 

- Not Register 

- Register with IP 

- Register with Domain Name 

Master Gatekeeper IP If select Register with IP, here should input the IP address of 

Master Gatekeeper 

Domain 


RAS port 

master gatekeeper 

If select Register with IP, here should input the domain name 

of master gatekeeper 

Configure master gatekeeper RAS port, the range is 

0~65535 

Register Type Configure H.323 register type: 

- Register with gateway: corresponding “Alias Coding Mode” 

is “E.164” or “E.164 + Device H323ID” 

- Register with endpoint: corresponding “Alias Coding Mode” 

is “E.164” or “E.164 + Port H323ID” 

Alias Coding Mode There are 3 modes: 

- E.164 

- E.164 + Device H323ID 

- E.164 + Port H323ID 

Device H.323 ID Set Device H.323 ID If “register Type “ is “register with 

Encryption Configuration 

gateway “ and the “Alias Coding Mode” is “E.164 + Device 

H323ID” 

Enable Encryption Please select according to the server or the other side 

product configuration. If select Yes, please configure the 

encryption algorithm and key 

Static IP 

192.216.225.184 

255.255.255.0 

192.216.225.1 

192.216.224.4 

0.0.0.0 

H.323 协议 

Not Register 

192.168.9.88 

1719 

Register with 

endpoint 

E.164 

v237 

No 

28

Encryption Algorithm - RC4 

- NET2COM 


Encryption Key Set the key voip 

DSP Configuration 

First Coder Set preferred voice codec: 

- G.711U 

- G.711A 

- G.723.1 

- G.729 

- G.729A 

- G.729B 

- G.729AB 

- G.726_16 

- G.726_24 

- G.726_32 

- G.726_40 

- iLBC_13 

- iLBC_15 

Other voice coder Other coders which can be supported at the same time. 

Fax Configuration 

Fax Trans Mode Fax Configuration: 

DTMF Configuration 

- T.38 

- pass through ( should adopt G.711) 

DTMF Trans Mode Set DTMF of secondary dial transmission mode: 

Phone Number and Route 

Dial Special Number to 

Change route 

The Number to 

Change Route 

Default Route (only for 

dual mode) 

IP Routing Fallback to 

PSTN 

- Pass through 

- RFC2833 

- signal 

For SIP protocol, signaling is realized through SIP INFO. 

No matter which DTMF transmission mode is choosed, you 

had better enable echo cancel ( refer to section 5.3.3) 

- Disable 

- Enable 

If enable, you can change route by press special number 

Set the number, the range is 0~9, *, # # 

Set default route: 

- Default IP, if dial special number to PSTN 

- Default PSTN, Dial special number to IP ( only in dual 

mode) Please refer to section 6.8.3 

- Dsiable 

- Enable : if IP side fails, the call will choose FXO port to 

PSTN. For detailed description please refer to section 6.8.2 

RC4 

G.711U 

T.38 

Pass through 

Disable 

Default IP, if dial 

Special number 

to PSTN 

Enable 

29

Fallback within IP Net - Disable 

Local Phone Number 

Table 

Peer to Peer Call 

Routing Table 

Tips 

- Enable : If IP call fails, the call will try to choose another 

route within IP network first. For detailed information, please 

refer to 6.8.2. 

Open port configuration page. 

Open peer to peer calling route tale 


Disable 

� The parameters which are marked by “*” need reboot the system to 

take effect; 

� DSP parameters which are configured in Quicksetup page is valid for 

all ports; 

5.2.2 SIP Protocol 

30

Figure 12 SIP protocol quicksetup page 


31


SIP protocol quicksetup page provide key parameters of the system, including: network 

configuration, SIP protocol parameters, Encryption configuration, DSP configuration, Fax 

configuration, DTMF configuration, Phone Number and Route etc. 

When user finishes setting quick setup page, basic SIP service should be able to work. 

Detailed description please refers to table15. 

Table 15 SIP QUICK SETUP PAGE PARAMETERS DESCRIPTION 

Parameter Description Default 

Network configuration 

IP Mode - static IP: input IP address, NetMask, gateway address 

- DHCP: obtain dynamic IP address through DHCP 

IP Address Set system static IP address. The format is 

XXX.XXX.XXX.XXX. Such as 192.168.9.237 

NetMask Set system static NetMask corresponding to IP address and 

the format is XXX.XXX.XXX.XXX. Such as: 255.255.255.0 

Default Gateway Set system Gateway address. The format is 

XXX.XXX.XXX.XXX. Such as: 192.168.9.1 

Primary DNS Server Set IP address of primary DNS server. The format is 

Secondary DNS 

Server 

Protocol Configuration 

Protocol Type Set protocol type: 

XXX.XXX.XXX.XXX, such as: 202.106.0.20 

Set IP address of secondary DNS server. The format is 

XXX.XXX.XXX.XXX, such as 0.0.0.0 

- H.323 

- SIP 

Enable Registration Yes or no No 

Local Host Name Set local host name. If choose domain register, please input 

SIP server name here. 

Static IP 

192.216.225.184 

255.255.255.0 

192.216.225.1 

192.216.224.4 

0.0.0.0 

SIP protocol 

Registrar Address Set SIP register server IP address or domain name 0.0.0.0 

Registrar Receiving 

Port 

Set signal receiving port of SIP server, and the range is 

0~32767 

Register Period Set register period for registering. The range is: 20 ~ 7200s 60 

5060 

32

Register Type Select register type: 

- Register with gateway: Server will assign one gateway 

username and password to the whole equipment. During 

registering, only one register request need be sent which 

includes gateway username and password. If registered 

successfully, all ports are registered. But each port number 

also need be distributed by server, for calling authentication 

- Register with port: Server will assign a username and 

password to each port. And during registering, each port will 

send a register request which includes the username and 

password of the corresponding port. 

Gateway username Set gateway username if choose register with gateway 

Gateway Password Set gateway password if choose register with gateway 


Enable Proxy Yes or No No 

Proxy Address Set SI proxy IP address or Domain name 0.0.0.0 

Proxy Receiving Port Set SIP server signal receiving port, and the range is 0 ~ 


32767 





- NET2COM 


DSP Configuration 

First Coder Set preferred voice coder: 

- G.711U 

- G.711A 

- G.723.1 

- G.729 

- G.729A 

- G.729B 

- G.729AB 

- G.726_16 

- G.726_24 

- G.726_32 

- G.726_40 

- iLBC_13 

- iLBC_15 

Other voice coder Other coders which can be supported at the same time. 


Fax Trans Mode Fax Configuration: 

- T.38 

Register with 

port 

5060 

No 

RC4 

G.711U 

T.38 

33


- pass through ( should adopt G.711) 

DTMF Trans Mode Set DTMF of secondary dial transmission mode: 

Phone Number and Route 

Dial Special Number to 

Change route 

The Number to 

Change Route 

Default Route (only for 

dual mode) 

IP Routing Fallback to 

PSTN 

- Pass through 

- RFC2833 

- signal 

For SIP protocol, signaling is realized through SIP INFO. 

No matter which DTMF transmission mode is choosed, you 

had better enable echo cancel ( refer to section 5.3.3) 

- Disable 

- Enable 

If enable, you can change route by press special number 


Set the number # 

Set default route: 

- Default IP, if dial special number to PSTN 

- Default PSTN, Dial special number to IP ( only in dual 

mode) Please refer to section 6.8.3 

- Dsiable 

Fallback within IP Net - Disable 

Local Phone Number 

Table 

Peer to Peer Call 

Routing Table 

Notice 

- Enable : if IP side fails, the call will choose FXO port to 

PSTN. For detailed description please refer to section 6.8.2 

- Enable : If IP call fails, the call will try to choose another 

route within IP network first. For detailed information, please 

refer to 6.8.2. 

Open port configuration page. 

Open peer to peer calling route tale 

Pass through 

Disable 

Default IP, if 

dial Special 

number to 

PSTN 

Enable 

Disable 

� In the page, there are some parameters which are marked by *, such 

as “network configuration”, “protocol Configuration”, “Encryption 

Configuration”, these parameters modification only take effect after 

rebooting the system. 

� In quicksetup page, the config of DSP, Fax, DTMF will be valid for all 

ports. 

� In “other voice coder”, if the items choosed is the same as “first 

5.3 Basic Configure 

5.3.1 Protocol 

coder”, they will not be displayed after submission. 

34

5.3.1.1 H.323 Protocol 

Figure 13 H.323 Configurations Diagram 

For detailed configuration, please refer to Table 16. 

Table 16 H.323 PARAMETERS DESCRIPTION 



System Configuration 

H.323 & RTP Port 

From 

Configure initiative UDP port H.323 and RTP and the range 

is: 4000~65535. And you should avoid using the local signal 

port in use 

H.323 and Port to Configure end UDP port H.323 and RTP and the range is: 

4000~65535. And you should avoid using the local signal 

port in use. 

5000 

6000 

35

H.323 Registration Configuration 

Working with 

Gatekeeper 

Set H.323 working modes: 

- Not Register 

- Register with IP 

- Register with Domain name 

Master Gatekeeper IP If choose Register with IP, you need configure master 


Domain 


RAS Port 

gatekeeper IP address 

If choose Register with Domain name, you need configure 

master gatekeeper domain name 

Set master gatekeeper RAS port, and the range is: 0 ~ 

65535 


Not Register 

192.168.9.88 

1719 

Slave Gatekeeper IP If choose Register with IP, set Slave Gatekeeper IP address 0.0.0.0 

Slave Gatekeeper 

Domain 

Slave Gatekeeper 

RAS Port 

If choose Register with Domain name, set slave gatekeeper 

domain name 

Set slave gatekeeper RAS port, and the range is: 0 ~ 65535 0 

Default Gatekeeper IP If choose register with IP, set the default gatekeeper IP 

Default Gatekeeper 

Domain 

Default Gatekeeper 

RAS Port 

address 

If choose Register with Domain name, set the default 

gatekeeper domain name 

Set default gatekeeper RAS port, and the range is 0 ~6 5535 0 

0.0.0.0 

Local RAS Port Set Local RAS Port, and the range is 0 ~ 65535 1721 

Register Status Display the registration status 

Connection Configuration 

Connecting Mode Set H.323 working modes: 

H.225 Call Signaling 

Port 

TCP Keep Alive 

Interval 

Encryption Type 

- Normal 

- Fast Start 

- H.245 Tunneling 

- Fast Start + H.245 Tunneling 

Set local H.225 Call Signaling port, and the range is: 0 ~ 

65535 

Normal 

1720 

Set TCP keep alive interval, and the range is: 1 ~ 7200s 7200 





- NET2COM 


Other Configuration 

Send Local Ringback 

Tone 

1) Yes: FastStart mode, no matter receive remote media or 

not, all play local ringback tone; 

No 

RC4 

No 

36

2) Yes: Normal mode, play local ringback tone; 

3) No: FastStart mode, if remote end send the media signal, 

play remote media; if remote end does not send media 

signal, caller can not hear voice; 

4) No: Normal Mode: caller can not hear voice 

Interdigit Timeout Set interdigit timeout if overtime, system will regard the user 

Tips 

has finished dialing. The range is: 1 ~ 20 s 


� Modification of the parameters with * need reboot the system to take 

effect 

5.3.1.2 SIP Protocol 

5 

37

Figure 14 SIP protocol parameters settings 


This page provides parameters related to SIP protocol, for detailed information please 

refers to Table 17. 

Table 17 SIP PARAMETERS DESCRIPTION 


System Configuration 

RTP Base Port Set RTP base port number. The range is: 4000~60000. 

Note: avoid using the same port number as local used signal 

port. 

SIP Registration Configuration 

6000 

38

Local Host Name Set local host name. If domain register type is selected, 

recommend to input SIP server domain name here. 


Local Port Set local port of SIP protocol. The range is 0 ~ 32767 5060 

Enable Registration Yes or No No 

Registrar Address Set SIP server domain name and IP address 0.0.0.0 

Registrar Receiving 

Port 

Set SIP server signal receiving port. The range is 0 ~ 32767 5060 

Register Period Set register period. The range is 20 ~ 7200s 60 

Register Type - Select register type: 

- Register with gateway: Server will assign one gateway 

username and password to the whole equipment. During 

registering, only one register request which includes 

gateway username and password need to be sent. If 

registered successfully, all ports are registered. But each 

port number also need be distributed by server, for calling 

authentication 

- Register with port: Server assign a username and 

password to each port. And during registering, each port will 

send a register request which includes the username and 

password of the corresponding port. 

Gateway Username Set gateway username if choose register with gateway 

Gateway Password Set gateway password if choose register with gateway 

Gateway Register 

Status 

Display current status: 

- Not registered 

- Registered 

Enable Proxy Yes or Not No 

Register 

with port 

Proxy Address Set SIP server IP address or domain name 0.0.0.0 

Proxy Receiving Port Set SIP server signal receiving port. The range is 0 ~ 32767 5060 

Backup Proxy Address Set SIP backup Proxy IP address. When fault to register in 

Backup Proxy 

Receiving Port 

Enable Proxy Keep 

Alive 

Proxy Keep Alive 

interval 


primary server, system will switch to backup proxy server 

automatically, and park in backup proxy server. 

Note: backup proxy server does not support domain register 

Set SIP backup proxy server signal receiving port. The range 

is 0 ~ 32767 

Yes or No. Mainly apply in NAT No 

Set interval. Heartbeat message is OPTIONS. The range is 

20 ~ 7200s 





- NET2COM 

0.0.0.0 

5060 

60 

No 

RC4 

39



Other Configuration 

Send Local Ringback 

Tone 

Enable RFC2833 

Negotiation 

FXO Provisional 

Response Type 

Yes or No. 

1) Yes: receive 18X with SDP, play local ringback tone; 

2) Yes: receive 18X without SDP, play local ringback tone; 

3) No: receive 18X with SDP, play remote media; 

4) No: receive 18X without SDP, play local ringback tone 

Note: you must choose “No” if you want to realize 183 SDP 

( IP-> Termination gateway ->PSTN) and only receiving 180 

(IP - IP) at the same time. 

Yes or No. Set the DTMF secondary dialing transmission 

negotiate with RFC 2833 or not. 

Set the response type to 180 or 183, when FXO receives the 

call as callee in IP side. 

In IP-To-PSTN call, under One-stage mode, FXO reply183 

SDP message to realize CRBT transport transparently, 

which will transport CRBT or prompt tone of PSTN side to 

the caller of IP side. Normally, CRBT transport transparently 

function should cooperate with answer supervision function, 

which means PSTN shall provide reversal polarity signal, 

“FXO Answer Supervision” parameter is set to "after 

receiving reversal signal“ 

Function description please refer to section 6.3.1.1 

Interdigit Timeout Set the time value. If exceeded the value, system will 

Notice 

5.3.2 Port 

5.3.2.1 H.323 协议 

consider that user has finished dialing. Value range is: 1 ~ 

20s 

� the modification of parameters marked by *, such as “Local Port”, 

“Enable Encryption”, only take effect after “save” and “reoot”. 

� “Send local ringback tone” of SIP protocol in the old version, may 

have differences compared with old version, which will not be pinted 

out here one by one. 

No 

No 

180 

5 

40

Figure 15 H.323 Port Configuration Diagram 


In H.323 protocol, this page is used to configure all ports phone number, H.323 ID. 

If the “Registration Type” is set as “Register with Endpoint”, and the “Alias coding Mode” is 

“E.164 + Port H323ID”, you should configure H.323 ID parameters in “Port Configuration” 

page. 

During configuration, please input related parameters in corresponding port and click 

“submit”. 

Port configuration support batch configuration, for example: 

� Please input the range in the “Range” of the first port such as 1-8, which means 

batch configure 1-8 ports; 

� Please input port H.323 ID in the “H.323 ID” of the first port, such as 1678017; 

� Please input the phone number in the “Phone Number” of the first Port, such as 

1678017; 

Click “Submit”, and system will do the batch configuration for 1-8 ports, and the phone 

number increases, H.323 ID is the same with the first one. 

Notice � The modification of “H.323 ID” parameters nee reboot to take effect 

5.3.2.2 SIP Protocol 

41

Figure 16 SIP Protocol Port Settings 


In SIP protocol, port configure page is used to set PhoneNbr, AuthUsr, AuthPwd, and 

display the port RegStatus. 

For settings, you just need input parameter in the corresponding box and click “submit”. 

It support batch configure, for example: If you want to configure all the ports, 

� Input 1-8 in the range box of first port, which means you want to configure 1-8 

ports; 

� Input PhoneNbr of the first port; 

� Input AuthUsr; 

� Input AutPwd. 

Then click “Submit”. The system will configure 1-8 ports configurations automatically, in 

which, the PhoneNbr will increase, and the AuthPwd is the same as the first port. 

If you want to delete all the settings, just click “Delete All”, and system will popup a confirm 

box. Choose “yes” to confirm and “no” to cancel. 

5.3.3 Channel 

42

Figure 17 Channel settings 


This page is used to configure parameters about channel, including voice, fax, DTMF etc. 

For details, please refer to Table 18. 

43

Table 18 CHANNEL CONFIGURATIOIN DESCRIPTION 



Card Selection 

Card Number Choose the card number Card 1 

Port Selection 

Port Number Choose the port you want or all ports of the sub-card 1 

Voice Configuration 

Pactetization Interval Set packetization interval, the range is ( 5,10,20...200ms): 

- G.711U:5, 10, 20, 30, 40, 50, 60 

- G.711A:5, 10, 20, 30, 40, 50, 60 

- G.723.1:30, 60, 90 

- G.729A:10, 20, 30, 40, 50, 60, 70, 80 

- G.726_16:5, 10, 20, 30, 40, 50, 60 

- G.726_24:5, 10, 20, 30, 40, 50, 60 

- G.726_32:5, 10, 20, 30, 40, 50, 60 

- G.726_40:5, 10, 20, 30, 40, 50, 60 

- iLBC_15:20 

- iLBC_13:30 

Note: the smaller the period is, the better is the real time 

performance of the voice transmission. But when the packet 

becomes more, the burden of the network will be heavier; if 

the period is larger, packets need sent will become less, and 

the packet will become big; 

For the RTP packet received, as long as the interval is within 

the range above, MG3000-R32 can handle them. 

Packet to PCM Gain Set the packet to PCM gain, and the range is -14~6dB, in 1 

dB increment 

PCM to Packet Gain Set the PCM to packet gain, and the range is -14~6dB, in 

1dB increment 

Comfort Noise Set whether send comfort noise to PCM side when receive 

G.711 Redundant 

Depth 

mute from IP side: 

- Disable 

- Enable 

Set RTP redundant depth when using G.711. This function 

can prevent the voice from in continuous due to the packets 

loss. The options are: 

- Disable 

- 1 

- 2 

- 3 

Echo Cancel - Disable 

- Enable 

Note: In the call when one part hears the echo, you should 

20 

0.0 

0.0 

Disable 

Disable 

Enable 

44

Echo Cancel Tail 

Length 

Voice Activity 

Detection 

enable this function in the other opposite end. 

Set the echo cancel tail length, and if within the set range, 

the echo can be cancelled. The option is: 

- 8ms 

- 48ms 

- 128ms 

Voice Activity Detection activation function may implement a 

quiet voice signal compression basing on the detection 

threshold, reducing the use of network bandwidth. 

Option: 

- Disable 

- Enable. If choose enable, system can compress the mute 

voice according to the 

VAD Threshold Set VAD Threshold. The range is: 

- Maximum: most save bandwidth, but the voice quality is the 

worst 

- 5dB: relative big noise ( 5dB SNR) detected to be voice 

- 10dB: big noise ( 10dB SNR) detected to be voice 

- 20dB:relative small noise ( 20dB SNR ) detected to be 

voice 

- 30dB:occupy bandwidth most, but the voice quality is the 

best 

For the options above, the bandwidth save situation: 

Maximum>5dB>10dB>20dB>30dB; 

Voice quality: 

Maximum

Note: for the coder type, such as G.726_16, G.726_24, 

G.726_40, ILBC_13, iLBC_15, DTMF2833 etc., you should 

avoid setting them in the same value. 

G.726_24 PT Value Set the payload type if using G.726_24 coder. The value 

range is: 96 ~ 127. 

G.726_40 PT Value Set the payload type if using G.726_40 coder. The value 

range is: 96 ~ 127. 

iLBC_13 PT Value Set the payload type if using iLBC_13 coder. The value 

range is: 96 ~ 127. 

iLBC_15 PT Value Set the payload type if using iLBC_15 coder. The value 


range is: 96 ~ 127. 

Fax Trans Mode Set the fax trans mode: 

- T.38 

- pass through ( if use this mode, you should choose G.711 

voice coder) 

T.38 Fax ECM Set T.38 ECM error correction function. 

T.38 Error Recover 

Method 

Redundance count for 

fax page data 

Redundancy count for 

T.30 Messages 

- Enable 

- Disable 

Set T.38 recovery mode: 

- Redundancy 

- FEC 

Set the redundance count for fax page data. 

When T.38 fax fails, you can withdraw redundant data to 

recover. The bigger the value is, the larger the data packet of 

fax will be (because it includes redundant information). The 

options are: 

- 0 

- 1 

- 2 

- 3 

Set T.30 redundancy count of T.30 message. 

When T.30 message transmission fails, you can get the 

redundancy message to recover. The larger the value is, the 

bigger message of T.30 will be. The options are: 

- 0 

- 1 

- 2 

- 3 

- 4 

- 5 

- 6 

- 7 


103 

104 

105 

106 

T.38 

Enable 

Redundancy 

TCF Trans Mode Set TCF (Training Check Field) mode. Before you send a Remote 

0 

3 

46

fax, there is a TCF, which can be used to confirm the 

situation of the line. The options are: 

- Remote 

- Local 

T.38 Fax Relay Volume Set T.38 fax gain. The range is -15 ~ 0dB, increasing by 1 dB 

T.38 Fax Relay Max 

Rate 

T.38 Packet Loss 

Concealment 


increment 

Set the max fax rate: 

- No speed limit 

- 2400bps 

- 4800bps 

- 7200bps 

- 9600bps 

- 12000bps 

- 14400bps 

Set the error concealment mechanism when T.38 fax 

packets are lost. 

- No 

- by white lines 

- by last good line 

DTMF Trans Mode Set the transmission mode of DTMF secondary dial tone: 

RFC2833 Payload 

Type 

- pass through 

- RFC2833 

- Signal 

RFC2833 Scheme - IETF 

In H.323 protocol, signal transmission is realized by H.245 

UserInput message ; 

In SIP protocol, signal transmission is realized by SIP 

INFOmessage. 

Whatever mode of DTMF signal you choosed , you had 

better activate “ echo cancel ” function. 

Set RFC2833 payload type for DTMF secondary dial. The 

value range is : 96~127 

- AAL2 

5.4 Advanced Configuration 

5.4.1 FXS 


-6 

14400bps 

No 

P ass 

through 

97 

AAL2 

47

Figure 18 FXS Parameters Settings 


FXS function page consist of 2 parts : FXS advanced Configuration and FXS 

Configuration. For detailed information, please refer to Table 19. 

Table 19 FXS PARAMETERS DESCRIPTION 


FXS advanced configuration 

Card Number Choose the sub-card Card 1 

Port Number Choose the port you want or all ports of the sub-card 1 

Enable Port - Yes 

- No 

Send Reversal Send reversal signal or not. 

- Yes 

- No 

If you choose yes, when the callee pick up the phone, a 

reversal signal will be sent to the caller. 

Send Caller ID Send caller ID or not. 

- Yes 

- No 

If you choose yes, caller id will be sent to the caller. 

CallerID Type Set in which way to send CID: 

- FSK MDMF 

- FSK SDMF 

- DTMF 

Yes 

Yes 

Yes 

FSK 

MDMF 

48

FXS Configuration 

Open Circuit DC Feed 

Voltage 

Battery Switching 

Offset Voltage 

Current Limit Active 

Mode 

Set the DC feed voltage of between Telephone interface TIP 

and Ring in open circuit. The range is 36 ~ 57 V. 

DC feed voltage between TIP and RING can be high, and 

also can be low. You can change the threshold of battery 

switching offset voltage by modifying this parameter. When 

the voltage between TIP and RING is less than the reference 

value ( Reference value = low voltage + offset + voltage due 

to chip’s resistance + 4 V ) and reach 2 ms, it will switch 

from high voltage to low voltage; on the contrary, if it arrives 

2ms higher than the reference value, it will switch from low 

voltage to high voltage. The range is 3 ~ 12 V 

Telephone interface (TIP and Ring) provide constant-current 

feed to the load. The range is 18 ~ 49mA. 

Ring Waveform Set Ring waveform of FXS port: 

Ring Voltage ( Peak 

Value) 

- Sinusoid 

- Trapezoid 


Set FXS ring voltage (peak value). The range is: 1 ~ 155V 90 

Ring Frequency Set the FXS ring frequency, whose range is: 1 ~ 3400Hz 25 

Ring Tone Cadency Set FXS ring on-off time sequence. 

The format is: COn1/COff1,COn2/COff2, COn3/COff3, 

COn4/COff4 

For each period, utmost 4 tones are allowed. If ring tone 

adopt 1s on, 4s off, the format will be : 1000/4000, 0/0, 0/0, 

0/0 

Hook flash time Set up FXS flash time limit. The range is: 100 ~ 1000ms. If 

On-Off time length is smaller than the set value, it will be 

treated as flash signal. 

FXS Port Impedance Set resistance of external equipment which connects to FXS 

port. You can solve the voice quality problem which caused 

by unmatched resistance through adjusting this parameter. 

- 600ohm 

- 600ohm + 1000nF 

- 600ohm + 2160nF 

- 900ohm 

- 900ohm + 1000nF 

- 900ohm + 2160nF 

- 120ohm + (820ohm || 110nF) 

- 180ohm + (630ohm || 60nF) 

- 200ohm + (680ohm || 100nF) 

- 200ohm + (1000ohm || 100nF) 

- 215ohm + (1000ohm || 137nF) 

- 220ohm + (820ohm || 115nF) 

48 

9 

18 

Sinusoid 

1000/4000, 

0/0,0/0, 0/0 

300 

200ohm + 

(680ohm || 

100nF) 

49

- 220ohm + (820ohm || 120nF) 

- 270ohm + (910ohm || 120nF) 

- 300ohm + (1000ohm || 220nF) 

- 370ohm + (620ohm || 310nF) 

- 400ohm + (500ohm || 50nF) 

FXS Port Gain Set FXS port input and output gain: 

5.4.2 FXO 

- AD = 0db DA = 0db 

- AD = 0db DA = -3.5db 

- AD = -3.5db DA = 0db 

- AD = -3.5db DA = -3.5db 

Figure 19 FXO Parameters settings 


AD = 0db 

DA = 0db 

50

Figure 20 Set Forward Number 


FXO page consists of 2 parts: FXO advanced configuration and FXO configuration. For 

details please refer to Table 20. 

Table 20 FXO FUNCTION DESCRIPTION 


FXO Advanced Configuration 

Working Mode Set FXO working mode: 

Normal Mode Play 

Record 

- Normal Mode 

- Auto Mode: if you choose this option, you need click 

Display Auto FWD NBR to configure the number. ( refer to 

Figure 20 Set Forward Number) 

- Dual Mode: only available when sub-cards are all 4S4O or 

4S4L 

Set whether play record or not: 

- No: when the call is from PSTN to FXO, play secondary dial 

tone. 

- Yes: when the call is from PSTN to FXO, play the welcome 

words which are pre-recorded 

Note: when the record is playing, the function is unavailable 

Enable PSTN CallerID Set to activate display PSTN CallerID function or not. No 

Ring Times Detected When call from PSTN to FXO, set the system to detect the 

ring times in FXO port, and the range is 1 ~ 5. 

In normal mode or Auto mode, if you enable PSTN 

callerIDdisplay function, you have to set the ring times 

detected bigger than one, in order to guarantee system has 

enough time to do the detection. 

While in Dual mode, this parameter does not affect the 

Normal 

Mode 

No 

1 

51

FXO Answer 

Supervision 

Regenerate DTMF 

Digit Relay 

Regenerate DTMF 

Digit Interval 

FXO Configuration 

Operational Loop 

Current Min 

TIP/RING Voltage 

Adjust 

Min Ring Frequency 

Detected 

Max Ring Frequency 

Detected 

CallerID display function. The detailed description please 

refers to section 6.6. 

When set One-stage mode, FXO billing signal transmissions: 

- Normal: As soon as receive INVITE/Setup signal, system 

give a 200 OK/Connect response at once. 

- After receiving reversal signal: when receive INVITE/Setup 

signal, system does not reply 200 OK/Connect immediately, 

but answer it after detecting reversal signal from PSTN side. 

For detailed information please refer to section6.3.1.1. 

Set the time delay from FXO get connected to starting to 

send number when regenerated number is sent from FXO to 

PSTN. The range is: 0 ~ 5000ms. 

Adjust this parameter, so that FXO may only start to send 

number after the switch of PSTN side is ready.(begin to send 

dial tone) 

Set the transmitting interval between DTMF numbers, when 

regenerate the number from FXO to PSTN. 

- 0ms 

- 200ms 

- 400ms 

- 600ms 

- 800ms 

- 1000ms 

For the switch of PSTN, which is not very sensitive to DTMF 

signal detection, you can improve interval of DTMF 

regeneration number through adjusting this parameter. 

Set the minimum loop current in which FXO can work 

normal. The range is: 

- 10mA 

- 12mA 

- 14mA 

- 16mA 

Working voltage of DAA pin of FXO chip, supplied by feed of 

TIP and Ring. The range is: 

- 3.1V 

- 3.2V 

- 3.35V 

- 3.5V 

Set the min ring frequency of measurable ring signal, which 

should be not less than 4 Hz 

Set the max ring frequency of measurable ring signal, and it 

must be equivalent or more than Min Ring Frequency. 


Normal 

FXO Port Impedance Set impedance of FXO external circuit. You can avoid the 200ohm + 

1500 

0 

10 

3.5 

10 

83 

52

voice quality problem which caused by unmatching 

impedance through adjusting this parameter. 

- 600ohm 

- 900ohm 

- 270ohm + (750ohm || 150nF) 

- 220ohm + (820ohm || 120nF) 

- 370ohm + (620ohm || 310nF) 

- 320ohm + (1050ohm || 230nF) 

- 370ohm + (820ohm || 110nF) 

- 275ohm + (780ohm || 115nF) 

- 120ohm + (820ohm || 110nF) 

- 350ohm + (1000ohm || 210nF) 

- 200ohm + (680ohm || 100nF) 

- 600ohm + 2.16uF 

- 900ohm + 1uF 

- 900ohm + 2.16uF 

- 600ohm + 1uF 

- global complex impedance 

On Hook Speed Set on hook speed of FXO port. 

- less than 0.5ms 

- 2.9~3.1ms(ETSI standard) 

- 25.9~26.1ms 

Ring Indication Delay Set the ring indication delay of FXO port. The range is 0 

~1792ms. 

Ringer Impedance Set the ringer impedance which matches external ringer. 

- Maximum ( high ) ringer impedance 

- synthesized ringer impedance : mainly used in some 

special countries, such as Poland, South Africa, Slovenia 

etc. 

Last Ring Finish Time When system has detected a ringing signal, the interval 

between remaining ringing signal is called as ringing 

overtime, and it can be set to 128ms、256ms、...、1920ms 

Ring Threshold Set the voltage threshold of ringing detection: 

Current Limiting 

Enable 

- 13.5Vrms 16.5Vrms 

- 19.35Vrms 23.65Vrms 

- 140.5Vrms 49.5Vrms 

Set whether enable current limiting function of FXO port. 

- Yes 

- No 

Ring validation Time Set the minimum ringing signal time length to judge whether 

5.4.3 Voice 


it is a ringing signal. It can be set to 100ms、150 ms、200 ms、 

256 ms、384 ms、512 ms、640 ms、1024 ms 

(680ohm || 

100nF) 

Less than 

0.5ms 

0 

Maximum 

(high) 

ringer 

impedance 

512 

13.5Vrms 

16.5Vrms 

No 

100 

53

Figure 21 Voice Parameters settings 


For voice settings, it consists of Call Progress Tone Sending Configuration, Call Progress 

Tone Detecting Configuration. And the format of tone can be different for sending and 

detecting. 

The input box of self defined tone only pops up when you choose “other” in “Country and 

District” of “Call Progress Tone Sending Configuration”. 

And the parameters of self learned tone only pops up when you choose “Learning” in 

54

“Country and District” of “Call Progress Tone Detecting Configuration” 

Table 21 VOICE PARAMETERS DESCRIPTION 


Call Progress Tone Sending Configuration 

Country and 

District 

Choose the country which the sending tone belongs to: 

- China 

- Taiwan 

- Japan 

- Korea 

- USA 

- Germany 

- India 

- Russia 

- France 

- UK 

- Italy 

- Spain 

- Brazil 

- Other: selecting this, you can define the format of 

sending tone by yourself. 

Dial Tone When you select “other” in “Countyry and District”, you 

Ringback 

Tone 

can define the format of dial tone. And please pay 

attention to the format requirements. Please refer to 

Section 6.4.2. 

When you select “other” in “Countyry and District”, you 

can define the format of ringback tone. And please pay 



Busy Tone When you select “other” in “Countyry and District”, you 

Unobtainable 

Tone 

can define the format of busy tone. And please pay 




can define the format of unobtainable tone. And please 

pay attention to the format requirement. Please refer to 


Call Progress Tone Detection Configuration 

Country and 

District 

Choose the country which the tone needing be detected 

belongs to: 

- China 

- Taiwan 

- Japan 

- Korea 


China 

(450@-15+0@0,10000/0); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,1000/400 

0); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,350/350); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,100/100); 

(450@-15+0@0,100/100); 

(450@-15+0@0,100/100) 

中国 

55

Enable 

Busytone 

Learning 

- USA 

- Germany 

- India 

- Russia 

- France 

- UK 

- Italy 

- Spain 

- Brazil 

- Other: you can define the format of detecting tone. 

- Learning: the system can learn the format of detecting 

tone of the line. 

When you choose “Learning”, you can define whether to 

enable busytone learning or not. 

Learning Port Set the learning port. 

Learning 

Frequency 

Mode 

Dial Tone 拨 

号音 

Ringback 

Tone 

Set busytone learning frequency mode: 

- Using One Frequency 

- Using Two Frequency 


can define the format of dial tone. And please pay 




can define the format of ringback tone. And please pay 



Busy Tone When you select “other” in “Countyry and District”, you 

Unobtainable 

Tone 

Detection 

Type 

Noise 

Estimation 

Control 

can define the format of busy tone. And please pay 




can define the format of unobtainable tone. And please 

pay attention to the format requirements. Please refer to 


Set detection type: 

- One tone detecting use lower FFT 

- One or two tone detecting use higher FFT 

- Two tone detecting use lower FFT 

- Detect on band edges 

Set the parameters of noise estimation control: 

- After removing one tone 

- After removing two tones 

- After removing three tones 


No 

Using One Frequency 

(450@-15+0@0,10000/0); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,1000/400 

0); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,350/350); 

(0@0+0@0,0/0); 

(0@0+0@0,0/0) 

(450@-15+0@0,100/100); 

(450@-15+0@0,100/100); 

(450@-15+0@0,100/100) 

One or two tone detecting 

use higher FFT 

After removing two tones 

56

Cadence 

Cycles 

Detect Delay 

Time 

Drop Delay 

Time 

5.4.4 SPS 

Set the detecting cadence: 

- Verify one on time only 

- Verify one on –off cadence 

- Verify two on-off cadence 

- Verify three on-off cadence 

Set the delay time from detection signal start. The range 

is 10 ~ 500ms 

Set the delay time after the end of detecting signal. The 

range is 10 ~ 80ms 

Figure 22 SPS Parameters Settings 

This SPS page is used to configure supplementary service parameters. 


Verify one on-off cadence 

In this page, most of the supplementary service is SIP service. For H.323, it only supports 

hotline function. 

Table 22 SPS PARAMETERS DESCRIPTION 



300 

40 

57


Card Number Select one sub-card or all the cards to configure. Sub-card 1 


Port Number Select one port or all ports of sub-card to configure 1 

Enable / Disable 

Call Hold - On 

- Off 

Call Transfer - On 

Unconditional Call 

Forward 

If you choose “on”, either the caller or the callee, can press 

the hook or flash key to maintain or restore the call, 

- Off 

If you choose “on”, this port can realize call transfer function 

though pressing hook or flash key. 

Note: before using this function, the call hold function must 

be activated. 

- On 

- Off 

Busy Call Forward - On 

No Answer Call 

Forward 

If you choose “on”, regardless of the port status, all inbound 

calls to this port will be forwarded to the number which is set 

as the forwarder number. 

- Off 

If you choose “on”, when user of the port is busy, inbound 



- On 

- Off 

Selective Call Forward - On 

If you choose “on”, when the time of no answer exceeds the 

value set, inbound calls to this port will be forwarded to the 

number which is set as the forwarder number. 

- Off 

Caller ID Block - On 

If you choose “on”, when the port receives a call, and the 

caller ID accords with the conditions set for forward, inbound 



- Off 

Block Anonymous Call - On 

If you choose “on”, during the call, this port will conceal its 

caller ID. 

- Off 

If you choose “on”, if this port receives a call whose callerID 

is concealed, the call will be banned. 

Hotline - On off 

off 

off 

off 

off 

off 

off 

off 

off 

58

- Off 

Do Not Disturb - On 

If you choose “on”, when you pick up the phone and do not 

dial the number within the set time, the system will call the 

set hotline number automatically. 

- Off 

Call Forward Configuration 

Unconditional FWD 

NBR: 

If you choose “on”, this port will refuse all inbound calls. The 

caller will only hear the busy tone. But the outbound calls will 

not be restricted. 

Set unconditional FWD NBR to cooperate with 

"Unconditional Call Forward” functions. 

Busy Call FWD NBR Set busy call FWD NBR, to cooperate with "Busy Call 

No Answer Call FWD 

NBR 

No Answer Call FWD 

Timeout 

Forward” function. 

Set no answer call FWD NBR, to cooperate with "No Answer 

Call Forward” function. 

Selective Call Forward Configuration 

Set no answer call FWD Timeout value, to cooperate with 

“No Answer Call Forward” function. 

Caller NBR 1: Set the first caller number which is selected to forward 

FWD NBR 1 Set the forward number corresponding to the “Caller NBR 1” 

Caller NBR 2 Set the first caller number which is selected to forward 














Hotline Configuration 

Hotline Timeout Set the time value, after off-hook, exceed which will trigger 

the hotline function, cooperating with hotline function 

Hotline Number Set the hotline number to cooperate with hotline function 

5.4.5 Net Manage Setting 


off 

20 

0 

59

Figure 23 Net Manage Setting Diagram 


This page is used to configure parameters NMS, all of which obtain from the NMS server. 

Table 23 NMS PARAMETERS DESCRIPTION 


Register or not - Yes 

- No 

NMS Address Set NMS server IP address or Domain name 

NMS Port Set NMS server port number for registration, and the range 

is: 0 ~ 65535 

Register Status Display the register status of NMS server 

5.5 NBR Exchange 

5.5.1 Digit Map 

Figure 24 Digit Map setting Diagram 

No 

8800 

60


This page is used to set DigitMap parameters. When you are dialing a number, if it 

matches one dial plan rules successfully, the number will be sent to system for processing 

at once, and need not dial “#” or wait for overtime. 

Items of Digitmap should be written in the format of pattern matching rule (refer to Section 

6.5), supporting wildcard, and utmost 100 pieces. 

The Figure 24 has listed 3 ordinary dial plans: 

� 013XXXXXXXXX:Domestic long-distance phone number 

� [5-6;8]XXXXXXX:City fixed telephone number 

� 13XXXXXXXXX: City mobile number 

5.5.1.1 Add Dial Plan 

Figure 25 Adding Dial Plan Diagram 

In Figure 25, add one dial plan, and the number is 8001~8009: 

In the “Dial Plan" box, input 800Z, and then click “Add”. 

5.5.1.2 Modify Dial Plan 

If you want to modify one existing dial plan, please click the index number block, and the 

corresponding dial plan will appear in the box automatically. After you modify the “Dial 

Plan”, just click “Modify”. 

5.5.1.3 Delete one Dial Plan 

If you want to delete one existing dial plan, please click index number clock, and the 

corresponding dial plan will appear in the box automatically. Then click “delete”. 

5.5.1.4 Delete All 

If you want to delete all the dial plans, click “Delete All” directly, and system will pop up a 

confirm box. Click yes to confirm, then all the dial plans will be deleted. 

Notice � Please be cautious of “Delete All” operation 

61

5.5.1.5 Enable “#” as Number End Flag 

Figure 26 Set “#” as Number End Flag Diagram 


Besides dial plan, you can also enable/disable “#” as number end flag (Figure 26). If you 

disable this item, and system can not find matching dial plan rule, system will regard over 

time as number ending flag. 

Normally, this is set as Enable, and system will use “#”, dial plan, and over time as the 

number end flags at the same time. For some special situations, for example, if user want 

to send out”#57#”, which contains “#”, you need disable this item. At the same time, you 

should avoid using “#” in “Dial Special Number to Change route” function. 

5.5.2 SpeedDial 

Figure 27 Speed Dialing Diagram 

This page is used to set abbreviated number item, and utmost can set 50 pieces. 

After you set one abbreviated number for one telephone number, you only need dial 

abbreviated number when you want to dial that telephone number. For example, in Figure 

27, there are three speed dial items: 

� User dial 1, equal to dialing 8001; 



62

Tip 


� If you want to start the call immediately after you dial the abbreviated 

5.5.2.1 Add speed dialing rule 

number, you should set a cooresponding dial pan. For example, for 

the first speed dial item, you should set a dial plan to “1”. 

Figure 28 Add speed dial rule Diagram 

In Figure 28, add a speed dial rule: input 01391091188 in original number box, input 4 in 

abbreviated Number box, then click “Add” 

5.5.2.2 Modify Speed dialing rule 

If you want to modify an existing speed dialing rule, click index number box, and this rule 

will appear in the operation box automatically. Then modify the number in abbreviated 

number box, or original number box, and click “Modify”. 

5.5.2.3 Delete a speed dialing rule 

If you want to delete one existing speed dialing rule, click index number box, and the rule 

will appear in the operation box automatically. Then click “Delete”. 


If you want to delete all speed dialing rules, just click “Delete All”, and system will pops up 

a confirm box, after you confirm it, all the rules will be deleted. 

Notice � Please be cautious to use “Delete All” operation. 

5.5.3 Change Rule 

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Figure 29 Number Change Rule Table 


This page is used to configure all kinds of number change rules, and can support utmost 

100pieces. 

After you define the number change rule, the rule can be cited by “Callout Number 

Change Table”, “Caller Number Change Rule Index”, “Fall Back Number Change Rule 

Index” “Regeneration Number Change Table” 

Number Change Rule includes: Addprefix, Delprefix, Addsuffix, Delsuffix, Insertnbr, 

DelNbr, KeepNbr, and ReplaceNbr. For detailed information please refer to Section 6.7. 

Figure 29 cited several examples of number change rules: 

� Rule 1: Change Type: Addprefix, ChangeNbr: 010, applied in choosing IP route, 

and calling PSTN through termination gateway. 

� Rule 2:Change Type: Addprefix, ChangeNbr: 0, applied in calling mobile number, 

choosing IP route, and calling PSTN through termination gateway. 

� Rule 3: Change Type: Addprefix, ChangeNbr: 96446, applied in dialing long 

distance number through PSTN route. 

� Rule 4: Change Type: Delprefix, ChangeLen: 3, applied in switching to PSTN 

when IP network fails. 

� Rule 5: Change Type: Addprefix, ChangeLen: 1, applied in switching to PSTN to 

dial mobile number when IP network fails. 

5.5.3.1 Add Number Change Rule 

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Figure 30 Add Number Change Rule Diagram 


In Figure 30, add one number change rule: addprefix 17909, and pause for 2 seconds: 

Choose “AddPrefix” in “ChangeType”; input 17909 in “ChangeNbr” box; click “Add”. 

5.5.3.2 Modify number change rule 

Modify one existing number change rule: click index number block, and this rule will 

appear in the operation frame; you can modify “ChangeType”, “ChangePos”, 

“ChangeLen”, “ChangeNbr” according to your requirements; then click “Modify” button. 

5.5.3.3 Delete a number change rule 

Delete an existing number change rule: click index number block, and this rule will appear 

in the operation frame; then click “Delete” button. 

5.5.3.4 Delete all number change rules 

Delete all number change rules: click “Delete All” button directly, system will pop up a 

confirm box, after your confirmation, the rules will be deleted all. 

Notice � Please be cautious of using “Delete All” opeartion. 

5.5.4 DialedNBRChange 

Figure 31 Callout Number Change Table 

This page is used to configure Callout Number Change Rule and Fallback Number 

Change Rule. And for each rule, it can consist of multiple atom rules. 

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In the table, you should input the "DialNbr” item according to the format of pattern 

matching rule, (Refer to Section 8.5), utmost 50 pieces. Wildcard is supported. 

“Callout Number Change Rule Index “lists the rule which callout number change should 

adopt. The index number is the rule number of “Number Change Rule”, and common is 

used as the separate character between rule numbers. Callout number will change 

according to the rules one by one and then start next process. 

“Fallback Number Change Rule Index” lists the number change rules, when IP routing 

fails and switches to PSTN, whose numbers are index numbers of “Number Change Rule”, 

and these rule numbers are separated by common. 

For example: 

Suppose FXO connecting to PSTN need dial 0 to call out: (Figure 31) 

� Item 1: Dial fixed telephone number within the city such as 62303100: choose 

register IP route for registration; change into 01062303100 through number 

changing; If IP route fails, call will be switched to PSTN, and the fallback number 

will be 062303100 (according to Number Change Rule 4 and 2) and dialed out 

through PSTN. 

� Item 2: Dial mobile number within the city, such as 13010013298: choose PSTN 

route and the number will change into 013010013298, dial out through PSTN; 

� Item 3: Select P2P IP route for number with 9 at the beginning, such as 

962303100: if IP route fails, call will switch to PSTN, and the number will change 

into 062303100 (according to rule 5 and 2); dial out through PSTN; 

� Item 4: the number with 0 at the beginning select register IP route, such as 

02163078899: if IP route fails, call will switch to PSTN, and the fallback number 

will change into 09644602163078899; dial out through PSTN 

5.5.4.1 Set Callout Number Change Type 

Figure 32 Callout Number Change Type 

Callout Number Change Type support two models: “Number change before routing” and 

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“routing before number changing”: 


� Number change before routing: first change the outbound number, and then 

choose the route according to the number just changed. 

� Routing before number changing: first choose route according to outbound 

number, then change the outbound number. 

For example: the first item in Callout Number Change Table in Figure 31: 

[5-6;8]XXXXXXX:Add Prefix 010 

If Callout Number Change Type is changed to “routing before number change”, then the 

setting of “Routing Direction Table” is 

010[5-6; 8] XXXXXXX Register IP 

If the Callout Number Change Type is set as “Number change before routing”, then the 

setting of “Routing Direction Table” is 

010[5-6;8]XXXXXXX Register IP 

Tips: 

� If choose “ number change before routing”, recommend you to config 

“Callout Number Change Table”, then config “Routing Direction 

Table” according to the number after number change. 

� If choose “routing before number change”, you had better config 

“Routing Direction Table” first, then config “Callout Number Change 

Table”. 

5.5.4.2 Add Callout Number Change Item 

Figure 33 Callout Number Change Operation Diagram 

Add one callout number change item, as Figure 33 shows: 

Input rule of number in “DialNbr” box according to pattern matching grammar; input index 

number of number change rule in “Callout Number Change Rule Index” box; input index 

number of number change rule in “Fallback Number Change Rule Index”; Click “Add”. 

Click “Change Rule”, system will pop up a number change rule table, which you can refer 

to during configuration. 

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Tips 


� When the call switch to PSTN from IP route, the number change 

should start from the which has been changed before. 

For example, if you dial 62303100, then it will change into 

01062303100, and star to call through IP; If IP fails, call will be 

switched to PSTN, so now first you need delete the prefix 010, then 

the number can return to 62303100. 

5.5.4.3 Modify Callout Number Change Item 

If you want to modify an existing callout number change item, please click index number 

block, and this item will appear in the operation frame. After you modify the parameters, 

click “Modify”. 

5.5.4.4 Delete Callout Number Change Item 

If you want to delete Callout Number Change item, click the index number box, and this 

item will appear in the operation frame. Just click “delete”. 

5.5.4.5 Delete All Callout Number Change Items 

If you want to delete all callout number change items, just click “Delete All”, and system 

will pop up a confirm box. After your confirmation, all items will be deleted. 

Notice � Please be cautious of using “Delete All” function 

5.5.5 Caller Change 

Figure 34 Caller Number Change Rule 

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This page is used to configure Caller Number Change rules. And each one can consist of 

multiple basic rules. 

The port will use the changed number to replace original number, and call out, according 

to the number change rules set for the port. 

For example: 

In Section 5.5.3, 

The number change rule 4: “ChangeType” is “DelPrefix”, “ChangeLen” is 3; 

The number change rule 5: “Change Type” is “DelPrefix”, “ChangeLen” is 1; 

So for port 1, 1678017 will change into 8017, and call out. (According to rule 4) 

For Port FXO, if you choose “Yes” for “Enable PSTN Caller ID”, it can change PSTN 

caller id according to caller number change rule. 

Suppose that FXO ( port 5) has detected PSTN caller ID, which is 062302952, firstly 

original caller number 1678021 will change into 062302952, then change into 

62302952 according to rule 5, and finally call out. 

In SIP protocol, original port register number will be replaced by the changed number, 

and fill in the “DisplayName” property domain of INVITE message, in property domain 

of UserName, it is still original number. 

In H.323 protocol, original port register number will be replaced by the changed 

number, and input the changed number to Setup message. 

5.5.5.1 Set Caller Number Change Rule Item 

Input index number of Number Change Rule in “Caller Number Change Rule Index”, and 

click “Submit”. 

If you want to delete one item, just delete the index number of “Caller Number Change 

Rule Index”, and click “Submit”. 

Click “Change Rules”, and system will pop up “Number Change Rule Table”, for user’s 

reference. 

5.5.5.2 Delete All Caller Number Change Rule Items 

If you want to delete all items, please click “Delete All”, and system will pop up a confirm 

box. After you confirmation, all items will be deleted. 

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Notice � Please be cautious of using “ Delete All” function. 

5.5.6 RegenNBRChange 

Figure 35 Regeneration Number Change Diagram 


This page is used to configure number change rules from IP-To-PSTN, under One-stage 

mode. And for each rule, it can consist of multiple basic rules. 

One-stage mode (refer to Section 6.3.1), FXO port is used as trunk port of IP/PSTN. 

Caller of IP side, dial FXO number and callee number of PSTN side. Receiving the calling 

request, FXO port will change the number according to the number change rule of this 

port, and send the changed number to PSTN form FXO. 

In Section 5.5.3, 

The number change rule 4: “ChangeType” is “DelPrefix”, “ChangeLen” is 3; 

The number change rule 5: “Change Type” is “DelPrefix”, “ChangeLen” is 1. 

In Figure 35, all 1-8 ports of FXO are filled by 9. When IP side receive a call request of 

962302952 which is callee number, system will choose a free FXO port by port polling, 

(refer to Section 6.12) and change the number according to the number change rules, 

which means the callee number will be changed into 62302952 and sent to PSTN. 

5.5.6.1 Set Regenerate Number Change Rule Items 

If you want to add an item, please fill in “Regenerate Number Change Rule Index” with 

70


index number of number change rule, and click “Submit”. If you want to delete one item, 

please delete the index number of “Regenerate Change Rule Index”, and click “Submit”. 

If you click “Change Rules”, Number Change Rule Table will pop up. 


If you want to delete all items, please click “Delete All”, and system will pop up a confirm 

box. After your confirmation, all items will be deleted. 

Notice � Please be cautious of using “Delete All” function. 

5.6 Routing 

5.6.1 RouteDirection 

Figure 36 Route Direction Table 

This is used to configure route direction, utmost support 100 pieces. 

MG3000-R32 provides IP route and PSTN, and IP route of SIP protocol can be divided 

into register IP route and P2P IP route. (Refer to Section 6.8.1) 

Notice 

� The relation of route direction and number change rule is controlled 

by “Callout Number Change Type” parameters. Please refer to 

Section 5.5.4.1. 

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5.6.1.1 Add Routing Direction Table Item 


Input number rule in “DialNbr” in operation frame according to pattern matching rule, and 

choose “Routing Direction” parameters, then click “Add”. 

5.6.1.2 Modify Routing Direction Table Item 

If you want to modify an existing routing direction item, please click index number, and this 

item will appear in operation frame automatically. Modify “DialNbr”, “Routing Direction” 

according to requirements, and click “Modify”. 

5.6.1.3 Delete Routing Direction Table Item 

If you want to delete an existing routing direction item, please click index number, and this 

item will appear in operation frame automatically, then click “Delete”. 

5.6.1.4 Delete All Routing Direction Table Items 

If you want to delete all routing direction table items, please click “Delete All”, and system 

will pop up a confirm box. After you confirmation, all items will be deleted. 

Notice � Please be cautious of using “Delete All” operation. 

5.6.2 Peer to Peer Routing 

Figure 37 Peer to Peer Routing 

This page is used to configure parameters of “Peer to Peer Routing”, utmost support 100 

pieces. 

72

In Figure 37, there are 2 peer to peer calling route items: 


� Item 1: Dial Number is 9, Destination IP address is 192.216.224.168:5060 

� Item 2: Dial Number is 16700 ~ 16799, Destination IP address is 

192.216.224.167:5060. 

5.6.2.1 Add Peer to Peer Calling Route Table Item 

Input number rule in “ Dial Number” in operation frame according to pattern matching rule; 

input IP address in “Destination IP”; input port number in “ Destination IP”; Click “ Add ”. 

5.6.2.2 Modify Peer to Peer Calling Route Table Item 

If you want to modify an existing item, please click index number block, and this item will 

appear in the operation frame automatically. Please modify “Dial Number”, “Destination 

IP”, “Destination Port”, and click “Modify”. 

5.6.2.3 Delete Peer to Peer Calling Route Table Item 

If you want to delete an existing item, please click index number block, and this item will 

appear in operation frame automatically. Then click “Delete”. 

5.6.2.4 Delete all Peer to Peer Calling Route Table Items 

If you want to delete call items, please click “Delete All” directly. System will pop up a 

confirm box. After your confirmation, all items will be deleted. 

Notice � Please be cautious of “Delete All” operation. 

5.6.3 ACL 

73

Figure 38 Access Control Configuration Diagram 


This page is used to configure Access control, and utmost support 50 pieces. 

When access control is enabled, only the message from IP address in the list can get the 

reply. Otherwise, there will be no response. 

Notice 

� Access control function only works for H.323 and SIP singal 

message; does not affect the patch processing; and does not restrict 

the data of HTTP, TELNET, PING. 

5.6.3.1 Configure Access Control 

Choose “Enable” or “Disable” in pull-down menu of “Access Control Configuration”. And 

click “Submit”. If you choose “Enable”, you should configure “Access Control List”. 

5.6.3.2 Configure Access Control list Item 

Input IP addresses in “IP Address” of “Access Control List”, and click “Submit”. If you want 

to delete one address, you just need delete it from the table and click “Submit”. 

5.6.4 PSTN User Management 

MG3000-R32 can restrict calls in PSTN-To-IP direction through PSTN User Management, 

which can avoid illegal user calling from FXO port, or calling for free through illegal means 

in IP network. 

5.6.4.1 Set PSTN Inbound Limit Mode 

74

FXO Inbound Limit Mode includes: 

� No limit; 

� Limit, Keep Ringback; 

Figure 39 FXO Inbound Limit Mode Diagram 

� Limit, Off-hook for 1 s, then on-hook; 

� Limit, Off-hook and Play IVR. 


Choose one mode in pull-down menu of “FXO Inbound Limit Mode”, and choose “white 

list” or “Black list” in the pull-down menu of “Choice White/Black List, then click “Submit”. 

5.6.4.2 Set Black/White List 

Figure 40 White List Diagram 

If you choose “White list”, the page will display White List configure, and system will only 

permit the calls from “White List”; if you choose “Black list” the page will display Black List 

configure, and system will restrict the calls from “Black List” 

For both, list can support utmost 200 items. 

Take White List for example, only calls from 13301162166, 13301162177, 13301162188, 

and 13301162199 are permitted. (Refer to Figure 40) 

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Tips 

5.7 Information 

5.7.1 Port Status 


� Because the PSTN User Management function is based on the caller 

number, so this function requires PSTN line which FXO port connects 

to to support Caller ID service, and enable the detection of CallerID 

function in PSTN line; 

� the PSTN User Management function is based on original number in 

PSTN side, not the changed number, even if the number change 

rules is enabled in FXO port. 

Figure 41 Port Status Diagram 

Through this page, you can check the port status in real time. It will refresh every 5 

seconds. And in Figure 41, there will be 6 statuses: 

� Red: Disable 

� Green: Idle 

� Blue: Offhook 

� Yellow: Ringing 

� Black: Fail 

� Orange: FXO without Line or Line Error 

Tips � Facing rear panel, sub-cards and port list from left to right. 

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5.7.2 System Info 

Figure 42 System Information Diagram 


These page list program versions, Version Date, Device Series Number, System MAC 

Address, Current Time, Device Name etc... 

Figure 43 System Time Configuration Diagram 

In this page, you can set the system time. MG3000-R32 can support setting the time 

through “Manual” or “Net time”. 

In Figure 43, if you choose “Manual”, you can input the year, month, day, hour, minute, 

and second in corresponding items, and click “Submit”. 

Figure 44 NTP setting diagram 

While in Figure 44, if you choose “Net Time”, please fill in “Master NTP Server” and 

77


“SlaveNTP Server”, and choose the “Time Zone”, then click “Submit”. After the 

configuration, system will apply to NTP Server every 60 minutes, and refresh the net time 

according to the reply. 

5.8 Device Management 

5.8.1 Update 

Figure 45 Update Page 

MG3000-R32 supports 2 upgrade methods: HTTP and TFTP. You can get the upgrade 

patch from the new version patch released by our company, and the name of the patch 

should follow the specification of the rules. (Refer to Section 7.1) 

If you want to upgrade in HTTP, please select the upgrade patch, click “update”, then 

system will begin to upgrade. HTTP can upgrade 5 file systems: 

� Software: the packaged application software 

� Driver: the packaged driver program, including 3 modules: Boot, Firmware, and 

Kernel 

� Configure File: voip_config.ini、h323_config.val 

� Logo: customized enterprise Logo picture in the left top of the web page, and 

product name picture in the middle top of the web page. 

� Voice File: Voice package for inquiring IP address, and telephone number 

Notice 

� When you are uploading Logo pictures, please pay attention to the 

the picture size, it should be 224 (width) * 86 (height) pixels, and for 

product picture, the size should be 654 (width) * 86 (height) pixels. 

78


For both pictures, the file size should be no bigger than 100k bytes. 

� You need reboot the product to make the upgrade take effect. 

� During the upgrade, they product should be prevented from power 

off, otherwise the product can not run normally. 

If upgrade in TFTP, please fill in “TFTP server IP address”, and the name of the file, then 

click “Update”, the system will get the files from TFTP server and upgrade the system 

automatically. 

Notice 

5.8.2 Password 

� Through TFTP you can only upgrade software, drive and config file. 

� TFTP upgrade need run TFTP server in server. 

Figure 46 Password change Diagram 

This page is used to change the password. Username can not be changed here. User 

need input “Old password”, “New password”, and “Confirm New Password”. Click 

“Submit”, new password will take effect. 

5.8.3 Default setting 

Figure 47 Default setting Diagram 

Click “Recover Configuration”, system will pop up a confirm box. After your confirmation, 

all parameters of default setting will be recovered. 

Click “Clear Data”, system will pop up a confirm box. After your confirmation, all system 

parameters except basic network parameters will be recovered to default setting. 

79


When you finish the operation, save and reboot the system, parameters will be recovered 

to default setting. 

Notice 

5.8.4 Save 

� Please be cautious of using this funtion, because when you recover 

to default setting, all of your configuration will be lost. 

Click “Save” to save system configurations. 

Figure 48 SIP Protocol save Diagram 

This page also provides configure file extract function, so that you can back up your 

configuration. And in the future, you can download the configure file into the product 

through update function. (Please refer to Section 5.8.1) 

Figure 49 Configuration File Download Diagram 

SIP protocol‘s configure file name is voip_config.ini. You can click “Upload Configure”, and 

system will pop up a download box. You can select “Save” to download the file to local 

computer. (Please refer to Figure 49) 

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Figure 50 H.323 Save Configuration Diagram 


H.323 need two configuration files: voip_config.ini、h323_config.val. Click buttons to get 

the file, and you can choose “save” to download them to local computer. 

5.8.5 Reboot 

Figure 51 Reboot Diagram 

Click “Reboot”, system will pop up a confirm box. After you click “yes”, the product will 

reboot. 

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6 Service Function Description 


This section will describe the service principles and configure steps from service function 

point of view. 

6.1 Basic Service 

6.1.1 IP Voice Service 

MG3000-R32 can provide excellent IP voice quality, supporting G.711A, G.711U, G.729, 

G.7231, G.726 and iLBC etc. VAD, CNG and functions can reduce the bandwidth 

occupation. System also supports Voice Gain setting and echo cancel functions. 

6.1.1.1 Echo Cancel 

Figure 52 Echo caused by combined transmission imbalance 

Echo is a very important factor which has a very big impact on the voice quality. The 

breed-through delay when transport voice signal to receiver channel by mixed circuit (2/4 

line transform), or the reflection caused by the resistance mismatch, will cause the echo. 

In general, echo always exists in PSTN network, and sometimes it is not a problem. In fact, 

part of the phone signal will be coupled to receiver channel to generate side tone, which 

makes speaker can hear his own voice. If without sidetone, the speaker would not be 

certain that the other side can hear his voice or not and the dialogue would be very 

embarrassed. But if do not control the echo, it will affect the users’ call experience in 2 

ways: 

� The bigger the echo is, the greater the interference is. In many situations, there 

are low echo in the circuit, but the user can not feel it; 

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� Echo delay will affect the voice quality greatly. Echo delay means the time 

between the users speak and hear his own voice. If the echo delay is more than 

25 ms, the voice quality will be affected greatly. 

Mixed circuit completed the 2 lines to 4 lines conversion, and vice versa. Because of the 

combined imbalance of transmission (mismatching resistance, combining components not 

idea etc.), there is always some TX signal will enter Rx channel. (Refer to Figure 52) 

Figure 53 Echo caused by resistance mismatch 

Apart from the echo caused by combined transmission imbalance, mixed terminal 

resistance mismatch can also cause it. If the line terminal resistance is not appropriate, 

echo will be generated. Two lines local loop signal encounters mixed terminal resistance, 

and reflects back, thus the echo is generated. (Refer to Figure 53) If the terminal connects 

to inappropriate CPE products (such as phones or modems), echo may also be generated 

in local loop. 

Figure 54 Echo Distribution in VoIP network 

In VOIP network, transmission combined imbalance and resistance mismatch may cause 

echo. The echo caused by transmission combined imbalance will enter IP side, by 

resistance mismatch will enter PSTN side. In Figure 54, suppose A can hear the echo, 

which is overlapped by resistance mismatch in terminal A and transmission combined 

83


imbalance in terminal B. In general, from terminal A to phone A, the delay is short, so the 

echo caused by resistance mismatch in terminal can hardly impact the call quality; now 

the main echo source comes from the combined transmission imbalanced echo of 

terminal B. In order to eliminate the echo heard by phone A, we need eliminate echo in 

terminal B. 

MG3000-R32 provide the 128ms tail wave echo elimination capability (G.168), which can 

try to eliminate the combined transmission imbalance echo caused by transformation 

between 2 lines and 4 lines through the enhanced nonlinear echo control function. 

6.1.1.2 Configure Voice Coder 

In MG3000-R32, the PT values of G.726、iLBC、DTMF 2833 et. Coders need be 

configured by user, and should avoid conflict with the previous defined static PT value. In 

addition, the dynamic PT values also should avoid conflict with each other; otherwise the 

channel can not work normally. 

Table 24 VOICE CODER TYPE OF MG3000-R32 

Voice Coder Coder Type PT Value 

G.711U 0 0 

G.726-32 2 2 

G.711A 8 8 

G.7231 4 4 

G.729 40 18 

G.729A 18 18 

G.729B 42 18 

G.729AB 44 18 

G.726-16 22 dyn 

G.726-24 24 dyn 

G.726-40 26 dyn 

iLBC-13 28 dyn 

iLBC-15 30 dyn 

DTMF2833 NA dyn 

Table 25, Table 26 list the static PT values of RTP protocol (RFC 3551), and allow user to 

set. The coder whose PT type is “dyn” does not have designated static PT value in RTP 

protocol, thus the PT value can only be configured within dynamic range. 

Table 25 PT VALUE OF ALL KINDS OF VOICE CODERS 

PT Value Voice Coder Media Type Sampling rate(Hz) 

0 PCMU Audio 8000 

84

1 reserved Audio 


3 GSM Audio 8000 

4 G723 Audio 8000 

5 DVI4 Audio 8000 


7 LPC Audio 8000 

8 PCMA Audio 8000 

9 G722 Audio 8000 

10 L16 Audio 44100 

11 L16 Audio 44100 

12 QCELP Audio 8000 

13 CN Audio 8000 

14 MPA Audio 90000 

15 G728 Audio 8000 



18 G729 Audio 8000 


20 unassigned Audio 




dyn G726-40 Audio 8000 

dyn G726-32 Audio 8000 

dyn G726-24 Audio 8000 

dyn G726-16 Audio 8000 

dyn G729D Audio 8000 

dyn G729E Audio 8000 

dyn GSM-EFR Audio 8000 

dyn L8 Audio var. 

dyn RED Audio 

dyn VDVI Audio var. 

Table 26 PT VALUE OF ALL KINDS OF VIDEO CODER 


PT Value Video Coder Media Type Sampling Rate(Hz) 

24 unassigned Video 

25 CelB Video 90000 

26 JPEG Video 90000 


28 nv Video 90000 


85


31 H261 Video 90000 

32 MPV Video 90000 

33 MP2T AV 90000 

34 H263 Video 90000 

35-71 unassigned ? 

72-76 reserved N/A N/A 

77-95 unassigned ? 

96-127 dynamic ? 

dyn H263-1998 Video 90000 

6.1.1.3 Voice service configuration 


When you want to use basic voice services, please refer to the following steps: 

� Step 1: Basic Configure ---> Protocol, set protocol parameters (refer to 5.3.1); 

� Step 2: Basic Configure ---> Port, set port number (refer to 5.3.2 ); 

� Step 3: Basic Configure --->Channel, set channel parameters (refer to 5.3.3); 

� Step 4: NBR Exchange --->Change Rule, set number change rule (refer to 

5.5.3); 

� Step 5: Route ---> Routing Direction, set routing direction (refer to 5.6.1 ); 

6.1.2 IP FAX 

MG3000-R32 support 2 ways: T.38 fax and Pass Through: 

Pass through transports the fax signal in the same way as voice signal, decodes the 

signal in the opposite side, and sends the recovered modulating signal to the receiving fax 

machine. It need not special process of VOIP gateway, but packet loss, time delay, and 

jitter etc. will have high impact on the fax transmission quality. So if the network is not 

stable, pass through may fail or decrease the service quality. In addition, pass through is 

very sensitive to encoding signal, so when it is needed, we recommend set the voice 

coder as G.711, which takes more bandwidth. 

T.38 encapsulates the fax signal according to ITU-T T.38, which is independent of voice 

media transmission. When the transmitting gateway receives the fax signals and 

messages from transmitting fax machine, gateway will decode the signal into digital 

signals, which is encapsulated in IP fax protocol (IFP); the receiving gateway will 

modulate the IFP packets into fax signals and messages again, and send them to 

receiving fax machine. In order to decrease the transmission time delay, T.38 adopts UDP 

86


to send IFP packets. Besides, T.38 also defines the redundancy correcting mechanism, 

which can enhance the anti packet loss capability. Therefore, T.38 is a reliable IP FAX 

approach, and has no impact on voice coder function. 

Configuration steps: 

� Step 1: Basic Configure ---> Protocol, set protocol parameters (refer to 5.3.1); 

� Step 2: Basic Configure ---> Port, set port number (refer to 5.3.2 ); 

� Step 3: Basic Configure ---> Channel, set channel parameters (refer to 5.3.3); 

� Step 4: Routing ---> Route Direction, set routing direction (refer to 5.6.1 ); 

6.1.3 DTMF Secondary Dial 

MG3000-R32 supports three DTMF secondary dial methods: Pass Through, RFC2833 

and Signal. 

� Pass Through: encapsulate DTMF signals in the same way as voice signal, and 

send them to the opposite side for decoding and recovery; 

� RFC2833:encapsulate DTMF signals according to RFC2833/4733 requirements, 

which is independent of voice media transporting; 

� Signal: In H.323 protocol, use DTMF over H.245 (UserInput) to send DTMF; In 

SIP protocol, use SIP INFO signal to send DTMF signals. 

When use DTMF 2833 to send DTMF signals, you need pay attention to the parameter of 

“RFC2833 Payload Type”, and avoid conflict with system static definition or other defined 

PT values. For the set range, please refer to section 6.1.1.2. 

6.2 SPS (Supplementary Service) 

SPS can be achieved through terminal products and server. And for both methods, the 

problem of interconnection with server or opposite end product always exists. The 

following section will discuss about it. 

6.2.1 SPS through terminal 

SPS can be started in terminals, thus you can apply or cancel SPS in terminals. 

Adopting different protocols, MG3000-R32 can provide different SPS types: 

Table 27 SPS OF DIFFERENT PROTOCOLS 

SPS H.323 SIP 

Caller ID 来电显示 ● ● 

87

Call Hold ○ ● 

Call Transfer ○ ● 

Call Forward ○ ● 

Do Not Disturb ○ ● 

Caller ID Block ○ ● 

Block Anonymous Call ○ ● 

Hotline ● ● 

Notes:○---not support; ●---Support 

6.2.1.1 Caller ID 


MG3000-R32 supports sending and detecting the caller ID signals. The product can send 

the caller ID signals to FXS extensions, and detect the caller ID signals from FXO line. 

MG3000-R32 supports two Caller ID standards: FSK, DTMF, and for FSK, MDMF and 

SDMF are all supported. 

Refer to the standards: 

� > 

� > 

6.2.1.2 Call Hold 

6.2.1.2.1 Function Introduction 

When subscriber A establish a call with subscriber B, A can keep B in call hold status 

through pressing hook or flash key, and restore the call through pressing hook or flash key 

again. 

A is called as the subscriber initiating hold, and B is called as subscriber in hold. When in 

call hold status, voice data sending is stopped between A and B. Call hold function is not 

used alone, and normally applied with Call Forward function. 

6.2.1.2.2 Notes: 

1) Call Hold is only valid for SIP protocol; 

2) Not only caller but also callee can start the Call Hold function; 

3) When in Call Hold status, both parties can restore the call through pressing hook 

and flash key; 

4) In different applications, the phenomena may be different: if the product 

interconnects with server, the subscriber in hold can hear music or silence 

88


because of the different process capabilities of servers. And all are normal 

phenomena. While if the product interconnects with terminal directly, it is more 

often that the subscriber in hold will hear the silence. 

6.2.1.3 Call Transfer 


A complete Call Transfer flow is as follows: (suppose there are three subscribers: A, B, C) 

1) A dials the number of B; then B picks up the phone and intend to transfer A’s call 

to C; 

2) B press hook or flash key, make the call in hold status; 

3) When B hears the dial tone, B dial the number of C; 

4) C will hear the ring, while A and B will hear the ringback tone; 

5) C picks up the phone, and now the call between C and A is established; B will 

6.2.1.3.2 Notes: 

hear busy tone, and the Call transfer is done successfully. 

1) Call Transfer function is only valid for SIP protocol; 

2) Presmise: Call Hold function must be enabled at the same time 

3) MG3000-R32 adopts REFER-NOTIFY to realize Call Transfer function (Follow 

RFC3515 agreement), so require the server or the equipment interconnected 

with to support REFER and NOTIFY; 

4) MG3000-R32 call flow is based on “Session Initiation Protocol Call Control– 

Transfer” (draft-ietf-sipping-cc-transfer-06 March 5, 2006), and its characteristics 

are: 

a) B who start the Call Transfer will not establish a call with C; 

b) During the process of A, who is transferred to, establishing the call with C, B 

are noticed by NOTIFY, and can monitor the whole Call Transfer flow, deal 

with it in different way according to different Call Transfer results, which 

means if the Call Transfer fails, the call between A and B can be restored. 

5) For Call Transfer function, it can be realized through other ways. Normally, it can 

be realized basing on three party services. In this way, A, who starts the Call 

Transfer, need initiate the call to third party B (B can pick up the phone or not), 

89


and use REFER + REPLACES to realize. However currently, MG3000-R32 can 

not support this way. 

6) Not only Caller but also Callee can start the Call Transfer function. 

7) In configuration, please pay attention: for the equipment which starts the Call 

Transfer, it need configure the route to third party. If in peer to peer routing, 

product also need configure peer to peer route; while for the equipment, which is 

transferred to, it is not necessary to configure such routes. 

6.2.1.4 Call Forward 

6.2.1.4.1 Unconditional Call Forward 

No matter which status subscriber A is in, all the calls to A will be transferred to the forward 

number which is set by the subscriber before. 

6.2.1.4.2 Busy Call Forward 

When subscriber A is busy, all inbound calls to A will be transferred to the forward number 

which is set by the subscriber. 

6.2.1.4.3 No Answer Call Forward 

When the no answer time exceeds the set time, all inbound calls to A will be transferred 

the forward numbers set by the subscriber. 

6.2.1.4.4 Selective Call Forward 

If the inbound calls to subscriber A meet with the conditions set by the subscriber, they will 

be transferred to the numbers set before. 

6.2.1.4.5 Notes 

1) Call forward service is based on SIP protocol, so it is only valid for SIP version; 

2) Call forward service starts the function through responding to the 302 Moved 

Temporarily message, and the forward number and address are saved in the 

contact domain of 302. So during the call, the caller is required to process 302 

message, and recall the third party in Contact domain; 

3) For selective call forward service, MG3000-R32 decides whether to start the call 

forward function mainly according to the caller number, so user should set the 

“Caller NBR” and “FWD NBR”, and for each port, utmost 8 pairs can be 

configured. 

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4) For the forward services discussed above, the priority is as follows ( from high to 

low): Unconditional Call Forward, Busy Call Forward, Selective Call Forward, No 

Answer Call Forward: 

a) If subscriber A can enable all the forward services, all the calls will be 

forwarded to the unconditional call forward number, and other forward 

services do not come into effect 

b) If subscriber A enable the forward services other than unconditional call 

forward service, when A is busy, even if caller number meet with the 

selective call forward conditions, the call will still be forwarded to the busy 

call forward number A. 

c) If subscriber A enable the forward services other than unconditional call 

forward service, when A is available, if the caller number meets with the 

selective call forward conditions, the call will be transferred to the 

corresponding selective call forward number; 

d) If subscriber A enable the forward services other than unconditional call 

forward service, when A is available, if the caller number does not meet with 

the selective forward condition, if the callee does not pick up the phone 

when the time exceeds the “No Answer Call FWD Timeout”, the call will be 

transferred to the “No Answer Call FWD NBR”. 

5) Call forward service only has configure requirements on the callee products: for 

the forward numbers of all types and caller number of selective call forward 

service, only need input the caller number; besides, the corresponding routing 

direction of forward number need be configured. If it is peer to peer, the routes of 

peer to peer also need be configured. 

6.2.1.5 No Disturb 

6.2.1.5.1 Function Description 

When you do not want to be disturbed, you can enable “No Disturb” service. All inbound 

calls can only get the busy tone, but the outbound calls are not restricted. 

6.2.1.5.2 Notes: 

1) “No Disturb” service is based on SIP protocol, so it only take effect for SIP 

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version; 


2) When “No Disturb” service is enabled, all “Call Forward” service can not take 

effect, and directly reply in 486 Busy. 

6.2.1.6 Caller ID Block 


When you do not want your caller ID displayed in the callee’s terminals, you can enable 

the “Caller ID Block” service, and hide your caller ID. 

6.2.1.6.2 Notes: 

1) “Caller ID Block” service is based on SIP protocol, and it only take effect for SIP 

version; 

2) MG3000-R32 realizes Caller ID Block service through set Displayname in From 

Domain of INVITE message to be “Anonymous”, and does not change the 

Username. The reason is it will not affect the billing function of interconnection 

platform. But there still will have these problems: 

a) If the product in opposite side not only read the DisplayName segment, but 

also read UserName segment during displaying the caller ID, the “Caller ID 

Block” function will not take effect; 

b) Some platform will refuse a call directly because the DisplayName segment 

is “Anonymous” during authentication, and in this case, please do not enable 

“Caller ID Block” service. 

6.2.1.7 Block Anonymous Call 


This function is relative to “Caller ID Block” service. When you enable the “Block 

Anonymous Call”, and if the caller ID of incoming call is blocked, the call will be 

terminated. 

6.2.1.7.2 Notes: 

1) “Block Anonymous Call” is based on SIP protocol, and it is only valid for SIP 

version; 

2) MG3000-R32 only read DisplayName segment in From domain of INVITE 

message. If it is “Anonymous”, the call will be regarded as Anonymous call. And 

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if the “Block Anonymous Call” is enabled at the same time, the reply will be 406 

Not Acceptable. 

6.2.1.8 Hotline 热线功能 


If user does not dial number within the set time, the product will start a call to hotline 

number automatically. 

6.2.1.8.2 Notes: 

1) Hotline function can be realized, no matter the protocol is SIP or H.323; 

2) Besides the hotline number, you also need configure the hotline number routing, 

direction. If the routing direction is peer to peer, you need configure the peer to 

peer route. 

6.2.2 SPS through Terminal cooperation with Server 

6.2.2.1 Function Description 

Supplementary Service can also be realized through Server, but it needs terminal’s 

cooperation. Terminal can book or cancel the service. 

SPS can be booked or cancelled through web interface of the server; and it can also be 

realized by sending special code through terminals. There are 2 ways: 

1) Use INVITE message to book or cancel, and send the special code as the callee 

number to the server; (refer to section 6.2.2.3.1) 

2) According to RFC3265, SUBSCRIBE/NOTIFY mechanism should be used to 

book or cancel service. Terminal use SBSCRIBE message to book and cancel 

SPS, Server use NOTIFY message to inform terminal the status of booking or 

cancel. 

For different SPS, terminal can start the SPS in 2 ways: 

1) Terminal first build a call with server, and then send special code to server 

through DTMF secondary dial. For example, call transfer service of server is 

normally achieved in this way. 

2) Use INVITE message directly, set special code as callee number and send it to 

server. For example, Group Pickup, realized by server( for details refer to 

6.2.2.3.2) , requires the terminal to support 2 ways at the same time. 

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6.2.2.2 Notes 

1) Only valid for SIP version; 


2) For the SPS book and cancel function, MG3000-R32 currently only support 

INVITE message; but SPS through terminal can support both; 

3) Because server and terminal deal with the SPS in different ways, so we suggest: 

when you use server to realize the SPS, please disable the SPS function of 

terminal, and vice versa. The reason of doing so is that we can control the result 

of the service realization clearly. For example: Call Transfer function, both server 

and terminal can realize. If for server and terminal, the function is enabled at the 

same time, the “Unconditional Call Forward” will first choose the forward number 

of the server; but for “Busy Call Forward” function, it is not certain. So we disable 

the function of the terminal can avoid such situation. 

6.2.2.3 Examples 

6.2.2.3.1 Example 1 terminal book the service from server 

This case explains how to book the “Unconditional Call Forward” function from SS3000-T 

server, in which the booking code is *57*. 

Step 1: As Figure 55 shows, configure the number begin with * into “Register IP Route”; 

Figure 55 Configure *57* Routing direction 

Step 2: user pickup the phone and dial *57* + Unconditional FWD NBR, such as 

“*57*16801. MG3000-R32 send the request to the server, and the callee number is 

*57*16801 in INVITE message. 

94


We also need pay attention, booking code of some other server may start with #, such as 

#57#, MG3000-R32 should avoid other configuration relating to #. 

Step 3: Disable “Dial Special Number to Change Route” function, or change “the Number 

to Change Route” into other character except #. 

Figure 56 Disable “Dial Special Number to Change Route” 

Step 4: Disable # as the number end flag: (Figure 57) 

Figure 57 Disable “Use # as number end flag 

Step 5: Configure #57# dial plan. Because # is not used as the number end flag, the end 

flag is according to the dial pan or waiting overtime. Therefore you should configure 

number often used in the “DigitMap Table”. (Refer to Figure 58). 

95

Figure 58 #57# dial plan configure 


Step 6: configure #57# routing direction. Set #57# routing direction as “Register IP Route”. 

(Figure 59) 

Figure 59 #57# routing direction configure 

Step 7: User pickup the phone and dial #57# + Unconditional FWD NBR, such as 

#57#16801. MG3000-R32 send the request to Server, and in INVITE message, the callee 

number is #57#16801. 

6.2.2.3.2 Example 2: SPS realized by Server 

1 Call Transfer realized by SS3000-T server 

In SS3000-T, “9” is configured to start call transfer function, and need be sent to server 

through secondary dialing in SIP signal. 

Step 1: Subscriber A set “DTMF Trans Mode” as Signal (SIP INFO); 

Step 2: Subscriber A calls B through server, B pickup the phone and the call is 

established; 

Step 3: B send “*9” + number of C to server through secondary dial (SIP INFO signal); 

Step 4: Server send RE-INVITE message to A, and make A at call hold status; 

Step 5: Server send BYE to B, end the call from A; 

Step 6: Server start the call to C, C rings; 

Step 7: C picks up the phone, server sends RE-INVITE message to A, cancels A’s 

hold status; the call between A and C is built, call transfer is done. 

96

2 Group Pickup of SS3000-T 

In SS3000-T, the special code of Group Pickup is “*6”: 


Step 1: C configure the number with the initial letter “*” in Register IP Route; 

Step 2: A call B through server, and B rings 

Step 3: C pickup the phone, dial “*6” + the number of B. C sends the INVITE request 

to server, and the callee number is “*6” + the number of B; 

Step 4: Server transfer the call of B to C, B stops ringing, and the call between A and 

C is established. 

6.3 FXO Mode 

6.3.1 IP-To-PSTN 

For IP-To-PSTN direction, it can be divided into 2 phases: IP-FXO and FXO-PSTN; and 

according to the call establishing way, it can be divided into: One-stage and Two-stage. 

Under One-stage mode, secondary dial is not required for the user of IP side; FXO 

gateway can get the callee number of PSTN side, and send it to PSTN side automatically. 

Under Two-stage mode, secondary call is necessary for the user of IP side. 

Tips: � Under dual mode, IP-To-PSTN can not work. 

6.3.1.1 One-stage 

6.3.1.1.1 One-stage 

97

Figure 60 One-stage flow 


One-stage can help user to avoid doing the secondary call, the product can do the call of 

FXO-PSTN phase automatically, and the key point is how to get the callee number of 

PSTN side. In Figure 60: 

� When the user of IP side dials a call, the PSTN number must be dialed together 

with FXO number, which means the user dial all the number for one time, such 

as 962304138; 

� Receiving the call, R32 will extract the callee number from the signal message, 

and change the number according to the Regenerate Number Change Rule. For 

example, if the “ChangeType” is “DelPrefix” and the “ChangeLen” is “1”, 

962304138 should be changed into 62304138; 

� R32 will choose one available FXO port, off hook, and send 62304138 to PSTN 

Cautions: 

switch. 

1、 At first, in IP-FXO phase, no matter the Routing Direction is P2P IP Route or 

Register IP Route, it must be guaranteed that the call can be reached. Under 

one-stage mode, the callee number of IP-FXO call request may not be the same 

as the register number of FXO port, so some servers may not let it pass the call 

authentication. 

2、 When R32 automatically send the number to PSTN, the cooperation with PSTN 

switch may have trouble. If send the number too fast, and PSTN switch is not 

ready, the opposite side can not receive the total number; If the interval of 

sending number is too small, and the detection sensitivity of switch is not high 

enough, the opposite side can also not detect the total number. R32 provides 2 

parameters: “Regenerate DTMF Digit Interval” and “Regenerate DTMF Digit 

Delay”, which can help further adjust the cooperation with PSTN switch in 

sending number under One-stage mode. 

In fact, the time point of sending DTMF number to PSTN line can be identified through dial 

tone detection theoretically. But because the dial tone detection itself has a very high 

98


requirement on the compatibility of the product, is very complicated in realization, and also 

need a detection delay, so currently R32 does not do so. 

Main configuration: 

� Step 1: Basic Configure ---> Port ---> Port Configuration ---> set PhoneNbr, such 

as 9 ( refer to section 5.3.2); 

� Step 2: NBR Exchange ---> ChangeRule ---> Number Change Rule Table: such 

as delete prefix in one number (refer to section 5.5.3); 

� Step 3: NBR Exchange ---> RegenNBRChange: configure number change rule 

of One-stage mode corresponding to FXO port (refer to section 5.5.6); 

� Step 4: Advanced Configure ---> FXO: set “Regenerate DTMF Digit Delay” and 

“Regenerate DTMF Digit Interval”. Normally you can use the default value (Refer 

to section 5.4.2). 

6.3.1.1.2 Answer Supervision 

Figure 61 Answer supervision diagram 

During the call of IP-To-PSTN, billing is also divided into two phases. For IP-FXO phase, 

the billing usually takes place in server side, triggered by the connecting signal which is 

sent to server by R32; For FXO-PSTN phase, the start point of billing is the user of PSTN 

side pick up the phone. In Figure 61 flow, two phases of billing is not synchronized, and 

this situation may happen: user of PSTN is not reached, but the billing of FXO-IP phase 

has started. 

99


For solving this problem, R32 provide the Answer Supervision function, which need PSTN 

provide reversal signal. When R32 send the signal to PSTN line, it will not reply 200 OK 

responses to IP side. Only after user of PSTN side pickup the phone and the voltage of 

the line is reversed, R32 reply 200 OK to IP side. In this way, we can guarantee the billing 

of IP side and PSTN side to be consistent. 

Actually, Answer Supervision can also be realized through voice detection, answer delay 

etc., but all of them has some disadvantages, so R32 does not support them. 

Related configuration: 

� Step 1: Basic configure ---> Port ---> Port Configuration: Configure PhoneNbr = 

“9” (refer to section 5.3.2); 

� Step 2: NBR Exchange ---> ChangeRule ---> Number Change Rule Table: 

Configure number change rule, such as “ChangeType” = “DelPrefix”, 

“ChangeLen” = “1” ( refer to section 5.5.3 ); 

� Step 3: NBR Exchange ---> RegenNBRChange ---> Regeneration Number 

Change Table: Configure the “Regenerate Number Change Rule index” of FXO 

port in One-stage mode. ( refer to section 5.5.6 ); 

� Step 4: Advanced Configure ---> FXO: Set “Regenerate DTMF Digit Delay” and 

“Regenerate DTMF Digit Interval” (refer to 5.4.2 ); 

� Step 5: Advanced Configure ---> FXO: Configure “FXO Answer Supervision” = 

“After Receiving Reversal Signal. (Refer to 5.4.2 ). 

6.3.1.1.3 CRBT Transport Transparently 

100

Figure 62 CRBT Flow Diagram 


During the call of IP-To-PSTN, users hope system can transport the CRBT or voice 

prompts of unable to access of PSTN side to the user of IP side, so that the user can 

understand the real situation of the callee. In Figure 61, the flow can be modified a little: 

make R32 reply 183 messages when receiving INVITE request, so that in the process of 

establishing a call, user of IP side can hear ring back tone, CRBT and record prompts etc. 

transported from PSTN side transparently. 

However, CRBT transports transparently function must cooperate with Answer 

supervision function, which means reversal signal of PSTN line is required. Otherwise, 

R32 will not be able to know the callee of PSTN side has picked up the phone, but actually, 

at that time the media channel has been built, and the call is ok, while the billing is 

omitted. 

Related Configuration: 

� Step 1: Basic configure ---> Port ---> Port Configuration: Configure PhoneNbr = 

“9” (refer to section 5.3.2); 


Configure number change rule, such as “ChangeType” = “DelPrefix”, 

“ChangeLen” = “1” ( refer to section 5.5.3 ); 

� Step 3: NBR Exchange ---> RegenNBRChange ---> Regeneration Number 

Change Table: Configure the “Regenerate Number Change Rule index” of FXO 

101

port in One-stage mode. ( refer to section 5.5.6 ); 


� Step 4: Advanced Configure ---> FXO: Set “Regenerate DTMF Digit Delay” and 

“Regenerate DTMF Digit Interval” (refer to 5.4.2 ); 

� Step 5: Advanced Configure ---> FXO: Configure “FXO Answer Supervision” = 

“After Receiving Reversal Signal. (Refer to 5.4.2 ). 

� Step 6: Basic Configure ---> Protocol ---> Other Configuration ---> Configure 

“FXO Provisional Response Type” = “183” ( refer to 5.3.1); 

6.3.1.2 Two-stage Mode 

Figure 63 Two-stage Flow Diagram 

Two-stage mode dial twice: User of IP side dials FXO number first ( Figure 63), then FXO 

port off hook, user of IP side will hear the dial tone, then dials for the second time, reaches 

callee through PSTN Switch. 

Under Two-stage mode, billing of IP-FXO phase is always earlier than of FXO-PSTN 

phase, and sometimes this situation may happen: billing of IP side has started already but 

PSTN side is still not get through. 

6.3.2 PSTN-To-IP 

PSTN-To-IP can be divided into PSTN-FXO, FXO-IP, and has 3 modes: normal mode, 

Auto mode, and dual mode. 

6.3.2.1 Normal Mode 

102

Figure 64 Normal Mode Flow Diagram 


In normal mode, user of PSTN side need dial twice. Firstly, dial the FXO number, and 

secondly, dial the destination number of IP side. So there is a problem: billing of 

PSTN-FXO, FXO-IP is not synchronized, which means even IP side is not reached, PSTN 

side still has billing list. 

And the call of PSTN-To-IP direction usually has demand on displaying the caller ID of 

PSN side. R32 provides caller ID transport transparently function, which can replace the 

caller ID by the caller ID of PSTN side, and server need support at the same time. 

R32 can also change the Caller ID of PSTN side, such as change 062304138 into 

62304138. 

Related configuration: 

� Step 1: Advanced Configure ---> FXO ---> FXO Advanced Configuration: 

“Working Mode” = “Normal Mode”; ( refer to section 5.4.2 ) 


“Normal Mode Play Record” = “No”; ( refer to section 5.4.2 ) 

� Step 3: Advanced Configure ---> FXO ---> “Enable PSTN CallerID” = “Yes”; 

( refer to section 5.4.2 ) 

� Step 4: Advanced Configure ---> FXO ---> “Ring Times Detected” = “2” ( For 

detecting incoming call signal, and should be more than 1 time ); ( refer to 

section 5.4.2 ) 

103



“ChangeType” = “DelPrefix”, “ChangeLen”= 1; (refer to section 5.5.3 ) 

� Step 6: NBR Exchange ---> CallerChange ---> Caller Number Change Rule 

Index: configure “Caller Number Change Rule Index”. ( refer to section 5.5.5) 

6.3.2.2 Auto Mode 

IP PHONE 

OFFHOOK 

IP PHONE 

INVITE 

100 TRYING 

180 RINGING 

200 OK 

ACK 

FXO PSTN EXCHANGER 

VOICE 

FXO DETECT RING 

CALLER ID 


FXO OFFHOOK 

FXO SEIZES LINE 

Figure 65 Auto Mode Flow Diagram 

Auto mode is different compared with normal mode, because FXO of R32 does not pick 

up and get the call when it detects the incoming call from PSTN side. Instead, it will 

transfer the call to the destination number which is set before in IP side, and only after it 

receiving the connecting signals from IP side, FXO will be in pick up status. In this way, the 

system can guarantee the billing of PSTN-FXO and FXO-IP phase is in synchronization, 

so that the problem of starting billing in PSTN side without getting through will not happen. 

The key point of Auto mode is how to get destination of IP side automatically. R32 

designates it through configurations. 

You need pay attention that, user of PSTN side will always hear ringback tone no matter 

user of IP side is busy or available, because FXO is not in picking up status, so that user 

of PSTN side can not get the real situation of IP side. 

Auto mode can also support Caller ID of PSTN side transport transparently function. 

Related Configurations: 


104

“Working Mode” = “Auto Mode”; ( refer to section 5.4.2 ) 


� Step 2: Advanced Configure ---> FXO ---> FXO Advanced Configuration ---> 

Click “display Auto FWD NBR”; ( refer to section 5.4.2 ) 



� Step 4: Advanced Configure ---> FXO ---> “Ring Times Detected” = “2” ( For 

detecting incoming call signal, and should be more than 1 time ); ( refer to 

section 5.4.2 ) 





6.3.2.3 Dual Mode (4S4O) 

IP PHONE 

OFFHOOK 

IP PHONE 

NOTIFY TO RING 

NOTIFY TO SEND CID 

NOTIFY TO RING 

NOTIFY FXO OFFHOOK 

OPEN LOOPBACK CHNL 

FXO PSTN EXCHANGER 

VOICE 


CALLER ID 


FXO OFFHOOK 

FXO SEIZES LINE 

Figure 66 Dual Mode Flow Diagram 

Dual mode provides bridge function between pairs of FXS and FXO ports, which requires 

all subcards to be 4FXS4FXO. Dual mode can be applied to the user who has PSTN lines, 

and wants VOIP function at the same time. PSTN line of user connects to FXO port, and 

phone connects to FXO port. Then when a call from PSTN to IP (in PSTN-To-IP direction), 

the extension of FXS corresponding to FXO port rings immediately; if phone of FXS call 

out, it can choose both IP and PSTN route. 

105

Tips: 


� Dual mode can guarantee the Caller ID of PSTN side transport 

transparently function, which is not under the control of “Ring Times 

Detected”. 



“Working Mode” = “Dual Mode”; ( refer to section 5.4.2 ) 







6.3.3 PSTN Routing 

Figure 67 PSTN Routing Flow Diagram 

For the R32 with FXO ports, it can also configure the PSTN routing function. Acting as the 

caller, R32 is able to start the call which can choose IP route, direct the call to the protocol 

entity; or choose PSTN route, transfer the call to PSTN network through FXO port. 

When system confirms the PSTN route, it will poll all available FXO ports to find a random 

one to call out. The algorithm of polling ports can be set as: Round-robin mode, and 

increasing mode. (Refer to 6.13) 

You must pay attention to difference of dual mode realization. For dual mode, FXO and 

FXS are used in pairs. When FXS extension pick up the phone, the corresponding FXO 

106


port is also in pick up status at the same time; when dual mode choose PSTN route, it will 

use the fixed FXO port to call out, not poll the ports in random. 

6.4 Call Progress Tone 

MG3000-R32 support call tones of 13 countries, allow user to define the format, and also 

support the auto learning function. 

References: 

� 【ITU-T E.180/Q.35】-“…Tones in national signaling systems … Technical 

characteristics of tones for the telephone service” 

� 【Annex to ITU-T E.180, Annex to ITU Operational Bulletin No. 781. 1.II.2003…】 

- “Various tones used in national networks (according to ITU-T Recommendation 

E.180) (03/1998)”, Feb 2003. 

� 【ETSI – EN 300 001】- “…General technical requirements for equipment 

connected to an analogue subscriber interface in the PSTN” 

6.4.1 Tones of different countries 

In normal situation, user can choose the tone among 13 countries which has been 

configured previously. 

Table 28 MG3000-R32 TONES FORMAT 

Countries Tones Frequency(Hz) Signal melody 

China 

Taiwan 

Japan 

Korea 

Dial tone 450 Continuous 

Ring back tone 450 1.0 on 4.0 off 

Busy tone 450 0.35 on 0.35 off 

Unobtainable Tone 450 3x(0.1 on 0.1 off) 0.4 on 0.4 off 

Dial tone 350+440 Continuous 

Ring back tone 440+480 1.0 on 2.0 off 

Busy tone 440+480 0.5 on 0.5 off 

Unobtainable tone 








Unobtainable tone 450 0.2 on 0.1 off 0.2 on 1.5 off 

USA Dial tone 350+440 Continuous 

107

Germany 

India 

Russia 

France 

England 

Italy 

Brazil 

Spain 



Unobtainable tone 400 0.5 on 6.0 off 



















Ring back tone 400+450 0.4 on 0.2 off 0.4 on 2.0 off 


Unobtainable tone 400 Continuous 

Dial tone 425 0.2 on 0.2 off 0.6 on 1.0 off 







Unobtainable tone 425 0.75 on 0.25 off 0.25 on 0.25 off 




Unobtainable tone 425 0.5 on 2.5 off 

6.4.2 Self-defined tone format 

If the tone formats of 13 countries are all not suitable for the client’s request, MG3000-R32 

allows the user to define the tone according to the standardized format. MG3000-R32 

support the tone of 3 different melodies, and the format should be: 

(C1LowFreq@C1LowFreqLevel+C1HighFreq@C1HighFreqLevel, 

C1TOn/C1TOff);C2LowFreq@C2LowFreqLevel+C2HighFreq@C2HighFreqLevel,C2TOn 

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/C2TOff) ;(C3LowFreq@C3LowFreqLevel+C3HighFreq@C3HighFreqLevel,C3TOn/C3T 

Off) 

Explanations: 

Table 29 SELF-DEFINED TONE FORMAT DESCRIPTION 

Parameters Description Unit Range 

C1LowFreq Low frequency of first segment Hz 0~4000 

C1LowFreqLevel Low-frequency gain of first segment dbm -63~0 

C1HighFreq High frequency of first segment Hz 0~4000 

C1HighFreqLevel High-frequency gain of first segment dbm -63~0 

C1TOn On time of first segment ms 0~65535 

C1TOff Intermittent time of first segment ms 0~65535 

C2LowFreq Low frequency of second segment Hz 0~4000 

C2LowFreqLevel Low-frequency gain of second segment dbm -63~0 

C2HighFreq High frequency of second segment Hz 0~4000 

C2HighFreqLevel High-frequency gain of second segment dbm -63~0 

C2TOn On time of second segment ms 0~65535 

C2TOff Intermittent time of second segment ms 0~65535 

C3LowFreq Low frequency of third segment Hz 0~4000 

C3LowFreqLevel Low-frequency gain of third segment dbm -63~0 

C3HighFreq High frequency of third segment Hz 0~4000 

C3HighFreqLevel High-frequency gain of third segment dbm -63~0 

C3TOn On time of third segment ms 0~65535 

C3TOff Intermittent time of third segment ms 0~65535 

Tips 

� for continuous signal, Ton is time length, TOff is 0 

� If only one segment melody, all parameters of C2、C3 are set as 0. 

For example: user can accord the following format to set the tones of Oman: 

Dial Tone: 425Hz,Continuous: 

(425@-15+0@0, 1000/0);(0@0+0@0,0/0);(0@0+0@0,0/0) 

Ring Back Tone: 425Hz,0.4 on 0.2 off 0.4 on 2.0 off: 

(425@-15+0@0,400/200);(425@-15+0@0,400/2000);(0@0+0@0,0/0) 

Busy Tone: 425Hz,0.5 on 0.5 off: 

(425@-15+0@0,500/500);(0@0+0@0,0/0);(0@0+0@0,0/0) 

Unobtainable Tone: 425Hz,0.2 on 0.3 off: 

(425@-15+0@0,200/300);(0@0+0@0,0/0);(0@0+0@0,0/0) 

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6.4.3 Busy Tone Detection 


In FXO’s applications, busy tone detection is a very important function. Only FXO port can 

detect the busy tone sent by PSTN Switch accurately, and avoide the hanging up problem 

of FXO port. Because of the multiple types of PSTN switch, so FXO is required to be 

compatible to all kinds of busy tones. And MG3000-R32 has excellent compatibility and 

Adaptability. 

MG3000-R32 has 3 methods for busy tone detection: 

1) Pre-set 13 countries tone formats for user to choose; 

2) If the tone is not in 13 countries, and user know the tone parameters, user can 

define the tone according to 6.4.2 description; 

3) If the tone is not in these 13 countries, and the user does not know the exact tone 

parameters, user can operate the system into auto learning function according to 

6.4.4. Get the parameters of busy tone from the line, then use what has learned 

to do the detection. 

6.4.3.1 Tones detection of designated countries 

In “Call Progress Tone Detecting Configuration” ---> “Country or district”, the default is 

“China”, the range includes: 

� China 

� Taiwan 

� Japan 

� Korea 

� USA 

� Germany 

� India 

� Russia 

� France 

� England 

� Italy 

� Spain 

� Brazil 

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Detection parameters include: 

� Detect Type 

� Noice Control 

� Cadence Cycles 

� Detect Delay Time) 

� (Drop Delay Time) 

Parameters of Detect Type include: 

0:One-tone detecting use lower FFT 

1:One or two tone detecting use higher FFT 

2:Two-tone detecting use lower FFT 

3:detect on band endge 

Parameters of Noise Control include: 

0:Noise-estimate after removing one tone 

1:Noise-estimate after removing two tone 

2:Noise-estimate after removing three tone 

Parameters of Cadence Cycles: 

0:on time (Verify one on time only): default 

1:verify one on-off time (Verify one on-off time cadence) 

2:verify tow on-off time(Verify two on-off time cadence) 

3:verify three on-off time(Verify three on-off time cadence) 

The range of “Detect Delay Time” is 10~500ms, and the default is 300ms; 

The range of Drop Delay Time is 10~80ms, and the default is 40ms. 


In normal situation, when detecting the tones, for these 5 control parameters, use the 

default value is enough, but sometimes, they also need adjustment. For example, for the 

busy tone of Brazil (425Hz,0.25s on/0.25s off ), the on-off time is relative short, so “Detect 

Delay Time” (200ms) or “Drop Delay Time” need be shortened properly. 

6.4.3.2 Customized Tones Detection 

If the line busy tone is not the same with that of designated country or district, and user 

can get the specific parameters of the line busy tone, user can customize the tone format. 

The following parameters need be configured: 

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In “Call Progress Tone Sending Configuration”, the “Country or District” need be set to 

“other”; and in the page, dial tone, ring back tone, busy tone, unobtainable tone shows. 

Input the parameters of busy tone according to section 6.4.2. 

Also, the control parameters of Call Progress Detecting need be set. (Refer to section 

6.4.3.1) 

6.4.3.3 Busy Tone Auto Learning 

If line signal is different from the busy tone of the country or district, and user can not get 

the specific parameters of the line busy tone, user can get the line busy tone parameters 

through auto learning. 

In “Call Progress Tone Detecting Configuration”, the “Country or District” chooses 

Learning; and Enable Busy Tone Learning. (Refer to section 6.4.4 ) 

6.4.4 Busy tone Auto Learning 

Please operate according to the following steps: 

1、 Configure the learning parameters: 

� “Country or District” = “Learning”; 

� “Enable Busytone Learning” = “Yes” 

� “Learning Port”: choose which one you need to learn the busytone; 

� “Learning Frequency Mode” = “Using One Frequency” or “Using Two Frequency” 

2、 Connect line which sends the busy tone to the FXO port which is ready to learn, 

and send busy tone to FXO port in the following ways: 

a) FXO port is set to Normal Mode, call from PSTN line FXO port off hook and 

user does not dial. After 10s, the secondary dial is overtime, PSTN line 

sends busy tone; 

b) FXO port is set to Normal Mode, call from PSTN line. FXO port off hook, and 

user dials a wrong number, PSTN line sends busy tone; 

c) FXO port is set to Normal Mode, call from PSTN line,FXO port off hook, and 

user dials a correct number, but callee does not pick up the phone. When it 

is over time, the PSTN line will send busy tone; 

d) FXO port is set to Normal Mode, call from PSTN line,FXO port off hook, and 

user dials a correct number, callee answer the phone. After talk, caller of 

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PSTN line on hook first, and the PSTN line will send busy tone; 

During the learning process, keep busy tone for some time (>30s), in order to 

guarantee the learning accuracy. 

3、 Observer FXO Learning Busy Tone Status: 

Modify parameters: 


� “Enable Busytone Learning” = “Yes” and Submit 

Observer the format of busy tone in the display box change or not: 

If it changes, we may regard it has learned the busy tone, and use the busy tone 

learned to detect; if it does not change, we can conclude that it did not learn the busy 

tone, and adjust the “Learning Frequency Mode”, repeat 1, 2, 3. 

4、 Finish Learning Busy Tone, Switch to Busy Tone Detection Mode: 

Modify parameters: 


� “Enable Busytone Learning” = “No”; 

� Busy tone: display the busy tone learned 

� Detection Type 

� Noise Estimation Control 

� Cadence Cycles 

� Detect Delay Time 

� Drop Delay Time 

For the tones through auto learning, you need pay atthention when you configure the 

detection configurations: 

(1) If the busytone learned has only one frequency, “Detection Type” choose 

“one-tone detecting use lower FFT”; “Noise Estimation Control” choose “after 

removing one tone”; and “Cadence Cycles” choose “Verify one on-off 

cadency”; 

(2) If the busytone learned has two frequencies, “Detection Type” choose 

“two-tone detecting use lower FFT”; “Noise Estimation Control” choose “after 

removing two tones”; and “Cadence Cycles” choose “Verify one on-off 

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cadency”; 


(3) “Detect Delay Time” can be set according to your requirement. The default 

value is 300ms. If on time of the busy tone learned is short, you can decrease 

this value properly; 

(4) “Drop Delay Time” doe not have too much impact on the detection result, so 

you can use the default value 40ms; 

(5) After finish the busy tone learning, you can switch to busy tone detection. 

5、 Detecting busy tone according to step 2: 

Observe whether FXO port has detected correct busy tone signal or not. If busy tone 

is detected, FXO will be released immediately. 

If busy tone is learned successfully, but it can not be detected, you can modify the 

parameters and repeat the detection progress; if both learning and detection are 

successful, you need save the configurations, and back up the parameters of the 

busy tone which was learned just now. However, if for both of learning and detection 

are failed, you can record the busy tone by software or some other tools, and send it 

to our company. Our technical experts will help you make it. 

6.5 Number Match 

MG3000-R32 provides flexible number match rules, supporting pattern matching 

algorithm, which is very easy to configure. 

6.5.1 Pattern Match Rule 

MG3000-R32 support wildcard input and pattern grammar, which can be applied to 

DigitMap, callout number match, basing on the following table: 

Table 30 PATTERN MATCH RULE 

Pattern Description 

0 Matching number 0 








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* Matching number * 

# Matching number # 

X or x Matching any single digit between 0~9 

Z or z Matching any single digit between 1~9 

N or n Matching any single digit between 2~9 

. Matching any single digit or multiple digits between 0~9 

[5-8] Matching any single digit between 5~8 

[1;5-8] Matching any single digit among 1、5、6、7、8 


The following table shows the common number set rules, and user can choose according 

to the actual demand: 

Table 31 COMMON NUMBER RULE 

Dial number rule Description 

[5-6;8]XXXXXXX Local number 

ZXXXXXXX 8 digits number and the initial number is not 0 

NXXXXXXX 8 digits number and the initial number is not 0 or 1 

11X Local emergency call number, 110~119 

12[0-2] Local special service number, 120~122 

123XX Local special service number 

95XXX Local special service number 

0. Long-distance number, and no digits restriction 

013XXXXXXXXX Long-distance mobile number (12 digits) 

13XXXXXXXXX Local mobile number (11 digits) 

6.5.2 Match rule 

If you need configure multiple callout number rules, please refer to the following principles: 

1) Callout number matching: If at the same time there are several match rules, 

system will follow principle of matching the longest. For example: 

Rule 1:623. 

Rule 2:62301 

If the dialout number is 62301, system will choose rule 2. 

2) If there are several rules with the same matching length, and can match 

successfully, system will follow the principle of matching the most accurately. The 

match priority is : ’0’-‘9’ >’X’,’Z’,’N’>’.’。 

For example: 

Rule 1:623. 

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Rule 2:623XX 

Rule 3:62301 

If the dial out number is 62301, system will choose rule 3. 


In general, if the matching length is the same, system will compare each digit of 

the number, from highest number start. If the priority is the same, then system 

will compare the priority of relatively low number, until there will be an exact 

result. For the number of same position, system will compare it according to rule 

2). 

For example: 

Rule 1:6X301 

Rule 2:62X01 

Rule 3:623X1 

Rule 4:6230X 

Rule 5:62301 

Rule 6:623. 

If dial out number is 62301, the priority of the rules is: 

rule 5>rule 4>rule 3>rule 6>rule 2>rule 1 

6.5.3 Application 

In applications, “DigitMap”, “Routing Direction”, “Peer to Peer Routing”, “Callout Number 

Change” all need number matching rule function. 

6.5.3.1 拨号计划表 

MG3000-R32 adopts Enblock to collect and send the callee number, and provides 3 end 

methods: Digitmap, using # as end flag, interval overtime. 

In order not to change the habit of users, you can set the Digitmap to confirm that user has 

finish dialing the number: No need add # after the number, and also no need to prolong 

the response time for waiting overtime. 

DigitMap items should be filled according to pattern matching rules. Once the dial 

plan rule is set, if the callout number matches successfully, the number will be sent to 

system immediately for routing process. 

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Cautions 


� DigitMap comply with the pinciple of matching the shortest, so “.” is 

forbidden in the DigitMap. 

6.5.3.2 Routing Direction Table 

“Routing Direction Table” is used set the routing direction for matched number. Fill in the 

DialNbr according to the pattern match rules. 

Table 32 ROUTING DIRECTION TABLE 

Index DialNbr Routing Direction 

1 [5-6;8]XXXXXXX Register IP 

2 13XXXXXXXXX PSTN 

3 9. P2P IP Route 

4 0. Register IP 

In the table, there are several typical routing direction items: 

Item 1:Local fixed phone choose Register IP route, and call through termination 

gateway; 

Item 2:Local mobile phone choose PSTN route; 

Item 3:the number with initial digit “9” choose P2P IP Route (suppose one FXO 

gateway provides the termination function, and FXO port number is 9) 

表项 4:Long-distance number with initial digit “0” choose Register IP route, and call 

through termination gateway. 

6.5.3.3 Callout Number Change Table 

Callout Number Change table is used to change the callout number, and Fallback Number 

Change Rule. In the table, “Callout Number” should comply with the pattern match rules. 

Continue with the example in Section 6.5.3.2, suppose we need dial 0 to PSTN line, so 

the configuration can be as follows: (set the Callout Number Change Type: Routing before 

number changing) 

Rule 1: “Change Type” = “AddPrefix”, “ChangeNbr”= 010; 

Rule 2: “Change Type” = “AddPrefix”, “ChangeNbr”= 0; 

Rule 3: “Change Type” = “AddPrefix”, “ChangeNbr”=96446; 

Rule 4: “Change Type” = “DelPrefix”, “ChangeLen”= 3; 

Rule 5: “Change Type” = “DelPrefix”, “ChangeNbr”= 1; 

Table 33 CALLOUT NUMBER CHANGE EXAMPLE 

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Index Callout number Callout Number 

Change Rule Index 

1 [5-6;8]XXXXXXX 1 4,2 

2 13XXXXXXXXX 2 

3 9. 5,2 

4 0. 2,3 


Fallback Number 

Change Rule Index: 

Item 1: Dial fixed number within the city, such as 62303100, choose Register IP 

Route. The number changes into 01062303100 through the callout number change, 

and call through termination gateway. If IP route fails, fall back to PSTN, and the 

number will change into 062303100 calling out through PSTN route; 

Item 2: Dial mobile number within the city, such as 13010013298, choose PSTN route, 

and the number will change into 013010013298, which calls out through PSTN route; 

Item 3: the number with initial character “9” selects P2P IP Route, such as 962303100. 

If IP route fails, fall back to PSTN route, and the number will change into 062303100, 

which calls out through PSTN route. 

Item 4: the number with initial character “0” selects Register IP Route, such as 

02163078899. If IP route fails, fall back to PSTN route. The number will change into 

09644602163078899, which calls out through PSTN route. 

Tips 

� Callout number change and route direction, which should process 

first? It is controlled by “Callout Number Change Type”. In general, if 

choose “Routing Before Number Change”, you should config the 

“Routing Direction Table” first, and then config “Callout Number 

Change Table”; If choose “Number Change Before Routing”, you 

should config “Callout Number Change Table” first, and then config 

“Routing Direction Table” according to the changed number. 

6.5.3.4 Peer to Peer Calling Route Table 

“Peer to Peer Calling Route Table” gets the peer to peer destination IP according to the 

callout number which matches “Dial Number”. In the table, “Dial Number” should comply 

with the pattern match rules. 

Tips 

� No matter you choose“Routing Before Number Change” or 

“Number Change Before Routing”, system will both use the changed 

number to match the peer to peer route. So you should set the peer 

to peer route according to the changed number. 

118

6.6 PSTN Caller ID 


MG3000-R32 provides PSTN Caller ID detection, transformation and transportation 

function. Once FXO port detects the Caller ID of PSTN side, it will use this number to 

replace the origin FXO number, or if you have configured the “Caller Number Change 

Rule Index”, the Caller ID may change according to the change rules, and the changed 

Caller ID will be sent to callee of IP side. So, for the calls in PSTN-To-IP direction, the 

phone of user in IP side can display the true Caller ID of PSTN side. 

MG3000-R32 can detect 2 kinds of incoming call display signals: FSK and DTMF. User 

can change the incoming call number according to the requirements, and send the call 

message with modified Caller ID to the user of IP side. 

In SIP protocol, use the Caller ID of PSTN side to replace the origin port register number, 

and fill in the DisplayName domain of INVITE message, but the property domain of 

UserName still keep the origin port register number. Thus, the server is required to 

process the authentication and billing according to the UserName, and the Callee IAD is 

required to extract the Caller ID from DisaplyName domain first. 

In H.323 protocol, use the Caller ID of PSTN side to replace the origin port register 

number, and fill in Setup message. So the number which server get is the Caller ID of 

PSTN side. This also requires the server can accept the call from unregistered number, 

otherwise the call can not be established, and IP side can not display the Caller ID of 

PSTN side. 

For configuration, please refer to section 6.3.2. 

Tips 

� The PSTN line connecting to FXO port should support Caller ID 

service; 

6.7 Change Rule 

� If Caller ID from PSNT side need change again, you can realize it 

through configuring “Caller Number Change Rule Index”. 

MG3000-R32 has provided plenty of number change function. One complicated number 

change rule is made up of multiple smaller rules which here are called atom operation: 

� AddPrefix 

� DelPrefix 

119

� AddSuffix 

� DelSuffix 

� InsertNbr 

� DelNbr 

� KeepNbr 

� ReplaceNbr 


In “Number Change Rule Table”, we have defined specific change rules. When you want 

to do Callout Number Change, Caller Number Change, Fall Back Number Change, 

Regeneration Number Change, you can cite the index number of the rules, and combine 

them flexibly. 

For each atom operation, there are some parameters as follows: 

ChangeType: choose which operation you want 

ChangePos: Configure the place where you want to change in the origin number 

ChangeLen: Configure the length need change 

ChangeNbr: Configure the numbers which are used for change 

Table 34 NUMBER CHANGE RULE DESCRIPTION 

Item ChangeType ChangePos ChangeLen ChangeNbr 

AddPrefix Choose 

“AddPrefix“ 

DelPrefix Choose 

“DelPrefix” 

AddSuffix Choose 

“AddSuffix” 

DelSuffix Choose 

“DelSuffix” 

InserNbr Choose 

“InserNbr” 

DelNbr Choose 

“DelNbr” 

NA NA Fill in the Prefix 

NA Fill in how many 

characters of 

Prefix which need 

be deleted 

which need be 

added 

NA NA Fill in the Suffix 

NA Fill in how many 

Fill in the sequence 

number after which 

insert the numbers 

Fill in the sequence 

number after which 

characters of Suffix 

which need be 

deleted 

NA 

which need be 

added 

NA 

NA Fill in the number 

Fill in the length of 

deleted number 

which need be 

inserted 

NA 

120

KeepNbr Choose 

“KeepNbr” 

ReplaceNbr Choose 

“ReplaceNbr” 

delete the number of 

designated length 

Table 35 NUMBER CHANGE EXAMPLES 

Index Change Type Change 

Pos 

NA Fill in the length 

which need be kept 


NA NA Fill in the number 

Change 

Len 

Change 

Nbr 

NA 

for replacement 

Description 

1 AddPrefix 010 AddPrefix 010. e.g. 62303100 

change into 01062303100 

2 DelPrefix 1 Del prefix with 1 character. e.g. 

013010013298 change into 

13010013298 

3 AddSuffix 4138 AddSuffix 4238. e.g. 6230 change 

into 62304138 

4 DelSuffix 4 DelSuffix with 4 characters. e.g. 

62304138 change into 6230 

5 InserNbr 1 10 Insert 10 after the first character. 

e.g. 062303100 change into 

01062303100 

6 DelNbr 1 2 Del 2 characters after the first 

character. e.g. 01062303100 

change into 062303100 

7 KeepNbr 4 Keep 4 numbers. e.g. 

62304138change into 4138 

8 ReplaceNbr 62304138 Replace the number totally. e.g. 

During the rules configuration, please pay attention to: 

62303100 change into 62304138 

1) Support “*”, “#” characters input, which represent for the “*”, “#” in the keyboard; 

2) Support “;” character input, every “;” represents for 1s interval. It is normally used 

in PSTN route or RegenNBRChange, and controls the pause time between 

prefix and the origin number; 

3) For the complicated number change requirements, you can combine the rules to 

realize it and use “,” to separate the different rules; 

6.8 Intelligent Route 

MG3000-R32 provide intelligent rouging function: on the one hand, user need not change 

121


the dialing habit according to the routing direction which is set according to the number; on 

the other hand, when IP network have problems, the call can fall back to PSTN route, 

which is very reliable. 

6.8.1 Routing Direction 

MG3000-R32 provides IP route and PSTN route: 

� IP Route: 

� Register IP Route:send call request to registered server 

� P2P IP Route: inquire matched destination IP address according to P2P IP 

route table, and send call request 

� PSTN Route: call from PSTN side through FXO port 

For SIP protocol, the IP route can be divided into Register IP Route and P2P IP Route, 

which means SIP protocol can support register call and P2P call at the same time; H.323 

can not support both together: if enable registration, the call will choose Register Route, 

and if disable the registration, the call will choose P2P Route, and choose “IP” 

Tips 

6.8.2 IP Fall Back 

� If MG3000-R32 is not equipped with FXO, or FXO port does not 

connect to PSTN line, PSTN routing function will be unavailable 

� Default route is PSTN Route. If the route is not configured, it will 

choose PSTN route to call out. 

In actual applications, IP network, VOIP products sometimes may have troubles, which 

cause the VOIP call not work. MG3000-R32 can provide backup route if IP route can not 

get through, which can guarantee the reliability of the call. IP fall back function can be 

controlled by the switch. 

6.8.2.1 Fall back within IP net 

For SIP protocol, the fallback within IP net process sequence is: 

Register IP Route -> P2P IP Route -> Internal Call 

1) If register fails, or the call sending to the server fails, the call will fall back to P2P 

IP Route, and call the destination directly according to the parameters set in Peer 

o Peer “Route Calling Route Table”; 

2) If P2P call fails, and callee and the caller belong to the same R32, the call will fall 

back to internal and the internal of product will renew the call request; 

122


If the function of “Fall back within IP net” is disabled, IP call which fails will not fall back. It 

will go to the next step for judgment. (Please refer to section 6.8.2.2 ). 

For H.323 protocol, fallback within IP net process sequence is: IP net -> Internal Call 

Because H.323 protocol can not support register IP route and P2P route at the same time, 

so no matter the registration function is enabled or not, as long as the fallback within IP 

net is enabled, IP call which fails will fall back to the judgment for internal call directly. 

While if fallback within IP net function is disabled, IP call which fails will go to the next step 

for judgment. (Please refer to section 6.8.2.2 ) 

6.8.2.2 Auto Mode Call Fallback to Local FXS Extension 

In some special application occasions, user hope the PSTN-To-IP call can fall back to one 

random FXS extension of local R32 if it fails. This function is under control of the auto 

mode fallback switch which can be realized in command line. 

If the call within IP net fails, and the auto mode fallback function is disabled, system will 

judge the call direction. If the call is from auto mode, and there has local available FXS 

port, the call will be transferred to this free FXS port. Otherwise, the call will go to the next 

step. (Please refer to section 6.8.2.3 ) 

Related configurations: 

Use scf command, Enable “Enable call fallback to local idle FXS port in 

AutoForwarding mode ”: 

server) 

VOIP GATEWAY:\sys\devcfg>scf 

Enable SIP T38 traversal (e.g. working with SS3000-E) 

0 --- No 

1 --- Yes [ 0 ]: 

Enable SIP not respond retran invite request (e.g working with Russia CTCOM 

0 --- No 

1 --- Yes [ 0 ]: 

Enable SIP play ringback if recv bye while transferring (e.g working with 

ZHENGYOU server) 

0 --- No 

123

1 --- Yes [ 0 ]: 


Enable call fallback to local idle FXS port in AutoForwarding mode (Korea 

customers' requirement) 

0 --- No 

1 --- Yes [ 0 ]:1 

Enable T.38 special interworking process with WAYGOING server (Shanxi Provincial 

Department of Finance) 

0 --- No 

1 --- Yes [ 0 ]: 

Enable detect reversal when callee onhook (supplemental function ) 

0 --- No 

1 --- Yes [ 0 ]: 

The command has been completed successfully! 

6.8.2.3 IP Fallback to PSTN 

If after the previous two phases process, the call still does not succeed, we should confirm 

whether the function of “IP Routing Fallback to PSTN” is enabled or not. If enabled, the 

call will fall back to PSTN through FXO port. 

Because IP net and PSTN net numbers may have differences, after the call fall back to 

PSTN from IP net, the callee number can be changed to adapt to the dial requirements of 

PSTN. 

6.8.3 Dial Special Number to Change Route 

Under normal circumstances, after MG3000-R32 finishes collecting the number dialed, it 

will choose the matched route to call out. But under some special situation, user hopes 

switch the route through dialing a special number. 

Related configurations: 

� Quick Setup ---> Dial Special Number to Change Route: default is “disable” 

� Quick Setup ---> the Number to Change Route: default is “#” 

� Quick Setup ---> Default Route(only for dual mode): Default is: “Default IP, Dial 

Special Number to PSTN” 

6.8.3.1 Dial Special Number to PSTN 

124

1) FXS port pick up, and R32 play dial tone of IP side; 


2) Dial special number ( e.g. # ) to switch to PSTN, and hear the dial tone from 

PSTN line; 

3) Dial callee number of PSTN side ( such as 62303100), and the callee is 

Tips 

connected; 

� In this configuation, if user want to start a call from IP side, just 

directly dial the number of IP side after picking up and can enter IP 

call process. 

� If dial speical number to change to PSTN route, network parameters 

6.8.3.2 Dial Special Number to IP 

inquiry, local port number inquiry etc. can not take effect. 

1) FXS port off hook, and connects to FXO port at the same time, you will hear the 

dial tone from PSTN line; 

2) Dial special number ( such as # ), switch to IP route, you will hear dial tone 

played by R32; 

3) Dial Callee number of IP side, and the call is built. 

Tips 

� Under this configuration, If you want to call PSTN, just dial PSTN 

number directly after pick up the phone; 

� FXS off hook, if the call can not switch to IP route, network 

parameters inquiry, local port number inquiry can not take effect; 

� This function only take effect under Dual Mode 

6.9 DNS SRV Function 

When MG3000-R32 adopts SIP protocol, it support DNS SRV function (RFC 2782) of SIP 

server. DNS SRV function develops from DNS, and it allows the inquiry fro service or 

protocol, and returns the address records. 

When MG3000-R32 boots, and sends the first register request, system will start the DNS 

inquiry according to the domain name set by the user, and get an IP address list, and try 

registering to the first IP address of the list. If succeeds, keep the address, and the 

following registration or call message will be all sent to this IP address; If fails, delete the 

address from the list, and try registering to the second IP address; the rest may be done in 

the same way. Not until all the IP addresses of the list are deleted, and the list is empty, 

system will not repeat DNS RR (Round-Robin) process in one registration period, and get 

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a new IP list through DNS inquiry according to the domain name set by the user. 

In the future, if registration fails to the previous successful IP address, system will try it to 

the next address of the IP address list, until registration succeed. If IP address is empty, 

and system will do the DNS inquiry and register in one registration period. 

6.10 Answer Supervision 

Please refer to section 6.3.1.1.2. 

6.11 Disconnect Supervision 

For the call of IP-To-PSTN or PSTN-To-IP directions, if PSTN side on hook, the R32 which 

has the FXO port will be required to know the status of the user in PSNT side in time, to 

release the FXO ort and disconnect the call of IP side. Otherwise, FXO may hang up. 

For normal PSTN network, one part releases the call, and sends busy tone to the other 

part to remind user to on hook; or sends reversal signal to tell the other side that the call 

has been released. MG3000-R32 provides busy tone detection and reversal signal 

detection function, for supervising the on hook status remotely, and inform the release of 

IP side in time. 

6.11.1 Busy Tone Detection 

Please refer to section 6.4.3. 

6.11.2 Reversal Signal Detection 

This function requires the line to provide the reversal signal transporting function. 

6.12 Early Media 

MG3000-R32 does not provide the remote ringback tone function, or transparent 

transporting early media from PSTN side. 

Receiving Early Media: H.323 in faststart mode; SIP receive 183 SDP message and 

process it; 

Transport Early Media transparently: FXO send 183 SDP message. (Please refer to 

section 6.3.1.1.3 ) 

6.13 Port Polling 

In many circumstances, MG3000-R32 will do port polling operation: 

1) PSTN Route, poll a free FXO port and call out; 

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2) IP fall back to PSTN Route: poll free FXO port and call out; 


3) Auto Mode call fall back to Local FXS port: poll a free FXS port to answer it; 

4) Accept call from IP side: in the ports of same number, poll a free port to answer it; 

Polling approaches: 

� Round-robin Mode: each poll starts from the port next to the port which was used 

last time; and when the last port is polled, system will do it from the first one; 

� Increasing Mode: each poll from the first port of the product. 

Default is Round-robin mode. This mode can balance the service flow among all available 

ports; while increasing mode will concentrate the service in the fore ports, and will not use 

the back ports unless the fore ports are fully used, so the port service rate will decrease 

from the fore part to the back part. 

Tips 


� In Dual Mode, for PSTN route and IP fall back to PSTN route, the 

port will not do the poll, but use the fixed FXO port corresponding to 

FXS port: 1-5、2-6、3-7、4-8 

Port poll mode can only set by command line: 

sys\devcfg>ssppm 

6.14 Security Mechanism 

6.14.1 Encryption 

MG3000-R32 support encryption function in signal and media, which can improve the 

security of VOIP service, avoid being interfered or blocked. 

MG3000-R32 provides RC4 algorithm, Net-2-com algorithm, which requires the opposite 

side to use the same algorithm. 

Related configuration items: 

� Enable Encryption or not: if modified, need reboot the system; 

� Encryption Algorithm: take effect immediately 

� Encryption Key: take effect immediately 

6.14.2 ACL 

Considering the security, MG3000-R32 provides ACL function. For example, 

MG3000-R32 requires supervising the call from IP side when the product is used as 

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termination product. Only the call from authorized addresses can be accepted. In this way, 

you can prevent illegal users to peculate the termination resources and PSTN phones 

( refer to section 5.6.3). 

6.14.3 PSTN Incoming Call Authentication 

Please refer to section 5.6.4. 

7 Maintenance and Trouble Shooting 

7.1 Update 

MG3000-R32 update function has added version criterion inspection and data verifying 

mechanism, which increases the reliability of update operation, avoids the update failure 

due to wrong operation or packets loss in network transmission. 

For update, MG3000-R32 provides two ways: HTTP, TFTP. We recommend HTTP. 

Cautions: 

7.1.1 HTTP Update 

� Whatever you choose, the update patch should get from our 

company, and you should not modify the name of the program, or 

use unofficial version, otherwise, you may have trouble in updating. 

Http update files include: software, driver, Configure file, LOGO, Voice file. And the name 

of the program should comply with the following rules, otherwise it will cause verifying 

error. 

Table 36 VERSION NAME RULES 

TYPE NAME RULES EXAMPLE 

Software Product version number_app MG3000-R32_V1.3.5_20080320115736_app 

Driver Product version number_driver MG3000-R32_V1.3.5_20080320115736_driver 

Configure file System configure file: 

voip_config.ini 

H.323 protocol config file : 

h323_config.val 

LOGO Logo:companylogo.jpg 

Product name picture: 

productname.jpg 

Voice file Chinese: product version 

number_ivr_ch 

English: Product version 

number_ivr_en 

MG3000-R32_V1.3.5_20080320115736_ivr_ch 

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In version updating, the verifying steps include: 


1) Before update program is written into flash, firstly, check the version number 

which complies with the rules or not. It is required to comply with the rules of 

table 36. If it does not comply with it, update will be denied; 

2) Compare the update program version number and the current program 

version number, if they are same, update will be denied; 

3) Check the update program version in CRC, if it has error, the update will be 

denied. 

7.1.2 TFTP Update 

TFTP update files include: TFTP server IP and Update File Name. 

Cautions: 

� TFTP is a simple file transport protocol, basing on UDP, and the 

reliablilty is not as good as HTTP; 

� If update in TFTP, the version name should comply with the Table 36. 

7.1.3 Update procedures 

If the current version number is before MG3000-R32_V1.3.4_20071224135633, and want 

to update to the new version, (such as MG3000-R32_V1.3.5_20080320115736) you 

should: 

1. Firstly, update application program: Choose HTTP in update page, and select program 

file, file name choose “ramdisk_r32_v1.3.5_20080320115736.image”, click “update”. After 

update successfully, reboot the system; 

2. Update driver: after step 1, select “MG3000-R32_V1.3.5_20080320115736_driver” in 

“driver”, and after updating successfully, reboots the product; 

If the current version number is MG3000-R32_V1.3.4_20071224135633 or after this, and 

want to update to the new version (MG3000-R32_V1.3.5_20080320115736), you should: 

1. Firstly update software: choose “software” in the update page, and select the program 

“MG3000-R32_V1.3.5_20080320115736_app”, and click “update”. After update 

successfully, reboot the system; 

2. Update driver: after step 1, choose “Driver” in the update page, and select the program 

“MG3000-R32_V1.3.5_20080320115736_driver”, and click update; reboot the system 

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after update successfully. 

7.2 Diagnosis 

7.2.1 Port status check 


MG3000-R32 provide FXO port status check function, you can check the port status 

through web page (refer to section 5.7.1) 

Port Status includes: Disable, Idle, Offhook, Ringing, Fail, and FXO without Line or Line 

error. Disable, Fail, FXO without Line or Line error are abnormal status, which mean the 

FXO port is not available. Thus the call may have problems such as can not reach the 

destination (busy tone) or other abnormal phenomena, so please check the port status 

first. 

7.2.2 Output Trace information 

MG3000-R32 provides trace output interface, which can help engineers diagnose and 

trouble shoot. Trace currently can only output through command line. 

7.2.3 IP Packets 

Under some situation, you may need to catch IP packets to help diagnose the problem. 

Tcpdump, wireshark are the tools which are normal and very widely used. 

7.3 Debug Function 

7.3.1 WAN Port Mirror 

For MG3000-R32 whose switch chip is IP175D, you can set LAN to supervise the data 

sending and receiving situation of WAN interface, MAC chip, and MSP chip. So if you 

connect PC to LAN interface, you can get the supervising data. The configuration is as 

follows: 

VOIP GATEWAY:\debug>sls 

Please input LAN port supervision 

0--disable 

1--supervise WAN port 

2--supervise MAC port 

3--supervise MSP port : 1 


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Attention: sls command can only work when the switch chip is IP175D. If the switch 

typeisIP175C, sls command will not work, and you can only get the data packet through 

putting WAN interface and PC network interface in the same HUB; 

VOIP GATEWAY:\debug>sls 

SORRY: it is only available in IP175D switch! 


7.3.2 Output TDM Data 

Sometimes, you may need product provide TDM data to do the analysis. Please do it 

according to the following steps: 

� Step 1: Connect PC and R32 to 2 ports of the same HUB; 

� Step 2: Enable the output through sd command in debug. 

VOIP GATEWAY:\debug>sd 

TDM samples trace monitoring: 

DIAG_TDM function control param[0-disable; 1-enable]:1 

host ip address recv diag packet(monitoring host):192.216.2.1 

DIAG_TDM output direction[0-RX; 1-TX; 2-RX and TX]:2 


Please input the PC ip address in “host ip address recv diag packet” 

� Step 3: Recur the trouble, and catch the packets in PC, and do not filter the data 

packet of layer 2; 

� Step 4: Shutoff the function of output TDM data 

VOIP GATEWAY:\debug>sd 

TDM samples trace monitoring: 

DIAG_TDM function control param[0-disable; 1-enable]:0 


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8 Normal Troubles 


8.1 Can not establish a call due to configure error in 

number matching rules 

In “DigitMap”, “Callout Number Change Table”, “Routing Direction table”, “Peer to Peer 

Calling Route Table”, the number configuration should comply with pattern matching 

rules, otherwise, these functions can not work. 

There are some cases which happen very often, such as: 

callee. 

If User wants to configure a number with 0 as initial letter into Register IP Route, so 

he configures: 

Dialout number: 0 

Route direction: Register IP Route 

But he can only hear busy tone after he dials the number, and can not reach the 

Refer to Section 6.5.1, you should configure the dial out number as “0.” 

8.2 PSTN Caller ID display error 

If product can detect PSTN caller ID, but the result is not correct, and every time the 

detected Caller IDs are same. For this problem, it may because user set the “Caller 

Number Change Rule” in FXO port, so please check the caller number change table, and 

delete those unnecessary rules. 

For example: if in the PSTN side caller ID is 62304138, but in IP side the callee number is 

4138. It may because in FXO port, user has set a caller number change rule such as 

deleting 4 prefix numbers. 

If the caller ID of PSTN side can not be detected, and there is no line error, please refer to 

7.3.2, and catch data of TDM side. Then send these data to the engineer of our company, 

we will help you solve it.。 

8.3 If the callee number is #57#, how to configure it? 

Some servers require the terminal to start the call, in which the callee number is #57# for 

applying or canceling some SPS. In MG3000-R32, character # can be used to act as end 

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flag or other special functions, so you need modify related configurations for avoiding 

conflict. For specific configuration, please refer to section 6.2.2.3.1. 

8.4 System can not start after version update 

It is usually because you have downloaded wrong version program. So you need pay 

special attention to check the file name and size before downloading. After the 

confirmation, you can start to do the update. ( refer to 7.1) 

If the system still can not be started, please contact the engineer of our company. 

8.5 Can not ring with multiple phones in parallel 

connection 

MG3000-R32 allows 5 phones in parallel connection within short loop distance (300 

meters), 3 phones within long distance (1500 meters). When the number exceeds the 

maximum number, or the line distance is too long, or some phones equivalent resistances 

are big, MG3000-R32 may be unable to drive all of them into ringing . You can try it by 

adjusting FXS parameters ( Advanced Configuration ---> FXS Configuration ---> Ring 

Voltage (peak value), of which the default value is 90. You can reduce it and test the ring 

effect. If the phones still can not ring normally, please cut down the number of phones in 

parallel connection. 

8.6 Hear noise in ringback tone 

In some earlier version, SIP protocol, the call between 2 ports of one MG3000-R32, 

FXS/FXO, and the caller side may hear noise in ringback tone. This is because 

MG3000-R32 receive 183 SDP message, but the server do not send the color ring back 

tone signal, which cause some abnormal signals of remote side be sent to the caller side. 

This problem can be solved through configuring sending Color Ring Back Tone function in 

server, or updating MG3000-R32. 

8.7 Siemens Phone CID displays incorrectly 

Siemens 8000(3) P/TS D phone connects to FXS port of MG3000-R32. For the first call, it 

can ring and display correct Caller ID. But after several times, Caller ID can not be 

displayed. 

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This problem is because that Siemens phone will clear the Caller ID if the ring interval 

exceeds a set time, and wait for the next call. You can modify the FXS configuration of 

MG3000-R32 (refer to section 5.4.1): 

Advance Configuration ---> FXS ----> Ring Cadency, you can set it into: 

1000/3000,0/0,0/0,0/0 

8.8 IP route error, can not switch to PSTN 

Please check “Quick Setup” page, to find whether you have enabled “ IP switch to PSTN” 

function, because the default is disable. 

Then please check whether the product has available FXO ports. 

Information ---> Port Status ---> Status of Cards and Ports 

8.9 Establish a call in One-stage mode, but the call is 

interrupted very soon 

Under One-stage mode, in SIP protocol, when you start a call from IP-To-PSTN, and 

callee of PSTN side pick up the phone, the call is built. But it will be interrupted very soon, 

and the caller can only hear busy tone. 

If this problem happens, please check the configuration. If it complies with the 

following situation, FXO ports will be unable to answer the call, although early media 

channel has been built through 183 signals, and the call will be released for overtime: 

� PSTN line which connects to FXO port can not provide the reversal signal; 

� Advanced Configure ---> FXO ---> FXO Advanced Configuration ---> FXO 

Answer Supervision ---> After receiving Reversal Signal; 

� Basic Configure ---> Protocol ---> Other Configuration ---> FXO Provisional 

Response Type ---> choose 183. 

If the configuration is as the follows, even the callee at PSTN side pick up the phone, the 

caller will still hear the ring back tone, and the callee can not hear the voice either: 

� PSTN line connecting to FXO port does not provide reversal signal; 


Answer Supervision ---> After receiving Reversal Signal; 


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Response Type ---> choose 180. 


If the PSNT line connecting to FXO port does not provide reversal signal, you have to 

configure it according to the following steps, in order to make the product work normally: 


Answer Supervision ---> “Normal”; 


Response Type ---> choose 180 . 

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9 Appendix I VOIP Technical Terms 

Caller ID 


Caller ID: switch sends the caller information to callee, such as number, name, date and 

time etc.. 

Basic method: Switch of originating end send caller information to the switch of destination 

through inter-office command system , and the switch of destination will send the 

information to end user in FSK (Frequency-Shift Keying) or DTMF (Dual Tone 

Multi-Frequency). 

FSK has two sending ways: CID sending of on hook (Type 1), CID sending of off hook 

(Type 2). For type 1, it can be divided into: sending before the first ring, send between the 

first ring and the second ring. 

FSK CID modulation standard has two types: Bell202 and ITU-T V.23. ITU-T V.23 use 

frequency 1300±10Hz as 1, and frequency 2100±10Hz as 0; Bell202 use 1200±10Hz as 1, 

and 2200±10Hz as 0. The bit rates of both are 1200bps. 

FSK CID data has 2 types: SDMF (Single Data Message Format), MDMF (Multiple Data 

Message Format). For SDMF, the format structure is simple, and only a little information 

can be included (Caller ID, Date and Time); while for MDMF, the format structure is more 

complex, and it can include more information, besides what is included in SDMF, it can 

also include Caller ID name etc.. 

Currently, in these countries, such as USA, China, Japan, UK, Canada, Belgium, Spain, 

Singapore etc., FSK is adopted. However, the standards vary in different countries: North 

America and China adopt Bellcore standard; most of European countries adopt ETSI 

standard; Japan adopt NTT standard; UK adopt BT standard. Countries which adopt 

DTMF are India and Sweden etc.. 

FSK 

FSK (Frequency-shift keying) use digital signal to modulate the carrier frequency, which is 

applied very early and widely in data transmission because of its main advantages 

including: realizing easily, good performance in anti-noise and anti-attenuation. 

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DTMF 


DTMF (Dual Tone Multi-Frequency) converts the key events into the corresponding 

dual-tone multi-frequency signals, and each DTMF signal forms by overlapping two voice 

frequencies signals. According to ITU-T recommendations, the international adopted 

frequencies have 8 kinds: 697Hz, 770Hz, 852Hz, 941Hz, 1209Hz, 1336Hz, 1337Hz and 

1633Hz. In accordance with the “choosing 2 from 8” approach, the frequencies can have 

16 different combinations, representing 16 different number keys or function keys. 

As a fast and reliable telephone number transmission technology, DTMF has very strong 

anti-interference capability and high transmission speed. Besides this, it can also realize 

the long distance transmission of second dial - up number.。 

Table 37 DTMF COMBINATIONS 

1209Hz 1336Hz 1477Hz 1633Hz 

697Hz 1 2 3 A 

770Hz 4 5 6 B 

852Hz 7 8 9 C 

941Hz * 0 # D 

Silence Compression and CNG 

During general call, silence time will take up 50%. Silence Compression can detect line 

silence status automatically according to the threshold, and reduce the quantity of 

packages during silence period, so as to save the bandwidth effectively. 

However, Silence Compression may cause the incontinuous feeling for the receiver. The 

call quality can be improved by CNG in receiver side. 

iLBC 

ILBC is designed by Global IP Sound, the global famous voice engine provider, dedicated 

for packet switching network communication. The voice quality of ILBC is bettern than , at 

least equals to, G.729 and G.723.1 which are very popular now. It can realize very good 

voice quality even the packet loss rate is high. ILBC occupies very small bandwidth, and 

packs in 13.3kb/s (30ms/per frame) and 15.2kb/s (20ms/ per frame). For iLBC (30ms), the 

total bandwidth it occupies is smaller than that of G.729 (20ms) which is adopted normally. 

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RTP Redundancy 


RTP is based on UDP protocol, and does not provide data retransmission mechanism. 

VOIP often works on packet network of which the quality can not be guaranteed. So we 

need adopt some mechanism to decrease the impact caused by network packet loss, and 

improve the reliability. 

RTP Redundancy means combining voice data of current frame and a few frames before 

in RTP voice packet, so that when the packet loss happens, receiver can get related data 

in following packets. Then the loss of voice packets can be reconstructed and recovered, 

resolve the voice quality problems caused by packet loss. However, due to the packet size 

will be bigger than before for RTP redundant data, we must be careful that network 

congestion may occur and the packet loss will be even more serious. 

T.38 Fax 

ITU-T T.38 protocol defines the G3 fax transmission protocol basing on IP network. After 

the sending gateway receive the fax signal and message sent by sending fax machine, 

the fax signal and message will be demodulated into digital signal, packaged by IP Fax 

Protocol and sent; the receiving gateway will remodulate the IFP packets into fax signal 

and message and sent to destination fax machine. In order to reduce the transmission 

delay, T.38 general sends IFP packets by UDP, defines UDPTL redundancy and FEC 

error-correcting mechanism, to enhance the capability of anti packet loss. Therefore, T.38 

enhances the IP fax reliability. 

T.38 fax data transmission separated from voice data transmission, so it will not restrict 

voice quality and all voice coders can be used with T.38 fax together. 

Bypass Fax 

Bypass Fax has a high requirement on voice quality, so normally need G.711 as voice 

coder, which is non-compressing coder. Otherwise, bypass fax can not work normally. 

Under these circumstances, bypass fax provides a means of compromise: In voice 

communication, allowing the use of compressing coder, if one side starts the fax 

requirement, the fax gateway will initiate a flow to modify the media parameters when 

detecting the fax signal, change the coder into G.711, and combine the bypass fax signal. 

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Answer Supervision 


This function means we can monitor the answer situation of far end of PSTN far end 

through switch connected to PSTN, FXO of PBX, and digital E1/T1 interface, and process 

it correspondingly. Normally we have four ways to realize it: 

1) ISDN/PRI signal (T1/E1 digital signal): during the call, PSTN switch will inform 

the far end through the signal no matter the remote end answer or not. 

2) CAS-E&M signal (T1/E1 digital signal) : inform user of far end answer 

information through E&M signal. 

3) Reversal polarity (analog signal): For switch of PSTN, it will send the line 

reversal polarity signal to the user of near end, if the user of far end answer or 

release the call.由 PSTN. 

4) Line busy tone detection (analog signal): Detecting the line signal through the 

equipment of FXO port, product can know the user of far end answering or 

releasing the call according to the voice signal and type of signal. 

For FXO port, only 3) and 4) are available. Detecting line voice signal is not reliable. For 

example, if far end is unreachable, PSTN switch may provide IVR to caller, which will 

cause the wrong judgment that the user of far end has answered. 

Disconnect Supervision 

Disconnect Supervision function means you can monitor call release situation of far end of 

PSTN remotely, and process accordingly through FXO, E1/T1 of PSTN Switch and PBX. 

Just like remote Answer Supervision, Disconnect Supervision can be achieved through 

four ways: 

For FXO port, only 3) and 4) ways are available. For ordinary PSTN line, when one side 

releases the call, it will send busy signal to opposite side and prompt him to hang up. So 

normally line busy tone detection method is adopted to monitor the call release situation of 

remote site. If line busy tone can not be provided or detected, we can also get the call 

release information of far end through monitoring the polarity of line voltage during the call. 

If the polarity reverses again, then the call is released in the far end. 

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FXS、FXO、E&M Interface 

FXO, FXS, and E&M are all analog signal interfaces. 


FXS (Foreign Exchange Station), a voice interface, is used to connect digital telephone 

switch and POTS telephone. It simulates as PABX extension interface ( or user interface 

of Central Office), and can realize the connection between an ordinary telephone and a 

multiplexer. In simple words, FXS port is used to connect POTS ordinary telephone, and 

FAX machine. FXS port can provide remote feed. 

FXO (Foreign Exchange Office), also a voice interface, is used to connect central switch 

and digital telephone exchange system. For Central Office, it simulates as PABX 

extension, which can realize the connection between an ordinary telephone and a 

multiplexer. In simple words, FXO is used to connect PSTN line, and it needs remote feed. 

E&M interface is a private analog trunk interface of PBX, of which the information and 

connection signal are separated. It has two types: two line E&M and four line E&M. During 

interconnection of E&M, you must adopt the same dialing way, signal interface, and E line 

should be connected to the M line of opposite side. And it is necessary to configure a 

private E&M interface card for PBX. 

The difference between FXO and Lifeline 

For MG3000-R32, FXO supports IP-To-PSTN, PSTN-To-IP service, provides PSTN Route, 

Fall Back For IP Route Failure, Switch route by dialing special key, and Lifeline for power 

or network failure; 

Lifeline port only support Switch route by dialing special key and Lifeline for power failure. 

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