Synway SMG2030, SMG2060, SMG2120, SMG3008, SMG3016 User Manual

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Synway SMG Series Digital Gateway

SMG2030 SMG2060 SMG2120 SMG3008 SMG3016

Digital Gateway

Version 1.6.0

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SMG Series Digital Gateway User Manual (Version 1.6.0) Page i

Content

Content .................................................................................................. i

Revision History ..................................................................................... v

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

1.1 Typical Application ......................................................................................... 1

1.2 Feature List .................................................................................................... 2

1.3 Hardware Description .................................................................................... 3

1.4 Alarm Info ...................................................................................................... 5

Chapter 2 Quick Guide ...................................................................... 6

Chapter 3 WEB Configuration ......................................................... 12

3.1 System Login ............................................................................................... 12

3.2 Operation Info .............................................................................................. 13

3.2.1 System Info ............................................................................................................. 13

3.2.2 PSTN Status ........................................................................................................... 15

3.2.3 SS7 Server ............................................................................................................. 18

3.2.4 Call Count ............................................................................................................... 22

3.3 VoIP Settings ............................................................................................... 23

3.3.1 SIP Settings ............................................................................................................ 24

3.3.2 SIP Trunk ................................................................................................................ 26

3.3.3 SIP Register ............................................................................................................ 27

3.3.4 SIP Account ............................................................................................................ 30

3.3.5 SIP Trunk Group ..................................................................................................... 32

3.3.6 Media Settings ........................................................................................................ 35

3.4 PCM Settings ............................................................................................... 37

3.4.1 PSTN ...................................................................................................................... 38

3.4.2 Circuit Maintenance ................................................................................................ 39

3.4.3 PCM ................................................................................................ ........................ 39

3.4.4 PCM Trunk .............................................................................................................. 41

3.4.5 PCM Trunk Group ................................................................................................... 44

3.4.6 Number-receiving Rule ................................................................ ........................... 46

3.4.7 Reception Timeout .................................................................................................. 48

3.4.8 Number Attribution .................................................................................................. 50

3.5 SS7 Settings ................................................................................................ 50

3.5.1 SS7 ......................................................................................................................... 51

3.5.2 TUP ........................................................................................................................ 52

3.5.3 TUP Number Parameter ......................................................................................... 53

3.5.4 ISUP ....................................................................................................................... 55

3.5.5 ISUP Number Parameter ........................................................................................ 57

3.5.6 Original CalleeID Pool ............................................................................................ 60

3.5.7 Redirecting Number Pool ........................................................................................ 61

3.5.8 SS7 Server ............................................................................................................. 63

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3.6 ISDN Settings .............................................................................................. 69

3.6.1 ISDN ....................................................................................................................... 70

3.6.2 Number Parameter ................................................................................................. 72

3.6.3 Redirecting Number ................................................................................................ 73

3.6.4 Add Gateway .......................................................................................................... 73

3.7 SS1 Settings ................................................................................................ 75

3.8 Fax Settings ................................................................................................. 76

3.8.1 Fax .......................................................................................................................... 77

3.9 Route Settings ............................................................................................. 78

3.9.1 Routing Parameters ................................................................................................ 79

3.9.2 IP to PSTN .............................................................................................................. 79

3.9.3 PSTN to IP .............................................................................................................. 81

3.10 Number Filter ............................................................................................... 82

3.10.1 Whitelist .................................................................................................................. 83

3.10.2 Blacklist .................................................................................................................. 85

3.10.3 Number Pool ........................................................................................................... 86

3.10.4 Filtering Rule ........................................................................................................... 87

3.11 Number Manipulation ................................................................................... 90

3.11.1 IP to PSTN CallerID ................................................................................................ 90

3.11.2 IP to PSTN CalleeID ............................................................................................... 93

3.11.3 IP to PSTN Original CalleeID .................................................................................. 94

3.11.4 PSTN to IP CallerID ................................................................................................ 94

3.11.5 PSTN to IP CalleeID ............................................................................................... 97

3.11.6 PSTN to IP Original CalleeID .................................................................................. 98

3.11.7 CallerID Pool ........................................................................................................... 98

3.12 System Tools ............................................................................................... 99

3.12.1 Network ................................................................................................................ 101

3.12.2 Management ......................................................................................................... 102

3.12.3 SNMP Config ........................................................................................................ 103

3.12.4 Radius .................................................................................................................. 104

3.12.5 Configuration File .................................................................................................. 106

3.12.6 Signaling Capture ................................................................................................. 107

3.12.7 Signaling Call Test ................................................................ ................................ 108

3.12.8 Signaling Call Track .............................................................................................. 109

3.12.9 PING Test ............................................................................................................. 110

3.12.10 TRACERT Test ..................................................................................................... 111

3.12.11 Modification Record .............................................................................................. 112

3.12.12 Backup & Upload .................................................................................................. 113

3.12.13 Factory Reset ....................................................................................................... 113

3.12.14 Upgrade ................................................................................................................ 113

3.12.15 Change Password ................................................................................................ 114

3.12.16 Device Lock .......................................................................................................... 114

3.12.17 Restart ................................................................................................ .................. 115

Chapter 4 Typical Applications ..................................................... 116

4.1 Application 1 ............................................................................................... 116

4.1.1 Configurations for Headquarters ........................................................................... 117

4.1.2 Configurations for Branch A .................................................................................. 119

4.1.3 Configurations for Branch B .................................................................................. 122

4.2 Application 2 .............................................................................................. 126

4.2.1 Configurations for Headquarters ........................................................................... 126

4.2.2 Configurations for Branches ................................................................................. 129

Appendix A Technical Specifications ............................................... 130

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Appendix B Troubleshooting ............................................................ 131

Appendix C ISUP (ISDN) Pending Cause to SIP Status Code ........ 132

Appendix D TUP Pending Cause to SIP Status Code ..................... 134

Appendix E Technical/sales Support ............................................... 135

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All rights reserved; no part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, without prior written permission from Synway Information Engineering Co., Ltd (hereinafter referred to as „Synway‟).

Synway reserves all rights to modify this document without prior notice. Please contact Synway for the latest version of this document before placing an order.

Synway has made every effort to ensure the accuracy of this document but does not guarantee the absence of errors. Moreover, Synway assumes no responsibility in obtaining permission and authorization of any third party patent, copyright or product involved in relation to the use of this

document.

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SMG Series Digital Gateway User Manual (Version 1.6.0) Page v

Revision History

Version

Date

Comments

Version 1.3.0

2014-06

Initial publication.

Version 1.3.1

2014-08

New revision

Version 1.3.2

2014-10

New revision

Version 1.5.0

2014-12

Add description on the new series SMG3016

Version 1.5.1

2015-01

Add description on the new series SMG3008

Version 1.6.0

2015-03

New revision

Note: Please visit our website http://www.synway.net to obtain the latest version of this document.

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SMG Series Digital Gateway User Manual (Version 1.6.0) Page 1

Chapter 1 Product Introduction

Thank you for choosing Synway SMG Series Digital Gateway! The Synway SMG series digital gateway products (hereinafter referred to as ‘SMG digital

gateway’) are mainly used for connecting PSTN or enterprise PBX with the IP telephony network or IP PBX. It provides a powerful, reliable and cost-effective VoIP solution for such occasions as IP call centers and multi-branch agencies.

The SMG series digital gateway has five models:

 SMG2030: 1 E1/T1 interface (30 digital ports)  SMG2060: 2 E1/T1 interfaces (60 digital ports)  SMG2120: 4 E1/T1 interfaces (120 digital ports)  SMG3008: 8 E1/T1 interfaces (240 digital ports)  SMG3016: 16 E1/T1 interfaces (480 digital ports)

1.1 Typical Application

Branch B

Subscriber

Terminal

IP Phone IP Phone

SMG Digital

Gateway

LAN

Router

IP PBX

Headquarters

SMG Digital Gateway

Router

FXS

PBX

Tel Tel Tel

SIP

Branch A

SMG Digital Gateway

SIP

FXS

PBX

Tel Tel Tel

Router

Internet

Tel Tel Fax Fax

……

PSTN

Figure 1-1 Typical Application

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1.2 Feature List

Basic Features

Description

PSTN Call

Call initiated from PSTN to a designated SIP trunk, via routing and number manipulation.

IP Call

Call initiated from IP to a designated PCM trunk, via routing and number manipulation.

Number Manipulation

Peels off some digits of a phone number from left/right, or adds a prefix/suffix to a phone number.

PSTN/ VoIP Routing

Routing path: from IP to PSTN or from PSTN to IP.

Fax

Multiple fax parameters: fax mode, maximum fax rate, fax train mode, error correction mode, etc.

Echo Cancellation

Provides the echo cancellation feature for a call conversation.

Signaling & Protocol

Description

SS7

SS7-TUP, SS7-ISUP

ISDN

ISDN User Side, ISDN Network Side

SS1

SS1 Signaling

SIP Signaling

Supported protocol: SIP V1.0/2.0, RFC3261

Voice

CODEC

G.711A, G.711U, G.729A/B, G723, G722, AMR, iLBC

DTMF Mode

RFC2833, SIP INFO, INBAND

Fax

Fax Mode

T.38, Pass-Through

Baud Rate

14400bps, 9600bps, 4800bps

Network

Description

Network Protocol

Supported protocol: TCP/UDP, HTTP, ARP/RARP, DNS, NTP, TFTP, TELNET, STUN

Static IP

IP address modification support

DNS

Domain Name Service support

Security

Description

Admin Authentication

Support admin authentication to guarantee the resource and data security

Maintain & Upgrade

Description

WEB Configuration

Support of configurations through the WEB user interface

Language

Chinese, English

Software Upgrade

Support of user interface, gateway service, kernel and firmware upgrades based on WEB

Tracking Test

Support of Ping and Tracert tests based on WEB

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SysLog Type

Three options available: ERROR, WARNING, INFO

1.3 Hardware Description

The SMG digital gateway features 1U rackmount design and integrates embedded LINUX system within the POWERPC+DSP hardware architecture. It has 1/2/4/8/16 E1/T1 ports and 2 Kilomega-Ethernet ports (LAN1 and LAN2) on the chassis.

（a） See below figures for SMG20000 series appearance：

Figure 1-2 Front View

Figure 1-3 Rear View

Figure 1-4 Left View

（b） See below figures for SMG3000 series appearance：

Figure 1-5 Front View

Ventilation

Holes

Screw Holes for

Foot Bracket

Power1 Key

220V AC

Power1

220V AC

Power2

Power2 Key

E1/T1

Interface

LAN

Grounding

Stud

LAN1 Indicator

Alarm Indicator

Power 1 Indicator

LAN2 Indicator

Console Port

Run Indicator

PCM Indicator

Power 2 Indicator

Reset Button

LAN1 Indicator

Alarm Indicator

Power 1 Indicator

LAN2 Indicator

Console Port

Run Indicator

PCM Indicator

Power 2 Indicator

Reset Button

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Figure 1-6 Rear View

Note: The left view for SMG3000 series is same as that for SMG2000 series, refer to Figure 1-4. The table below gives a detailed introduction to the interfaces, buttons and LEDs illustrated

above:

Interface

Description

LAN

Amount: 2

Type: RJ-45

Bandwidth: 10/100/1000Mbps

Self-Adaptive Bandwidth Supported

Auto MDI/MDIX Supported

E1/T1

Amount: 1/2/4/8/16

Type: RJ-45

Console Port

Amount: 1

Type: RS-232

Baud Rate: 115200 bps

Connector: RJ45 (See Figure 1-7 for signal definition)

Data Bits: 8 bits

Stop Bit: 1 bit

Parity Unsupported

Flow Control Unsupported

Button

Description

Power Key

Power on/off the SMG digital gateway. You can turn on the two power keys at the same time to have the power supply working in the hot-backup mode.

Reset Button

Restore the gateway to factory settings.

LED

Description

Power Indicator

Indicates the power state. It lights up when the gateway starts up with the power cord well connected.

Run Indicator

Indicates the running status. For more details, refer to 1.4 Alarm Info.

Alarm Indicator

Alarms the device malfunction. For more details, refer to 1.4 Alarm Info.

Link Indicator

The green LED on the left of LAN, indicating the network connection status.

ACT Indicator

The orange LED on the right of LAN, whose flashing tells data are being transmitted.

E1/T1 Indicators

The green LED on the right of E1/T1 interface lights up and keeps on after the E1/T1 module is successfully synchronized.

Power1 Key

220V AC

Power1

220V AC

Power2

Power2 Key

E1/T1

Interface

LAN

Grounding

Stud

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Channel Indicators

Indicates the synchronization status of E1/T1 channels. It will light up and keep on if E1/T1 is synchronized; otherwise, it will go out.

Note: The console port is used for debugging. While connection, the transmitting and receiving lines of the gateway and the remote device should be cross-linked. That is, connect the transmitting line of the gateway to the receiving line of the remote device, and vice verse. The figure below illustrates the signal definition of the console port on the gateway.

Figure 1-7 Console Port Signal Definition

For other hardware parameters, refer to Appendix A Technical Specifications.

1.4 Alarm Info

The SMG digital gateway is equipped with two indicators denoting the system‟s running status: Run Indicator (green) and Alarm Indicator (red). The table below explains the states and meanings of the two indicators.

LED

State

Description

Run Indicator

Go out

System is not yet started.

Light up

System is starting.

Flash

Device is running normally.

Alarm Indicator

Go out

Device is working normally.

Light up

Upon startup: Device is running normally. In runtime: Device goes abnormal.

Flash

System is abnormal.

Note:

 The startup process consists of two stages: System Booting and Gateway Service

Startup. The system booting costs about 1 minute and once it succeeds, both the run indicator and the alarm indicator light up. Then after the gateway service is successfully started and the device begins to work normally, the run indicator flashes and the alarm indicator goes out.

 During runtime, if the alarm indicator lights up or flashes, it indicates that the device goes

abnormal. If you cannot figure out and solve the problem by yourself, please contact our technicians for help. Go to Appendix E Technical/sales Support to find the contact way.

8 7 6 5 4 3 2 1

GND

Gateway TX

Gateway RX

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Chapter 2 Quick Guide

This chapter is intended to help you grasp the basic operations of the SMG digital gateway in the shortest time.

Step 1: Confirm that your packing box contains all the following things.

 SMG Series Digital Gateway *1  Angle Bracket *2, Rubber Foot Pad *4, Screw for Angle Bracket *8  220V Power Cord *2  Warranty Card *1  Installation Manual *1

Step 2: Properly fix the SMG digital gateway.

If you do not need to place the gateway on the rack, simply fix the 4 rubber foot pads. Otherwise, you should first fix the 2 angle brackets onto the chassis and then place the chassis on the rack.

Step 3: Connect the power cord.

Make sure the device is well grounded before you connect the power cord. Check if the power socket has the ground wire. If it doesn‟t, use the grounding stud on the rear panel of the device (See Figure 1-3) for earthing.

Note: Each SMG digital gateway has two power interfaces to meet the requirement for power supply hot backup. As long as you properly connect and turn on these two power keys, either power supply can guarantee the normal operation of the gateway even if the other fails.

Step 4: Connect the network cable. Step 5: Connect the E1/T1 trunk. Connect the E1/T1 interface of the digital gateway to that

of the remote device by E1/T1 trunk. After connection, check if the synchronization indicator (green LED) is lit and keeps on, which indicates that the E1/T1 trunk is well connected and the E1/T1 module is successfully synchronized.

For the 75Ω-unbalanced coaxial cable, in consideration of various line conditions, each PCM on the digital gateway is equipped with two grounding jumpers which respectively control the grounding of the transmitting and the receiving end. Under normal condition, that is, the chassis of the gateway is well grounded, the grounding jumpers at the receiving end should be disconnected and the ones at the transmitting end should be short-circuited. This configuration is the factory default setting and applicable in most situations so that there is usually no need to change it. For the 120Ω-balanced twisted pair cable, the grounding jumpers at both ends should be disconnected.

You can construct an E1 trunk according to Figure 2-1. Prevent reverse connection of the transmitting and receiving lines. The state of the receiving line can be checked by the synchronization indicator (green LED) of the E1 interface. When the receiving line is in a normal state, the indicator is lit and keeps on. If the indicator is off or flashing, it means that the connection of the receiving line may probably be reversed. However, the state of the transmitting line can only be examined by the opposite terminal. The synchronization indicator starts working only after the device is powered on and successfully initialized.

8 7 6 5 4 3 2 1

RTip RRing

Receive

TTip TRing

Transmit

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Figure 2-1 Pin Layout for E1 Interface

Step 6: Log in the gateway.

Enter the original IP address (LAN 1: 192.168.1.101 or LAN 2: 192.168.0.101) of the SMG digital gateway in the browser to go to the WEB interface. The original username and password of the gateway are both „admin‟. For detailed instructions about login, refer to 3.1 System Login. We suggest you change the initial username and password via „System Tools  Change Password‟ on the WEB interface as soon as possible after your first login. For detailed instructions about changing the password, refer to 3.12.15 Change Password. After changing the password, you are required to log in again.

Step 7: Modify IP address of the gateway. You can modify the IP address of the gateway via „System Tools  Network‟ on the WEB interface

to put it within your company‟s LAN. Refer to 3.12.1 Network for detailed instructions about IP modification. After changing the IP address, you shall log in the gateway again using your new IP address.

Step8: Set PCM.

On your initial use of the SMG digital gateway, you shall enter the PCM interface and set the configuration items „Signaling Protocol‟ and „Interface‟. These items must be in conformity with the physical connection. You may use the default values of other configuration items. Refer to 3.4.3

PCM for detailed instructions about PCM Settings.

Note: You shall restart the service to validate the settings in this step. Refer to 3.12.17 Restart for detailed instructions.

Step 9: Configure signaling protocol parameters.

Further configure the signaling protocol you set in Step 8. Different protocols are configured on different interfaces. See below for detailed instructions.

 SS7-ISUP: Note: For your easy understanding and manipulation, this step does not involve the ISUP

quasi-associated mode configuration and the dual gateway feature. For descriptions about these configurations, refer to 3.5 SS7 Settings.

The configuration interfaces related to SS7-ISUP include: SS7, ISUP and SS7 Server. On your initial use of the SMG digital gateway, you may adopt the default values of the

configuration items on the SS7 and ISUP interfaces. Note that the SS7 Server interface must be configured properly. Otherwise, the PSTN trunks may be unavailable. Follow the instructions here to configure the SS7 Server:

Step 1: Set OPC, Server IP and Signaling Point Code Standard. The OPC is generally allocated

by the central office. The Server IP is the IP address of the SS7 server and you may use its default value. The Signaling Point Code Standard, which varies on the PBX model, can be set to 24 or 14. After modification, click the ‘Modify’ button on the right to save the settings.

Step 2: Modify the current link or click the „Add New‟ button below the signaling link list to add a

new link. Enter the physical address of the actually used signaling PCM (E1 interface) and click „Save‟ to save the modification. If only one PCM is used for signaling in the gateway, you need just configure one signaling link.

Step 3: Modify the current linkset or click the „Add New‟ button below the signaling linkset list to

add a new linkset. You shall select the link configured in Step 2 for „Link‟ and use the default values for the other configuration items. After modification, click ‘Save’.

Step 4: Modify the current DPC or click the „Add New‟ button below the DPC list to add a new

DPC. Fill in „SP Code‟ with the signaling point code of the remote end (i.e. signaling destination), select the linkset configured in Step 3 for „Linkset‟ and use the default values for the other configuration items. After modification, click ‘Save’.

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Step 5: Modify the current CIC routing rule or click the „Add New‟ button below the ISUP_CIC

routing rule list to add a new CIC routing rule. Select the DPC configured in Step 4 for „DPC‟, fill in „CIC_PCM‟ according to the actual allocation and use the default values for the other configuration items. After modification, click ‘Save’. Note that if multiple PCMs in the gateway are used for voice transmission, they should be configured with multiple CIC routing rules accordingly.

Note: After configuring SS7-ISUP related interfaces, you shall restart the service to validate the settings. Refer to 3.12.17 Restart for detailed instructions.

 SS7-TUP: Note: For your easy understanding and manipulation, this step does not involve the TUP

quasi-associated mode configuration and the dual gateway feature. For descriptions about these configurations, refer to 3.5 SS7 Settings.

The configuration interfaces related to SS7-TUP include: SS7, TUP and SS7 Server. On your initial use of the SMG digital gateway, you may adopt the default value of the

configuration items on the SS7 and TUP interfaces. Note that the SS7 Server interface must be configured properly. Otherwise, the PSTN trunks may be unavailable. Follow the instructions here to configure the SS7 Server:

Step 1: Set OPC, Server IP and Signaling Point Code Standard. The OPC is generally allocated

Step 2: Modify the current link or click the „Add New‟ button below the signaling link list to add a

Step 3: Modify the current linkset or click the „Add New‟ button below the signaling linkset list to

add a new linkset. You shall select the link configured in Step 2 for „Link‟ and use the default values for the other configuration items. After modification, click ‘Save’.

Step 4: Modify the current DPC or click the „Add New‟ button below the DPC list to add a new

Step 5: Modify the current CIC routing rule or click the „Add New‟ button below the TUP_CIC

Note: After configuring SS7-TUP related interfaces, you shall restart the service to validate the settings. Refer to 3.12.17 Restart for detailed instructions.

 ISDN User Side/Network Side: The configuration interface related to ISDN User Side/Network Side is ISDN. On your initial use of

the SMG digital gateway, you may adopt the default value of the configuration items on this interface.

Note: After configuring the ISDN interface, you shall restart the service to validate the settings. Refer to 3.12.17 Restart for detailed instructions.

 SS1: The configuration interface related to SS1 is SS1. On your initial use of the SMG digital gateway,

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you may adopt the default value of the configuration items on this interface. Note: After configuring the SS1 interface, you shall restart the service to validate the settings.

Refer to 3.12.17 Restart for detailed instructions.

Step 10: Check the PSTN status.

After the configuration of signaling protocols, you can check the status of the PSTN trunks via „Operation Info  PSTN Status‟. Refer to 3.2.2 PSTN Status for detailed introductions. When Time Slot 0 shows „Frame Synchronized‟, the signaling time slot is in the state of „Signaling Channel‟ and all the other channels are „Idle‟, it indicates the PCM is well configured. If Time Slot 0 or the signaling time slot shows „Faulty‟ or the other channels are in the state of „Unavailable‟, there may be errors in the signaling protocol configurations and we suggest you return to Step 9 for check.

Step 11: Set routing rules for calls.

Note: For your easy understanding and manipulation, all examples given in this step do not involve registration.

Situation 1: IP  PSTN

Step 1: Configure the IP address of the remote SIP terminal which can establish conversations

with the gateway so that the calls from other terminals will be ignored. Refer to „VOIP Settings  SIP Trunk‟ for detailed instructions. Fill in „Remote IP‟ and „Remote Port‟ with the IP address and port of the remote SIP terminal which will initiate calls to the gateway. You may use the default values for the other configuration items.

Example: Provided the IP address of the remote SIP terminal is 192.168.0.111 and the port is 5060. Add SIP Trunk 0; set Remote IP to 192.168.0.111 and Remote Port to

5060.

Step 2: Add the IP address of the remote SIP terminal configured in Step 1 into the

corresponding SIP trunk group. Refer to „VoIP Settings  SIP Trunk Group‟ for detailed instructions. Select the SIP trunk configured in Step 1 as „SIP Trunks‟. You may use the default values for the other configuration items.

Example: Add SIP Trunk Group 0. Check the checkbox before 0 for SIP Trunks and keep the default values for the other configuration items.

Step 3: Add PCM into the corresponding PCM Group. Refer to „PCM Settings  PCM Trunk

Group‟ for detailed instructions. Select the PCM used for call conversation as „PCM‟. You

may use the default values for the other configuration items.

Example: Provided the PCM used for call conversation is PCM[1]. Add PCM Trunk Group 0, check the checkbox before PCM[1] and keep the default values for the other

configuration items.

Step 4: Add routing rules. Refer to „Route Settings  IPPSTN‟ for detailed instructions. Select

the SIP trunk group set in Step 2 as „Call Initiator‟ and the PCM trunk group set in Step 3 as „Call Destination‟. You may use the default values for the other configuration items.

Example: Select SIP Trunk Group[0] as Call Initiator and PCM Trunk Group[0] as Call Destination. Keep the default values for the other configuration items.

Step 5: Initiate a call from the SIP terminal configured in Step 1 to the IP address and port of the

SMG digital gateway. Thus you can establish a call conversation via PCM[1] with the PSTN terminal. (Note: The format used for calling an IP address via SIP trunk is as follows: username@IP address, in which, „username‟ is a called party number which conforms to the number-receiving rule of the remote device.)

Example: Provided the IP address of the SMG digital gateway is 192.168.0.101 and the port is 5060. Provided 123 is a number which conforms to the number receiving rule of the remote device. Initiate a call from SIP terminal 0 to the IP address 192.168.0.101 (in the format: 123@192.168.0.101) and you can establish a call conversation via PCM[1] to

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the number 123. Situation 2: PSTN  IP Step 1: Configure the called party numbers which are received from PSTN and will be processed

by the gateway. Refer to „Advanced Settings  Number-receiving Rule‟ for detailed instructions. Enter either a particular number or a string of „x‟s to represent several random numbers. For example, „xxx‟ denotes 3 random numbers. You may use the default value for „Index‟.

Example: Set Index to 99 and configure Dial Rule to 123. Step 2: Set the IP address of the SIP terminal to be called by the gateway. Refer to „VOIP

Settings  SIP Trunk‟ for detailed instructions. Fill in „Remote IP‟ and „Remote Port‟ with the IP address and port of the SIP trunk. You may use the default values for the other configuration items.

Example: Provided the IP address of the SIP trunk to be called is 192.168.0.111 and the

port is 5060. Add SIP Trunk 0; set Remote IP to 192.168.0.111 and Remote Port to

5060.

Step 3: Add the IP address of the remote SIP terminal configured in Step 2 into the

corresponding SIP trunk group. Refer to „VoIP Settings  SIP Trunk Group‟ for detailed instructions. Select the SIP trunk configured in Step 2 as „SIP Trunks‟. You may use the default values for the other configuration items.

Example: Add SIP Trunk Group 0. Check the checkbox before 0 for SIP Trunks and

keep the default values for the other configuration items. Step 4: Add PCM into the corresponding PCM Group. Refer to „PCM Settings  PCM Trunk

Group‟ for detailed instructions. Select the PCM used for call conversation as „PCM‟. You

may use the default values for the other configuration items.

Example: Provided the PCM used for call conversation is PCM[1]. Add PCM Trunk

Group 0, check the checkbox before PCM[1] and keep the default values for the other

configuration items. Step 5: Add routing rules. Refer to „Route Settings  PSTNIP‟ for detailed instructions. Select

the PCM trunk group set in Step 4 as „Call Initiator‟ and the SIP trunk group set in Step 3 as „Call Destination‟. You may use the default values for the other configuration items.

Example: Select PCM Trunk Group[0] as Call Initiator and SIP Trunk Group[0] as

Call Destination. Keep the default values for the other configuration items.

Step 6: Once PCM[1] receives a call from PSTN and the called party number conforms to the

number-receiving rules set in Step 1, it can establish a call conversation with the remote SIP terminal via the gateway.

Example: Once PCM[1] receives a call from PSTN with the called party number 123, it

will route the call to SIP Trunk 0 of the gateway.

Special Instructions:

 The chassis of the SMG digital gateway must be grounded for safety reasons, according

to standard industry requirements. A simple way is earthing with the third pin on the plug

or the grounding studs on the machine. No or improper grounding may cause instability

in operation as well as decrease in lightning resistance.

 As the device will gradually heat up while being used, please maintain good ventilation to

prevent sudden failure, ensuring that the ventilation holes (see Figure 1-4) are never

jammed.

 During runtime, if the alarm indicator lights up or flashes, it indicates that the device goes

abnormal. If you cannot figure out and solve the problem by yourself, please contact our

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technicians for help. Otherwise it may lead to a drop in performance or unexpected

errors.

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Chapter 3 WEB Configuration

3.1 System Login

Type the IP address into the browser and enter the login interface. See Figure 3-1.

Figure 3-1 Login Interface

The gateway only serves one user, whose original username and password are both „admin‟. You can change the username and the password via „System Tools  Change Password‟ on the WEB interface. For detailed instructions, refer to 3.12.15 Change Password.

After login, you can see the main interface as below.

Figure 3-2 Main Interface

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3.2 Operation Info

Operation Info includes four parts: System Info, PSTN Status, SS7 Server and Call Count, showing the current running status of the gateway. See Figure 3-3.

Figure 3-3 Operation Info

3.2.1 System Info

Figure 3-4 System Info Interface

See Figure 3-4 for the system info interface. You can click Refresh to obtain the latest system

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information. The table below explains the items shown in Figure 3-4.

Item

Description

MAC Address

MAC address of LAN 1 or LAN 2.

IP Address

The three parameters from left to right are IP address, subnet mask and default gateway of LAN 1 or LAN 2.

DNS Server

DNS server address of LAN 1 or LAN 2.

Receive Packets

The amount of receive packets after the gateway‟s startup, including three categories: All, Error and Drop.

Transmit Packets

The amount of transmit packets after the gateway‟s startup, including three categories: All, Error and Drop.

Current Speed

The current speed of data receiving and transmitting.

Work Mode

The work mode of the network, including five options: 10 Mbps Half Duplex, 10 Mbps Full Duplex, 100 Mbps Half Duplex, 100 Mbps Full Duplex and 1000 Mbps Full Duplex.

Runtime

Time of the gateway keeping running normally after startup. This parameter updates every 2s.

Operating Mode

The operating mode of the gateway includes:

Operating Mode

Description

Master Server

The current gateway applies the SS7 protocol and is used for both signaling and voice transmission. If the dual gateway feature is enabled, the current gateway serves as the master server.

Slave Server

The current gateway applies the SS7 protocol and is used for both signaling and voice transmission. This operating mode works only when the dual gateway feature is enabled and the current gateway serves as the slave server.

Client

The current gateway applies the SS7 protocol and is only used for voice transmission.

ISDN(User-side)

The current gateway is configured to be ISDN user-side..

ISDN(Network-side)

The current gateway is configured to be ISDN network-side.

SS1

The current gateway is configured to be SS1.

Serial Number

Unique serial number of an SMG digital gateway.

WEB

Current version of the WEB interface.

Gateway

Current version of the gateway service.

Uboot

Current version of Uboot.

Kernel

Current version of the system kernel on the gateway.

Firmware

Current version of the firmware on the gateway.

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3.2.2 PSTN Status

Figure 3-5 PSTN Status Interface for E1 Lines

Figure 3-6 PSTN Status Interface for T1 Lines

See Figure 3-5 and Figure 3-6 for the PSTN status interface which shows the real-time status of each PCM on the gateway, including line synchronization, signaling link information and channel states.

Item

Description

Port

Serial number of the E1/T1 port on the device.

Time Slot No.

PCM time slot number in the port.

State

Displays the channel state in real time. You can move the mouse onto the channel state icon for detailed information about the channel and the call, such as: call direction, calling party number and called party number.  For Time Slot 0, the channel state indicates the synchronization status of

E1/T1.

State

Color

Description

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Frame Sync

Frame synchronization normal. The synchronization status is 0x0.

Faulty

Configuration errors or hardware failure. You can move the mouse onto the icon for the hexadecimal value for synchronization status which consists of 16 bits and bit 0 is the lowest valid bit. If the bit value is equal to 0, it indicates that the synchronization status is normal; if the bit value is equal to 1, see below for details: bit0=1: basic frame synchronization loss bit1=1: duration of the basic frame synchronization loss exceeds 100ms bit2=1: CAS re-synchronization bit3=1: CRC re-synchronization bit4=1: remote alarm indication bit5=1: signal alarm indication bit6=1: all-ones alarm signal of time slot 16 bit7=1: signal loss bit9=1: MF alarm from the remote end bit10=1: open circuit bit11=1: short circuit Other bits: reserved, all remain 0

 For the signaling time slot, the channel states include:

State

Color

Description

Signaling

For SS7, this state indicates „SS7 in service‟. For ISDN, this state indicates „multiple frames established‟ or „timer recovery‟. For SS1, this state indicates „time slot synchronization normal‟.

Faulty

Configuration errors or hardware failure. For SS7, this state indicates „SS7 out of service‟, „initial alignment‟, „aligned ready‟, „aligned not ready‟ or „processor outage‟. For ISDN, this state indicates „TEI unassigned‟, „assign awaiting TEI‟, „establish awaiting TEI‟, „TEI assigned‟, „awaiting establishment „or „awaiting release‟. For SS1, this state indicates „time slot synchronization abnormal‟.

Unused

This state indicates the signaling time slot on this E1/T1 is not used.

 For the other channels, the channel states include:

State

Icon

Description

Unusable

The channel is unavailable.

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Circuit Reset

The circuit is being reset.

Idle The channel is available.

Local Block

The channel is blocked by the local application program and cannot receive incoming calls.

Remote Block

The channel is blocked by the specific circuit/circuit group blocking messages sent from the remote PBX and cannot make outgoing calls.

Both Block

The channel is blocked by the local end so as not to receive incoming calls, meanwhile, it is blocked by the remote PBX so as not to make outgoing calls either.

Wait Answer

The channel receives the ringback tone and is waiting for the called party to pick up the phone.

Ringing

The channel is in the ringing state.

Talking

The channel is in a conversation.

Pending

The channel is in the pending state

Dialing

The channel is dialing.

Wait Message

The channel is waiting for the message from remote PBX.

Statistics

The total amount of the channels for the corresponding status.

Note: The gateway provides the fuzzy search feature on this interface. After you click any characters on Figure 3-5, Figure 3-6, and press the „F‟ button, the search box will emerge on the right top of this page. Then you can input the key characters and the gateway will locate the channel on which there is an ongoing call that conforms to the fuzzy search condition.

Take an example: As shown in Figure 3-7, after we input the character 888 to the search box, and click the Search button, the gateway does a fuzzy search and locates that the ongoing call whose CallerID contains the character 888 occurs on Channel 9.

Figure 3-7 Search Calls

Note: Click Record to start recording on the matched channel. If more than one channels match a condition, only the channel with the largest number among them will be recorded.

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3.2.3 SS7 Server

Users can see the SS7 Server option in the menu only when the configuration item Signaling Protocol on the PCM settings interface is set to SS7-TUP or SS7-ISUP.

Figure 3-8 SS7 Server Info Interface

See Figure 3-8 for the SS7 server info interface. This interface contains 7 status bars (Status Bar 1~7 in the above figure) and a configuration region (Region 8 in the above figure). Below are the detailed introductions.

 Status Bar 1 & 2: Receive/transmit message list The receive/transmit message lists display the received and sent messages respectively, used for

gateway debugging. The display content in these lists can be set by the configuration items in Region 8.

 Configuration Region 8: Properties configuration for receive/transmit message list The table below explains the items in Configuration Region 8.

Item

Description

Server IP

IP address of the SS7 server, this item can be configured on the SS7 interface.

Work Mode

Work mode of the SS7 server which includes three modes: Master Server, Slave Server and Client.

Display Sent Messages

If this item is ticked, the transmit message list will display the message sent to the remote end.

Display Received Messages

If this item is ticked, the receive message list will display the message received from the remote end.

Display DPC & OPC

If this item is ticked, the receive/transmit message list will display DPC and OPC.

Display SNT Messages

If this item is ticked, the receive/transmit message list will display the SNT messages.

Display SNM Messages

If this item is ticked, the receive/transmit message list will display the SNM messages.

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Auto Translate

If this item is ticked, the received/sent messages displayed on this interface will be translated automatically in the following format:

Date Time Total number Signaling link number# SIO Content For the TUP messages, SIO is just „TUP‟ (0x84), followed by the message content. It is usually in the following format:

Title code CIC=PCM:TS Message body If this item is not ticked, the received/sent messages displayed on this interface will be hexadecimal raw data.

Users can configure the display content of the receive/transmit message list via the checkbox before each configuration item. After modification, click Save to apply the configurations. The changes will be shown in the list in real time. Click Download and you can download the log information of the SS7 server.

 Status Bar 3: Linkset/signaling link information This region displays the information about signaling links and linksets. The table below explains

the information items in Status Bar 3.

Item

Description

Linkset No.

Linkset number.

Linkset Status

Working state of the linkset, including In service and Out of service. A signaling linkset will go into the state In service as long as one link in it is at the state of In service.

Link@Physical Address

Signaling link number and its physical position. For example, „0 @ IP[0]:PCM[0]‟ means the physical position of Link 0 in this gateway is the E1 with the local PCM numbered 0 on Client 0.

Network

Whether the signaling link is registered to the gateway, including two states: Connected and Disconnected (or no display). The signaling link can be used normally only in the state of Connected.

Sync

Basic frame synchronization (Time Slot 0), including two states: Sync and Async. The signaling link can be used only in the state of Sync.

Link Status

Working state of the signaling link, including In service and Initial alignment. You can refer to „Status Bar 6: Link information‟ for detailed information about link status.

Hold Time

Duration since the last time the signaling link enters into the state of In service.

Position

Times of positioning that occurs on the signaling link since the program starts.

Send MSU

Total number of messages sent on the signaling link since the program starts.

Receive MSU

Total number of messages received on the signaling link after the program starts.

Switchover MSU

Total number of messages switched over on the signaling link since the program starts.

 Status Bar 4: Client information This region displays the information about client IP address and connection state. The table below

explains the information items in Status Bar 4.

Item

Description

Client

Client number.

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IP Address

IP address of the client. You can click the link of the IP address to visit the WEB interface of the client.

Network

Whether the client has been successfully connected to the gateway, including two states: Connected and Disconnected (or no display).

 Status Bar 5: DPC Information This region displays the information about DPC. The table below explains the information items in

Status Bar 5.

Item

Description

DPC#

DPC number which starts from 0.

DPC Code

Destination point code which is usually allocated by the central office.

DPC Status

Indicates whether the route to this DPC is available, involving two states Available and Unavailable. The message can be sent to the DPC only when the route to this DPC is at the state of Available. The DPC will turn into the state of Available as long as one of the linksets reaching the DPC is at the state of In Service.

DPC Route Table

Route to the DPC, i.e. linkset number.

 Status Bar 6: Link information This status bar displays the detailed information on the state of all signaling links, usually used for

searching the cause of service interrupt on a signaling link.

Link#

STA

POC

LSC

FSN

ERR

CHO

Link

Number

Link States

0-6

Link Failure

Causes

(interrupt)

Processor

Failures

0-3

Live

Communication

Server Service

0-1

Forward

Sequence

Number

spare

0: uploaded but not started

0: normal

0: service is unavailable

1: service interrupt

1: BSNR illegal

1: the local end processor failure

1: service is available

2: initial positioning

2: FIBR illegal

2: the remote end processor failure

3: positioned/ ready

3: T2 timeout

3: both ends processor failure

4: positioned/ not ready

4: T6 timeout, the remote end busy

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5: service on

5: L3 sends a command to stop

6: processor failure

6: signaling error rate too high

7: during the course of initial positioning, fail to enter a normal position

8: Timer 1 timeout

9: positioned and ready, receive the interrupt signal of the remote end

10: positioned but not ready, receive the interrupt signal of the remote end

11: in the state of Service On, receive the interrupt signal of the remote end

12: in a processor failure, receive the interrupt signal of the remote end

 Status Bar 7: Runtime Log Runtime log records all MTP3 commands and error information that pops up during the operation.

This status bar displays all the log records generated after the digital gateway starts.

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3.2.4 Call Count

Figure 3-9 Call Count Interface

See Figure 3-9 for the call count Interface. The above list shows the detailed information about all the calls counted from the startup of the gateway service to the latest open or refresh of this interface. This interface includes three parts: PSTN Call Statistics, Statistics on PSTN Release Cause and Statistics on Sip Release Cause. You can click Refresh to obtain the latest call count information. The table below explains the items shown in Figure 3-9.

Item

Description

Trunk No.

The number of the PCM trunk, numbered from 0.

Signaling Type

The signaling protocol applied on the digital trunk, including: ISDN User Side, ISDN Network Side, SS7-TUP, SS7-ISUP, and SS1.

Current Number of IP PSTN

The number of current calls from IP to PSTN.

Total Number of IP PSTN

The total number of current calls from IP to PSTN.

Connection Rate of IP PSTN

The percentage of successful IP PSTN calls to total IP PSTN calls.

Current Number of PSTN  IP

The number of current calls from PSTN to IP.

Total Number of PSTN  IP

The total number of current calls from PSTN to IP.

Connection Rate of PSTN  IP

The percentage of successful PSTN  IP calls to total PSTN  IP calls.

Total

Total number and connection rate of calls on all available tunks

Release Cause

Reason to release the call.

Normal call clearing

Total number of the calls which are normally cleared.

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Cancelled by calling party

Total number of the calls which are cancelled by the calling party.

User busy

Total number of the calls which fail as the called party has been occupied and replies a busy message.

No answer from user

Total number of the calls which fail as the called party does not pick up the call in a long time or the calling party hangs up the call before the called party picks it up.

Routing failed

Total number of the calls which fail because no routing rules are matched.

Resource unavailable

Total number of the calls which fail because no voice channel is available.

Unallocated number

Total number of the calls which fail as the called party number is unallocated.

Call rejected

Total number of the calls which fail as the called party replies a rejection message.

Normal unspecified

Total number of the calls which fail as the called party number is normal but unspecified.

Call failed

Total number of the calls which fail as the called party number does not conform to the number-receiving rule or for relative reasons.

Others

Total number of the calls which fail due to other unknown reasons.

Percentage

The percentage of the calls with a release cause to total calls.

3.3 VoIP Settings

VoIP Settings includes five parts: SIP, SIP Trunk, SIP Register, SIP Account, SIP Trunk Group and Media. See Figure 3-10. SIP is used to configure the general SIP parameters; SIP Trunk is used to set the basic and register information of the SIP trunk; SIP Register is used for the registration of SIP; SIP Account is used for registering SIP accounts to the SIP server; SIP Trunk Group is to manage SIP trunks by group; and Media is to set the RTP port and the payload type.

Figure 3-10 VoIP Settings

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3.3.1 SIP Settings

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Figure 3-11 SIP Settings Interface

See Figure 3-11 for the SIP settings interface where you can configure the general SIP parameters. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the items shown in Figure 3-11.

Item

Description

SIP Address

IP address for SIP signaling, using LAN 1 by default.

SIP Port

Monitoring port of SIP signaling. Range of value: 1024~65535, with the default value of 5060.

183 Message Behavior

Sets whether to send the 183 message instead of 180 to respond to the ringing tone when the SIP end serves as the called party. By default this feature is enabled.

Obtain CallerID from

There are two optional ways to obtain the calling party number: from Username of

“From” Field or from Displayname of “From” Field. The default value is from Username of “From” Field.

Obtain CalleeID from

There are two optional ways to obtain the called party number: from “To” Field or from “Request” Field. The default value is from “Request” Field.

Obtain Redirecting Number/Original CalleeID from Diversion Field

Sets whether to enable the feature of obtaining the Redirecting Number/Original CalleeID from Diversion Field. By default, the feature is disabled.

STUN Traversal

Sets whether to enable the STUN server for NAT traversal. By default the STUN server is disabled.

STUN Server Address

Address of the server for STUN traversal.

SIP Transport Protocol

There are two modes UDP and TCP available for running the SIP protocol. The default value is UDP.

SIP Encryption

Once this feature is enabled, you can encrypt the SIP signal following selecting an encryption criterion and setting a key. By default it is disabled.

Encryption Criterion

The criterion used to encrypt the SIP signal. At present only VOS1.1 is supported.

Key

The key to encrypt the SIP signal.

RTP Encryption

Once this feature is enabled, you can encrypt the RTP package. By default it is disabled.

RTP Self-adaption

When this feature is enabled, the RTP reception address or port carried by the signaling message from the remote end, if not consistent with the actual state, will be updated to the actual RTP reception address or port. By default, this feature is disabled.

UDP Header Checksum

When this feature is enabled, the gateway will automatically calculate the check sum of the UDP header during RTP transmission.

Rport

When this feature is enabled, a corresponding Rport field will be added to the Via message of SIP. By default, it is disabled.

DSCP

Sets whether to enable the DSCP differentiated services code point. By default, it is disabled.

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Voice Media

Sets the priority of the voice media for DSCP. The voice media with a bigger value has a higher priority. The value range is 0~63, with the default value of 46.

Signal Control

Sets the priority of the signal control for DSCP. The signal control with a bigger value has a higher priority. The value range is 0~63, with the default value of 26.

Maximum Wait Answer Time

Sets the maximum time for the SIP channel to wait for the answer from the called party of the outgoing call it initiates. If the call is not answered within the specified time period, it will be canceled by the channel automatically. The default value is 60, calculated by s.

Maximum Wait RTP Time

Sets the maximum time for the SIP channel to wait for the RTP packet. If no RTP packet is received within the specified time period, the channel will enter the pending state automatically and release the call. The default value is 0, calculated by s.

Maximum Wait PSTN Resource Time

Sets the maximum wait time to search the idle PSTN resource for the incoming call from IP. The call will be failed if no channel is found during this time. The value range is 0~10000, calculated by ms, with the default value of 5000.

3.3.2 SIP Trunk

Figure 3-12 SIP Trunk Settings Interface

See Figure 3-12 for the SIP trunk settings interface. A new SIP trunk can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-13 for the SIP trunk adding interface.

Figure 3-13 Add New SIP Trunk

The table below explains the items shown in Figure 3-13.

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Item

Description

Index

The unique index of each SIP trunk.

Remote Address

Address of the SIP trunk, i.e. the IP address or domain name of the remote SIP terminal which will establish call conversation with the gateway.

Remote Port

Port of the SIP trunk.

Outgoing Voice Resource

Maximum number of voice channels for the outgoing calls allocated by the SIP trunk to the gateway.

Incoming Voice Resource

Maximum number of voice channels for the Incoming calls allocated by the SIP trunk to the gateway.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-12 to modify a SIP trunk. See Figure 3-14 for the SIP trunk modification interface. The configuration items on this interface are the same as those on the Add New SIP Trunk interface.

Figure 3-14 Modify SIP Trunk

To delete a SIP trunk, check the checkbox before the corresponding index in Figure 3-12 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all SIP trunks at a time, click the Clear All button in Figure 3-12.

3.3.3 SIP Register

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Figure 3-15 SIP Register Configuration Interface

See Figure 3-15 for the SIP Register Configuration interface. By default, there is no SIP register available on the gateway. Click Add New to add them manually. See Figure 3-16.

Figure 3-16 Add SIP Register Interface

The table below explains the items shown in the above figure.

Item

Description

Index

The unique index of each SIP register.

SIP Trunk No.

The number of the SIP trunk which registers to the SIP server.

Username

When the gateway initiates a call to SIP, this item corresponds to the username of SIP; when the gateway initiates a call to PSTN, this item corresponds to the displayed CallerID.

Password

Registration password of the gateway. To register the gateway to the SIP server, both configuration items Username and Password should be filled in.

Address of the SIP server to which the SIP trunk is registered.

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The signaling port of the SIP trunk.

Domain Name

Domain name of the gateway used for SIP registry.

Validity period of the SIP registry. Once the registry is overdue, the gateway should be registered again. Range of value: 10~3600, calculated by s, with the default value of 3600.

IMS Network

Once this feature is enabled, the gateway will send signaling messages to the corresponding externally bound address and port when it registers to the server. Only when this feature is enabled will these items Externally Bound Address,

Externally Bound Port and Authentication Username be shown.

Externally Bound Address

Externally bound IP address for registration.

Externally Bound Port

Externally bound port for registration.

Authentication Username

Authentication username for registration.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Figure 3-17 SIP Register Information List

Click Modify in Figure 3-17 to modify a SIP register. The configuration items on the SIP Register Modification Interface are the same as those on the Add New SIP Register interface.

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Figure 3-18 SIP Register Modification Interface

To delete a SIP register, check the checkbox before the corresponding index in Figure 3-17 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all SIP registers at a time, click the Clear All button in Figure 3-17.

3.3.4 SIP Account

Figure 3-19 SIP Account Settings Interface

See Figure 3-19 for the SIP account settings interface. A new SIP account can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-20 for the

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SIP account adding interface.

Figure 3-20 Add New SIP Account

The table below explains the items shown in above figures.

Item

Description

Index

The unique index of each SIP account.

SIP Trunk No.

The number of the SIP trunk to which the SIP account is registered.

Username

The registration username of the SIP account. Once the SIP account is successfully registered, the SIP server can initiate calls to the gateway via Username.

Password

The registration password of the SIP account. To register the SIP account to the SIP trunk, both configuration items Username and Password should be filled in.

The validity period of the SIP account registry. Once the registry is overdue, the SIP account should be registered again. Range of value: 10~3600, calculated by s, with the default value of 3600.

The registration status of the SIP account. It is either Registered or Failed.

Authentication Username

Authentication username of a port, used to register the port to the SIP server when IMS network is enabled.

Note: This item appears only when IMS Network is enabled on the SIP trunk corresponding to this SIP account.

Description

More information about each SIP account.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-19 to modify a SIP account. See Figure 3-21 for the SIP account modification interface. The configuration items on this interface are the same as those on the Add

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New SIP Account interface.

Figure 3-21 Modify SIP Account

To delete a SIP account, check the checkbox before the corresponding index in Figure 3-19 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all SIP accounts at a time, click the Clear All button in Figure 3-19.

3.3.5 SIP Trunk Group

Figure 3-22 SIP Trunk Group Settings Interface

See Figure 3-22 for SIP trunk group settings interface. A new SIP trunk group can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-23 for the SIP trunk group adding interface.

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Figure 3-23 Add New SIP Trunk Group

The table below explains the items shown in Figure 3-23.

Item

Description

Index

The unique index of each SIP trunk group, which is mainly used in the configuration of routing rules and number manipulation rules to correspond to SIP trunk groups.

Description

More information about each SIP trunk group.

SIP Trunk Select Mode

When the SIP trunk group receives a call, it will choose a SIP trunk based on the select mode set by this configuration item to ring. The optional values and their corresponding meanings are described in the table below.

Option

Description

Increase

Search for an idle SIP trunk in the ascending order of the SIP trunk number, starting from the minimum.

Decrease

Search for an idle SIP trunk in the descending order of the SIP trunk number, starting from the maximum.

Cyclic Increase

Provided SIP Trunk N is the available SIP trunk found last time. Search for an idle SIP trunk in the ascending order of the SIP trunk number, starting from SIP Trunk N+1.

Cyclic Decrease

Provided SIP Trunk N is the available SIP trunk found last time. Search for an idle SIP trunk in the descending order of the SIP trunk number, starting from SIP Trunk N-1.

SIP Trunks

The SIP trunks in the SIP trunk group. If the checkbox before a SIP trunk is grey, it indicates that the SIP trunk has been occupied. The ticked SIP trunks herein will be displayed in the column „SIP Trunks‟ in Figure 3-22.

After configuration, click Save to save the settings into the gateway or click Cancel to cancel the settings.

Click Modify in Figure 3-22 to modify a SIP trunk group. See Figure 3-24 for the SIP trunk group modification interface. The configuration items on this interface are the same as those on the Add New SIP Trunk Group interface.

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Figure 3-24 Modify SIP Trunk Group

To delete a SIP trunk group, check the checkbox before the corresponding index in Figure 3-22 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all SIP trunk groups at a time, click the Clear All button in Figure 3-22.

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3.3.6 Media Settings

Figure 3-25 Media Settings Interface

See Figure 3-25 for the media settings interface where you can configure the RTP port and payload type depending on your requirements. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the items shown in Figure 3-25.

Item

Description

DTMF Transmit Mode

Sets the mode for the IP channel to send DTMF signals. The optional values are RFC2833, In-band and Signaling, with the default value of RFC2833.

RFC2833 Payload

Payload of the RFC2833 formatted DTMF signals on the IP channel. Range of value: 90~127, with the default value of 101.

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RTP Port Range

Supported RTP port range for the IP end to establish a call conversation, with the lower limit of 2000 and the upper limit of 60000 and the difference between larger than 512. The default value is 6000-10000.

Silence Suppression

Sets whether to send comfort noise packets to replace RTP packets or never to send RTP packets to reduce the bandwidth usage when there is no voice signal throughout an IP conversation. The optional values are Enable and Disable, with the default value of Disable. Note: When G723 is selected as CODEC, this configuration setting will turn to Enable automatically.

Noise Reduction

Once this feature is enabled, the volume of the noise accompanied with the line will be reduced automatically. The default setting is Enable.

JitterMode

Sets the working mode of JitterBuffer. The optional values are Static Mode and Adaptive Mode, with the default value of Static Mode.

JitterBuffer

Acceptable jitter for data packets transmission over IP, which indicates the buffering capacity. A larger JitterBuffer means a higher jitter processing capability but as well as an increased voice delay, while a smaller JitterBuffer means a lower jitter processing capability but as well as a decreased voice delay. Range of value: 0~280, calculated by ms, with the default value of 100.

JitterUnderrunLead

Sets the initial delay applied to received packets upon accepting packets later than the expected value set in JitterBuffer Item. Rnage of value: 0~280, calculated by ms, with the default value of 100, Note: Only when JitterMode is to Static Mode will this item be shown.

JitterOverrunLead

Sets the beforehand time inserted if receiving packets is ahead of time (the time of receiving is earlier than 300 minus the value set in JitterBuffer). Rnage of value: 0~280, calculated by ms, with the default value of 50, Note: Only when JitterMode is to Static Mode will this item be shown.

JitterMin

Sets the minimum delay that can be set by the adaptive jitter function. It can not be larger than the value set in JitterBuffer. Rnage of value: 0~280, calculated by ms, with the default value of 80. Note: Only when JitterMode is to Adaptive Mode will this item be shown.

JitterDecreaseRatio

Sets the rate of the delay that can be reduced under the adaptive mode. It defines the maximum percentage of silence that can be removed if reducing the delay. Rnage of value: 0~100, with the default value of 50, Note: Only when JitterMode is to Adaptive Mode will this item be shown.

JitterIncreaseMax

Sets the maximum delay can be increased during one silence period. Rnage of value: 0~280, calculated by ms, with the default value of 30, Note: Only when JitterMode is to Adaptive Mode will this item be shown.

Voice Gain Output from IP

Adjusts the voice gain of call from IP to the remote end. The value must be a multiple of 3. Range of value: -24~24, calculated by dB, with the default value of 0.

CODEC Priority

Supported CODECs and their corresponding priority for the IP end to establish a call conversation. The table below explains the sub-items:

Sub-item

Description

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Priority

Priority for choosing the CODEC in an SIP conversation. The smaller the value is, the higher the priority will be.

CODEC

Seven optional CODECs are supported: G711A, G711U, G729AB, G723, G722, AMR and iLBC.

Packing Time

Time interval for packing an RTP packet, calculated by ms.

Bit Rate

The number of thousand bits (excluding the packet header) that are conveyed per second.

By default, all of the seven CODECs are supported and ordered G711A, G711U, G729AB, G723, G722, AMR and iLBC by priority from high to low. The packing time and bit rate supported by different CODECs are listed in the table below. Those values in bold face are the default values.

COEDC

Packing Time (ms)

Bit Rate (kbps)

G711A

5 / 10 / 20 / 30 / 40 / 50 / 60

G711U

5 / 10 / 20 / 30 / 40 / 50 / 60

G729AB

20 8 G723

30 / 60 / 90

5.3 / 6.3

G722

5 / 10 / 20 / 30 / 40

AMR

20 / 40 / 60 / 80 / 100

4.75 / 5.15 / 5.90 / 6.70 / 7.40 /

7.95 / 10.20 / 12.20

iLBC

20 / 40

15.2

13.3

13.3 / 15.2

3.4 PCM Settings

PCM Settings includes eight parts: PSTN, Circuit Maintenance, PCM, PCM Trunk, PCM Trunk Group, Number-Receiving Rule, Reception Timeout and Number Attribution. See Figure

3-26.

Figure 3-26 PCM Settings

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3.4.1 PSTN

Figure 3-27 PSTN Settings Interface

See Figure 3-27 for the PSTN Settings interface. The table below explains the items shown in the above figure.

Item

Description

Interface

Actual type of the line connected with the E1/T1 interface on the gateway. Currently, only E1/T1 is supported.

Encoding Format

Sets the voice data encoding format for the voice channels on the digital trunk. The optional values are A-law and u-law, with the default value of A-law.

Echo Canceller

Sets whether to enable the echo cancellation feature for call conversations over the digital trunk. By default, this feature is enabled and the effect can reach 128ms.

Ringback Tone Provided by E1

Sets whether to enable the E1 end to provide the ringback tone, with the default value of disable.

Ringback Tone Provided by IP

Sets whether to enable the IP end to provide the ringback tone, with the default value of disable.

Ringback Tone Volume

Sets the volume of the ringback tone. Range of value: -35~-2, calculated by dB, with the default value of -25.

Voice Gain Output from PSTN

Adjusts the voice gain of call from PSTN to the remote end. The value must be a multiple of 3. Range of value: -24~24, calculated by dB, with the default value of 0.

Hot Back-up for E1

Sets whether to enable the feature of hot back-up for E1, with the default value of disable.

Gateway IP for Hot Back-up

Set the IP of the gateway for the hot back-up for E1.

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After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions.

3.4.2 Circuit Maintenance

Figure 3-28 Circuit Maintenance Interface

See Figure 3-28 for the Circuit Maintenance interface. You can block or unblock PCMs, ports and channels on this interface. Check All means to select all available items for the current port; Uncheck All means to cancel all selections for the current port; Inverse means to uncheck the selected items and check the unselected.

3.4.3 PCM

Figure 3-29 PCM Settings Interface

See Figure 3-29 for the PCM settings interface. The above list shows the detailed information and configurations of each PCM. The table below explains the items shown in the above figure.

Item

Description

PCM No.

The number of the PCM, numbered from 0. This item is not configurable.

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Signaling Protocol

The signaling protocol applied on the digital trunk. It includes ISDN User Side, ISDN Network Side, SS7-TUP, SS7-ISUP, and SS1 in E1, and only includes ISDN User Side, ISDN Network Side in T1.

Note: 1, Changing the interface type from E1 to T1 will forbid those non-ISDN

signaling modes in E1. And in such case, the gateway will by default set this item to ISDN User Side. 2, For SMG3008, a single gateway can be configured with two different

signaling modes simultaneously.

3, For SMG3016, a single gateway can be configured with three different

signaling modes simultaneously.

Clock

The clock mode for the digital trunk, including Line-synchronization, Free-run and Slave.

Signaling Time Slot

Sets the time slot used for signaling transmission on the digital trunk. If the configuration item Signaling Protocol is set to ISDN and SS1, the signaling time slot is Time Slot 16 in E1 or Time Slot 24 in T1 (SS1 not supported in T1 by far), which cannot be modified.

Signaling Link Type

Indicates whether the PCM is used as a signaling link or a voice link. If no time slot is used to transmit signaling, the PCM is a voice link.

Connection Line

Physical connection line type.

Incoming Call Start TS, Amount

ets a certain amount of channels which starts from a certain TS to process the incoming calls and others on the PCM to process outgoing calls. This is valid only when the configuration item Signaling Protocol is set to SS1.

CRC-4

Sets whether to enable the CRC-4 verification feature. By default, this feature is Enabled.

Click Modify in Figure 3-29 to modify a PCM. See Figure 3-30 for the PCM modification interface. Most configuration items on this interface are the same as those on the PCM Settings interface.

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Figure 3-30 Modify PCM

The table below explains the other configuration items on the PCM modification interface.

Item

Description

Use 'Signaling Time Slot' for Signaling

If this item is checked, it indicates that the signaling time slot configured in Signaling Time Slot is used for signaling transmission. You can see this item only when the configuration item Signaling Protocol is set to SS7-TUP or SS7-ISUP.

Apply to All PCMs

Check this item to apply the above settings (excluding Clock) to all PCMs.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

3.4.4 PCM Trunk

Figure 3-31 PCM Trunk Configuration Interface

See Figure 3-31 for the PCM Trunk Configuration interface. By default, there is no PCM trunk available on the gateway. Click Add New or Batch Add to add them manually. See Figure 3-32,

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Figure 3-33.

Figure 3-32 Add PCM Trunk Interface

Figure 3-33 PCM Trunk Batch Add Interface

The table below explains the items shown in the above figure.

Item

Description

Index

The unique index of each PCM trunk

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PCM NO.

The number of the PCM, numbered from 0.

Including Ts

Sets the TS included in this PCM which can make incoming/outgoing calls.

Including PCM

Sets the PCM included in the PCM trunk.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Figure 3-34 PCM Trunks List

Click Modify in Figure 3-34 to modify a PCM trunk. The configuration items on the PCM Trunk Modification Interface are the same as those on the Add PCM Trunk interface.

Figure 3-35 PCM Trunk Modification Interface

To delete a PCM trunk, check the checkbox before the corresponding index in Figure 3-34 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all PCM trunks at a time, click the Clear All button in Figure 3-34.

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3.4.5 PCM Trunk Group

Figure 3-36 PCM Trunk Group Settings

See Figure 3-36 for the PCM trunk group settings interface. A new PCM trunk group can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-37 for the PCM trunk group adding interface.

Figure 3-37 Add New PCM Trunk Group

The table below explains the items shown in Figure 3-37.

Item

Description

Index

The unique index of each PCM trunk group, which is mainly used in the configuration of routing rules and number manipulation rules to correspond to PCM trunk groups.

Description

More information about each PCM trunk group.

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PCM Trunk Select Mode

When the PCM trunk group receives a call, it will choose a PCM trunk based on the select mode set by this configuration item to ring. The optional values and their corresponding meanings are described in the table below.

Option

Description

Increase

Search for an idle PCM trunk in the ascending order of the PCM number, starting from the minimum.

Decrease

Search for an idle PCM trunk in the descending order of the PCM number, starting from the maximum.

Cyclic Increase

Provided PCM Trunk N is the available PCM trunk found last time. Search for an idle PCM trunk in the ascending order of the PCM number, starting from PCM Trunk N+1.

Cyclic Decrease

Provided PCM Trunk N is the available PCM trunk found last time. Search for an idle PCM trunk in the descending order of the PCM number, starting from PCM trunk N-1.

PCM Trunks

The PCM trunks in the PCM trunk group. If the checkbox before a PCM trunk is grey, it indicates that the PCM trunk has been occupied. The ticked PCM trunks herein will be displayed in the column „PCM Trunks‟ in Figure 3-36.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-36 to modify a PCM trunk group. See Figure 3-38 for the PCM trunk group modification interface. The configuration items on this interface are the same as those on the Add New PCM Trunk Group interface.

Figure 3-38 Modify PCM Trunk Group

To delete a PCM trunk group, check the checkbox before the corresponding index in Figure 3-36 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all PCM trunk groups at a time, click the Clear All button in Figure 3-36.

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3.4.6 Number-receiving Rule

The gateway uses a number-receiving plan to filter the numbers received from PSTN. Only those numbers which match the plan will be processed. The number-receiving plan consists of multiple number-receiving rules, each of which has a priority in sequence to avoid conflict.

Figure 3-39 Number-Receiving Rule Configuration Interface

See Figure 3-39 for the Number-receiving Rule Configuration interface. The list in the above figure shows the number-receiving rules with their priorities and description. A new number-receiving rule can be added by the Add New button on the bottom right corner. See Figure 3-40 for the number-receiving rule adding interface.

Figure 3-40 Add New Number-Receiving Rule

The table below explains the items shown in Figure 3-40.

Item

Description

Index

The unique index of each number-receiving rule, which denotes its priority. A number-receiving rule with a smaller index value has a higher priority and will be checked earlier while matching.

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Number-Receiving Rule

Up to 99 number-receiving rules can be configured in the gateway, and the maximum length of each number-receiving rule is 127 characters. See below for the meaning of each character in the number-receiving rule. The gateway will do instant matching for your receiving number based on the number-receiving rule and regard your receiving as finished upon receiving „#‟ or reception timeout.

Character

Description

“0”~”9”

Digits 0～9.

“x”

A random number. A string of „x‟s represents several random numbers. For example, „xxx‟ denotes 3 random numbers.

“.”

„.‟ indicates a random amount (including zero) of characters

after it.

“[ ]”

„[ ]‟ is used to define the range for a number. Values within it only

can be digits „0~9‟, punctuations „-‟ and „,‟. For example, [1-3,6,8] indicates any one of the numbers 1, 2, 3, 6, 8.

“-”

‟-‟ is used only in „[ ]‟ between two numbers to indicates any

number between these two numbers.

“,”

‟,‟ is used to separate numbers or number ranges, representing

alternatives.

By default, there is only one rule configured on the gateway. The table below lists 20 rules as example for your easy use and understanding. See below for detailed information.

Priority

Dialing Rule

Description

Any number in any length.

01[3,5,8]xxxxxxxxx.

Any 12-digit number starting with 013, 015 or 018

010xxxxxxxx

Any 11-digit number starting with 010

02xxxxxxxxx

Any 11-digit number starting with 02

0[3-9]xxxxxxxxxx

Any 12-digit number starting with 03, 04, 05, 06, 07, 08 or 09

120

Number 120

11[0,2-9]

Number 110, 112, 113, 114, 115, 116, 117, 118 or 119

111xx

Any 5-digit number starting with 111

123xx

Any 5-digit number starting with 123

95xxx

Any 5-digit number starting with 95

100xx

Any 5-digit number starting with 100

1[3-5,8]xxxxxxxxx

Any 11-digit number starting with 13, 14, 15 or 18

[2-3,5-7]xxxxxxx

Any 8-digit number starting with 2, 3, 5, 6 or 7

8[1-9]xxxxxx

Any 8-digit number starting with 81, 82, 83, 84, 85, 86, 87, 88 or 89

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80[1-9]xxxxx

Any 8-digit number starting with 801, 802, 803, 804, 805,.806, 807, 808 or 809

800xxxxxxx

Any 10-digit number starting with 800

4[1-9]xxxxxx

Any 8-digit number starting with 41, 42, 43, 44, 45, 46, 47, 48 or 49.

40[1-9]xxxxx

Any 8-digit number starting with 401, 402, 403, 404, 405, 406, 407, 408 or 409

400xxxxxxx

Any 10-digit number starting with 400

8xxx

Any 4-digit number starting with 8

Description

Remarks for the number-receiving rule. It can be any information, but can not be left empty.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-39 to modify the number-receiving rules. See Figure 3-41 for the number-receiving rule modification interface. The configuration items on this interface are the same as those on the Add New Number-receiving Rule interface.

Figure 3-41 Modify Number-receiving Rule

To delete a number-receiving rule, check the checkbox before the corresponding index in Figure 3-39 and click the „Delete‟ button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all number-receiving rules at a time, click the Clear All button in Figure 3-39.

3.4.7 Reception Timeout

Figure 3-42 Number-receiving Timeout Info Interface

See Figure 3-42 for the number-receiving timeout info interface. The table below explains the

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items shown in the above figure.

Item

Description

Inter Digit Timeout

Sets the largest interval between two digits of a receiving number. Range of value: 1~10, calculated by s, with the default value of 6. In case your number-receiving rules do not include “.”, the call will fail if there is no digit received or no number-receiving rule matched during this interval; in case your number-receiving rules include “.”, the gateway will wait until this interval ends and match to the number-receiving rule “.” if there is no digit received or no other number-receiving rule matched during this interval.

Description

More information about the configuration item Inter Digit Timeout, such as the reason for adopting the current value.

Click Modify in Figure 3-42 to modify the number-receiving timeout info. See Figure 3-43 for the number-receiving timeout info modification interface. The configuration items on this interface are the same as those on the Number-receiving Timeout Info Interface.

Figure 3-43 Modify Number-receiving Timeout Info

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

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3.4.8 Number Attribution

Figure 3-44 Number Attribution Setting Interface

See Figure 3-44 for the Number Attribution Setting interface, which is used to set whether to remove 0 or the area code from CalleeID. Click Save to save your settings into the gateway or click Reset to restore the configurations. Click Export to export and check the information about the number attribution; or select the required number attribution file via Browse… and click Import to import it into the gateway.

Note: By far only those numbers with the start 13x, 14x, 15x, 17x, or 18x can be imported.

3.5 SS7 Settings

Users can see the SS7 option in the menu only when the configuration item Signaling Protocol on the PCM settings interface is set to SS7-TUP or SS7-ISUP. SS7 Settings includes eight parts:

SS7, TUP, TUP Number Param, ISUP, Number Param, Original CalleeID Pool, Redirecting Number Pool and SS7 Server. See Figure 3-45.

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Figure 3-45 SS7 Settings

3.5.1 SS7

Figure 3-46 SS7 Settings Interface

See Figure 3-46 for the SS7 settings interface where you can configure the general SS7 parameters. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the items shown in Figure 3-46.

Item

Description

As Client Only

Sets whether the gateway serves as Client only or not. If it is set to No (default), the SS7 server will be disabled.

Master IP

Sets the IP address of the master SS7 server, with the default value of 127.0.0.1, which indicates that there is only one SS7 server available.

Slave IP

Sets the IP address of the slave SS7 server. Only when the item Dual Gateway is ticked can this item be configured.

Local IP Address

Sets the IP address of the local PC, with the default value of 127.0.0.1.

Dual Gateway

If this feature is enabled, two SS7 servers are used at the same time in the system. The configuration items Master IP and Slave IP are respectively used to set the IP addresses of the master and slave servers.

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3.5.2 TUP

Figure 3-47 TUP Settings Interface

See Figure 3-47 for the TUP settings interface. Users can see this interface and configure the general TUP parameters only when the configuration item Signaling Protocol on the PCM settings interface is set to SS7-TUP. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17

Restart for detailed instructions. The table below explains the items shown in Figure 3-47.

Item

Description

Send GRM Group Message Using All-0 Field

If this configuration item is enabled, when the local driver sends the circuit group message to the remote PBX, this message covers all time slots TS1~31. By default this item is enabled.

Send ST Signal with CallerID in Outgoing Call

If this configuration item is enabled, the calling party number string sent by the gateway contains the ST signal in the outgoing call. By default this item is disabled.

Default Caller Parameter

Sets the address indicator in the calling line identification field in the IAI message. The optional values are: Local subscriber number, Spare national number, Valid

national number and International number, with the default value of Valid national number.

Set Caller Parameter in case of Original CalleeID

Once this feature is enabled, if the IP end carries the original CalleeID in a call from IP to PSTN, you shall set a separate value for the address indicator in the calling line identification field in the IAI message, i.e. Caller Parameter ( with Original CalleeID). By default this configuration item is disabled.

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Caller Parameter (with Original CalleeID)

This item is valid only when Set Caller Parameter in case of Original CalleeID is enabled. It sets the address indicator in the calling line identification field in the IAI message when the IP end carries the original CalleeID in a call from IP to PSTN. The optional values are: Local subscriber number, Spare national number, Valid

national number and International number, with the default value of Valid national number.

Default Original Callee Parameter

Sets the address indicator in the original called party address field of the IAI message. The optional values are: Local subscriber number, Spare national

number, Valid national number and International number, with the default value of Valid national number.

Maximum Wait Answer Time (s)

Sets the maximum time to wait for the answer from the called party of an outgoing call. If the call is not answered within the specified time period, it will be canceled by the channel automatically. The default value is 60, calculated by s.

Minimum Length of the CalleeID of an Incoming Call

Sets the minimum length of the CalleeID under the fixed-length mode. The value range is 1n40. Provided it is set to n, that is, the local end has received all the n digits of the called party number of the incoming call, the number reception will be regarded as finished.

3.5.3 TUP Number Parameter

Figure 3-48 TUP Number Parameter Configuration Interface

See Figure 3-48 for the TUP Number Parameter Configuration interface, which is used to set the corresponding parameters for the calling party number in TUP.

A new TUP number parameter can be added by the Add New button. See Figure 3-49 for the calling party number adding interface.

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Figure 3-49 Add New Calling Party Number Parameter

The table below explains the items shown in the above figure.

Item

Description

No.

The corresponding number for a calling party number parameter, which starts from

Prefix

A string of numbers at the beginning of a calling party number.

Parameter

Sets the parameter for a calling party number.

Set Parameter if Original CalleeID Available

Set whether to enable the feature of setting this parameter only if the original CalleeID is available.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-48 to modify the calling party number parameter. See Figure 3-50 for the calling party number parameter modification interface. The configuration items on this interface are the same as those on the Add New Calling Party Number Parameter interface.

Figure 3-50 Modify Calling Party Number Parameter

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To delete a calling party number parameter, check the checkbox before the corresponding index in Figure 3-48 and click the „Delete‟ button. To clear all calling party number parameters at a time, click the Clear All button in Figure 3-48.

Note: If there are two or more calling party numbers with the same prefix, the one numbered the smallest is valid and all the others become invalid.

3.5.4 ISUP

Figure 3-51 ISUP Settings Interface

See Figure 3-51 for the ISUP settings interface. Users can see this interface and configure the general ISUP parameters only when the configuration item Signaling Protocol on the PCM settings interface is set to SS7-ISUP. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17

Restart for detailed instructions. The table below explains the items shown in Figure 3-51.

Item

Description

Calling Party’s Category

Sets the calling party‟s category indicator in the IAM message. The optional values are: National operator, Ordinary calling subscriber, Calling subscriber with priority,

Data call, Test call and Payphone/Others, with the default value of Ordinary calling subscriber.

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Default Caller Parameter

Sets the calling party number parameter field in the IAM message. The optional values are: Subscriber number, National number, and International number, with the default value of Subscriber number.

Default Callee Parameter

Sets the called party number parameter field in the IAM message. The optional values are: Subscriber number, National number, and International number, with the default value of National number.

Set Caller/Callee Parameter in case of Original CalleeID

Once this feature is enabled, if the IP end carries the original CalleeID in a call from IP to PSTN, you shall set separate values for the caller and callee parameters in the IAM message, i.e. Caller Parameter (with Original CalleeID) and Callee

Parameter (with Original CalleeID). By default this configuration item is disabled.

Caller Parameter (with Original CalleeID)

This item is valid only when Set Caller/Callee Parameter in case of Original CalleeID is enabled. It sets the calling party number parameter field in the IAM

message when the IP end carries the original CalleeID in a call from IP to PSTN. The optional values are: Subscriber number, National number, and International number, with the default value of Subscriber number.

Callee Parameter (with Original CalleeID)

This item is valid only when Set Caller/Callee Parameter in case of Original CalleeID is enabled. It sets the called party number parameter field in the IAM

Send Generic Number

Sets the generic number parameter in IAM message, with the default value of disabled.

Transmission Medium Requirement

Sets the transmission medium requirement parameter in the IAM message. The optional values are: Speech, 64 kb/s unrestricted, 3.1khz audio, Alternative: speech

(service 2)/ 64kbit/s unrestricted (service 1) (Spare), Alternative: 64kbit/s unrestricted (service 1)/ speech (service 2) (Spare), 64kb/s preferred, 2*64kb/s unrestricted, 384 kb/s unrestricted, 1920 kb/s unrestricted and Spare, with the default value of Speech.

Obtain First Called Party Number from

Sets where the first called party number is obtained from. The optional values are: Only original CalleeID and Original CalleeID/ Redirecting number, with the default value of Only original CalleeID.

Auto Reply INF upon Reception of Remote INR

If this feature is enabled, once the INR message is received from the remote PBX in an outgoing call, the driver will automatically reply it with the INF message. By default this feature is enabled.

Reset Circuit upon Service Start before Entering Idle State

If this feature is enabled, the circuit will send a circuit reset message before entering the idle state after the ISUP service is enabled. By default this feature is enabled.

Information on First Two Bytes of Redirecting Number

Sets the first two bytes of the redirecting number in the IAM message, including the nature of address indicator, numbering plan indicator and address presentation restricted indicator, with the default value of 0x1001.

Maximum Wait Answer Time (s)

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Minimum Length of the CalleeID of an Incoming Call

Forward Call Indicator

Sets the forward call indicator in the IAM message, with the default value of 0x0040.

Nature of Connection Indicator

Sets the nature of connection indicator in the IAM message, with the default value of 0x00.

User Service Information

Sets whether the IAM message contains the user service information. By default this feature is disabled. If this feature is enabled, its value is usually determined by the remote PBX, with the default value of 0x80, 0x90, 0xa3. This default value is applicable to Huawei PBXes.

Optional Forward Call Indicator

Sets whether the IAM message contains the optional forward call indicator. By default this feature is disabled. If this feature is enabled, its value is usually determined by the remote PBX, with the default value of 0x00.

3.5.5 ISUP Number Parameter

Figure 3-52 ISUP Number Parameter Configuration Interface

See Figure 3-52 for the ISUP Number Parameter Configuration interface, which includes two parts: Calling Party Number Parameter and Called Party Number Parameter.

A new calling/called party number parameter can be added by the Add New button. See Figure 3-53, Figure 3-54 for the calling/called party number parameter adding interface.

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Figure 3-53 Add New Calling Party Number Parameter

Figure 3-54 Add New Called Party Number Parameter

The table below explains the items shown in above figures.

Item

Description

No.

The corresponding number for a calling/called party number parameter, which starts from 0.

Prefix

A string of numbers at the beginning of a calling/called party number.

Parameter

Sets the parameter for a calling/called party number.

Set Parameter if Original CalleeID Available

Set whether to enable the feature of setting this parameter only if the original CalleeID is available.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

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Click Modify in Figure 3-52 to modify the calling/called party number parameter. See Figure 3-55, Figure 3-56 for the calling/called party number parameter modification interface. The configuration items on this interface are the same as those on the Add New Calling/Called Party Number Parameter interface.

Figure 3-55 Modify Calling Party Number Parameter

Figure 3-56 Modify Called Party Number Parameter

To delete a calling/called party number parameter, check the checkbox before the corresponding index in Figure 3-52 and click the „Delete‟ button. To clear all calling/called party number parameters at a time, click the Clear All button in Figure 3-52.

Note: If there are two or more calling/called party numbers with the same prefix, the one numbered the smallest is valid and all the others become invalid.

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3.5.6 Original CalleeID Pool

Figure 3-57 Original CalleeID Pool Interface

See Figure 3-57 for the Original CalleeID Pool interface, which is used to add the original CalleeID for all outgoing calls or some special calls which contain the specified calling/called prefix.

A new original CalleeID can be added by the Add New button. See Figure 3-58 for the original CalleeID adding interface.

Figure 3-58 Add New Original CallerID

The table below explains the items shown in above figures.

Item

Description

No.

The corresponding number for an added original CalleeID. The value range is 0~99.

CallerID Prefix

A string of numbers at the beginning of a calling party number, which can be numbers or “*” (indicating any string).

CalleeID Prefix

A string of numbers at the beginning of a called party number, which can be numbers or “*” (indicating any string).

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Original CalleeID Range

The range of the original CalleeID in the Original CalleeID Pool. It must be filled in with numbers and can not be left empty.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-57 to modify the calling/called party number parameter. See Figure 3-59, for the original CalleeID modification interface. The configuration items on this interface are the same as those on the Add New Original CalleeID interface. Note that the item No. cannot be modified.

Figure 3-59 Modify Original CalleeID

Note: If there are two or more calling/called party numbers with the same prefix, Starting Original CalleeID and Number Amount corresponding to the one numbered the smallest are valid and all the others become invalid.

3.5.7 Redirecting Number Pool

Figure 3-60 Redirecting Number Pool Interface

See Figure 3-60 for the Redirecting Number Pool interface, which is used to set the redirecting number in the setup message for all outgoing calls or some calls which contain a specified

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calling/called prefix. This feature is only applicable to ISUP calls. A new redirecting number can be added by the Add New button. See Figure 3-61 for the

redirecting number adding interface.

Figure 3-61 Add New Redirecting Number

The table below explains the items shown in above figures.

Item

Description

No.

The corresponding number for an added redirecting number. The value range is 0~99.

CallerID Prefix

A string of numbers at the beginning of a calling party number, which can be numbers or “*” (indicating any string).

CalleeID Prefix

A string of numbers at the beginning of a called party number, which can be numbers or “*” (indicating any string).

Redirecting Information

Sets the redirection information field in the IAM message. The parameter type of the redirection information field is 0x13, which contains 2 bytes. By default, it is set to 0x0321, i.e. call forwarding on no answer. Refer to the ISUP protocol standard for the detailed description of each byte.

Redirecting Number Range

The range of the redirecting number in the Redirecting Number Pool. It must be filled in with numbers and can not be left empty.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-60 to modify the redirecting number parameter. See Figure 3-62 for the redirecting number modification interface. The configuration items on this interface are the same as those on the Add New Redirecting Number interface. Note that the item No. cannot be modified.

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Figure 3-62 Modify Redirecting Number

To delete a redirecting number parameter, check the checkbox before the corresponding index in Figure 3-60 and click the „Delete‟ button. To clear all redirecting number parameters at a time, click the Clear All button in Figure 3-60.

Note: If there are two or more calling/called party numbers with the same prefix, only the one numbered the smallest are valid for Starting Redirecting Number and Number Amount.

3.5.8 SS7 Server

Figure 3-63 SS7 Server Configuration Interface

When the gateway uses the SS7 signaling, it must run the SS7 server first. See Figure 3-63 for the SS7 configuration interface, where you can set the SS7 server configuration file (Ss7server.ini). Follow the instructions below to accomplish the configurations step by step.

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Step 1: Set Server IP and Signaling Point Code Standard. See Region 1 in Figure 3-63. The table below explains these configuration items.

Item

Description

Server 1 IP

Sets the IP address for the master SS7 server. If only one server is used in the system, there is no need to set the configuration item Server 2 IP.

Server 2 IP

Sets the IP address for the slave SS7 server.

Signaling Point Code Standard

The value of this item varies on the PBX model. The optional values are 14 and 24, with the default value of 24. The China SS7 uses 24.

Subservice Code

Sets the SS7 subservice code. The optional values are: International network, Spare international network, National network, Spare national network, with the default value of Spare national network.

Send SLTM

Sets whether to regularly send the Signaling Link Test Message (SLTM) to the remote PBX. By default it is disabled.

After configuration, click Save to save the settings into the gateway. Step 2: Configure the client. See Region 2 in Figure 3-63. A new client can be added by the Add New button on the bottom right corner of the client list. See

Figure 3-64 for the new client adding interface.

Figure 3-64 Add New Client

The table below explains the configuration items in the above figure.

Item

Description

No.

The unique index of each client, which is mainly used in the configuration of signaling links to correspond to the client, numbered from 0.

IP Address

IP address of the client.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

To modify a client, click Modify in the client list. The configuration items on the modification interface are the same as those on the Add New Client interface.

To delete a client, check the checkbox before the corresponding index and click the Delete button under the list. To clear all clients at a time, click the Clear All button. Note: If a client is occupied by a signaling link, it cannot be deleted or cleared unless you delete the signaling link first. You can only delete the clients in turn from back to front.

Step 3: Configure signaling links and linksets. See Region 3 in Figure 3-63. The link used to transmit signaling messages between two signaling points is called Signaling

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Link. Each signaling link maps a physical address. A new signaling link can be added by the Add New button on the bottom right corner of the signaling link list. See Figure 3-65 for the new signaling link adding interface.

Figure 3-65 Add New Signaling Link

The table below explains the configuration items in the above figure.

Item

Description

No.

The unique index of each signaling link, which is mainly used in the configuration of signaling linksets to correspond to the signaling link, numbered from 0.

Client

Client number. This configuration item together with PCM determines the physical address of the E1 interface of the signaling link. Each physical address maps a signaling link.

PCM

Local PCM number. This configuration item together with Client determines the physical address of the E1 interface of the signaling link.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

To modify a signaling link, click Modify in the signaling link list. The configuration items on the modification interface are the same as those on the Add New Signaling Link interface.

To delete a signaling link, check the checkbox before the corresponding index and click the Delete button under the list. To clear all signaling links at a time, click the Clear All button. Note: If a signaling link is occupied by a signaling linkset, it cannot be deleted or cleared unless you delete the signaling linkset first. You can only delete the signaling links in turn from back to front.

A group of signaling links used to connect two signaling points directly constitute a signaling linkset. A new signaling linkset can be added by the Add New button on the bottom right corner of the signaling linkset list. See Figure 3-66 for the new signaling linkset adding interface.

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Figure 3-66 Add New Signaling Linkset

The table below explains the configuration items in the above figure.

Item

Description

No.

The unique index of each signaling linkset, which is mainly used in the configuration of DPC to correspond to the signaling linkset, numbered from 0.

Link

The signaling links in the linkset. If the checkbox before a link is grey, it indicates that the link has been occupied.

OPC

Originating Point Code for the signaling server which is usually allocated by the central office,. See the table below for the format and the value range:

14 bit

24 bit

Decimal (a.b.c)

a, c: 0~7, b: 0~255

a, b, c: 0~255

Hexadecimal (abc)

a, c: 3-digit hexadecimal number, b: 8-digit hexadecimal number

a, b, c: hexadecimal

number inbetween 00~ff

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

To modify a signaling linkset, click Modify in the signaling linkset list. The configuration items on the modification interface are the same as those on the Add New Signaling Linkset interface.

To delete a signaling linkset, check the checkbox before the corresponding index and click the Delete button under the list. To clear all signaling linkset at a time, click the Clear All button. Note: If a signaling linkset is occupied by a DPC, it cannot be deleted or cleared unless you delete the DPC first. You can only delete the signaling linksets in turn from back to front.

Step 4: Configure DPC. See Region 4 in Figure 3-63. The signaling point that receives messages is called Destination Point Code (DPC). A new DPC

can be added by the Add New button on the bottom right corner of the DPC list. See Figure 3-67 for the new DPC adding interface.

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Figure 3-67 Add New DPC

The table below explains the configuration items in the above figure.

Item

Description

No.

The unique index of each DPC, which is mainly used in the configuration of TUP_CIC Route or ISUP_CIC Route to correspond to the DPC, numbered from 0.

Associated Mode/ Quasi-associated Mode

Sets the way to transmit signaling messages between two signaling points, including Associated Mode and Quasi-associated Mode. Directly connecting the signaling links between two signaling points to transmit the inbetween signaling messages is called Associated Mode. Connecting two or more than two signaling links serially via one or more than one signaling transport points to transmit signaling messages, provided the path of signaling messages through the signaling network is predetermined and fixed within a certain period of time, is called Quasi-associated Mode. These two concepts are vividly illustrated below.

SP Code

Signaling point code of the DPC, usually allocated by the central office.

STP

Sets the first STP (signaling transport point) the signaling message reaches during the transmission under the quasi-associated mode. Only when you select the quasi-associated mode can this item be seen and configured.

Linkset

The linkset which is used to transmit signaling messages. For the associated mode, this item sets the signaling linksets between the OPC and the DPC. For the quasi-associated mode, this item sets the signaling linksets between the OPC and the first STP (signaling transport point).

Legend:

Signaling Relationship

Signaling Link Set

STP

(a) Associated Mode

(b) Quasi-associated Mode

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After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

To modify a DPC, click Modify in the DPC list. The configuration items on the modification interface are the same as those on the Add New DPC interface.

To delete a DPC, check the checkbox before the corresponding index and click the Delete button under the list. To clear all DPCs at a time, click the Clear All button. Note: If a DPC is occupied by a CIC routing rule, it cannot be deleted or cleared unless you delete the routing rule first. You can only delete the DPCs in turn from back to front.

Step 5: Configure TUP_CIC or ISUP_CIC Route. See Region 5 in Figure 3-63. A new TUP_CIC routing rule can be added by the Add New button on the bottom right corner of

the TUP_CIC routing rule list. See Figure 3-68 for the new TUP_CIC routing rule adding interface.

Figure 3-68 Add New TUP_CIC Routing Rule

The table below explains the configuration items in the above figure.

Item

Description

No.

The unique index of each CIC routing rule, which is numbered from 0.

DPC

DPC used in the routing rule.

CIC_PCM

PCM number in the CIC field and the value is obtained by dividing the initial CIC number from the central office by 32.

CIC_PCM Range

Range of the PCM time slots corresponding to CIC.

Client

Client number. This configuration item together with PCM determines the local PCM in the CIC routing rule.

PCM

PCM number on the client.

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Consecutively add _CIC_PCM for this DPC

Consecutively adds one or more CIC_PCM routes for a DPC.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

To modify a routing rule, click Modify in the TUP_CIC routing rule list. The configuration items on the modification interface are the same as those on the Add New TUP_CIC Routing Rule interface.

To delete a routing rule, check the checkbox before the corresponding index and click the Delete button under the list. To clear all routing rules at a time, click the Clear All button.

For the ISUP_CIC route settings, click the ISUP_CIC Route tab in Region 5 in Figure 3-63. See Figure 3-69 for the ISUP_CIC route settings interface. The configuration items and operations on this interface are absolutely the same as those in the TUP_CIC route settings interface. Note: Besides the default setting, the CIC Range for ISUP_CIC route can also be user-defined.

Figure 3-69 ISUP_CIC Route Settings Interface

After completing the configurations on SS7 Server Configuration Interface (Figure 3-63), you shall restart the service to validate them. Refer to 3.12.17 Restart for detailed instructions.

3.6 ISDN Settings

Users can see the ISDN option in the menu only when the configuration item Signaling Protocol on the PCM settings interface is set to ISDN User Side or ISDN Network Side. See Figure 3-70.

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Figure 3-70 ISDN Settings

3.6.1 ISDN

Figure 3-71 ISDN Settings Interface

See Figure 3-71 for the ISDN settings interface where users can configure the general ISDN parameters. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the items shown in Figure 3-71.

Item

Description

TEI

Terminal Equipment Identifier, which is used to identify the service access point in the point-to-point data link connection. Range of value: 0~63, with the default value of 0. Note: The TEI values at the corresponding user side and the network side must be the same.

Ch Identification

Sets the way to represent channel identification messages on the digital trunk. The optional values are: Number and Time slot diagram, with the default value of Number.

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Default Callee Type

Sets the type of number and numbering scheme for the called party numbers in the SETUP message during the outgoing call. The optional values are: National number, International number, Network number, Subscriber number and Unknown, with the default value of National number.

Default Caller Type

Sets the type of number and numbering scheme for the calling party numbers in the SETUP message during the outgoing call. The optional values are: National number, International number, Network number, Subscriber number and Unknown, with the default value of National number.

CODEC

Sets the voice CODEC used on the digital trunk. The optional values are A-Law and u-Law, with the default value of A-Law.

CRC Check

Sets whether to enable the feature of CRC check for the digital trunk at ISDN user side or network side. By default this feature is enabled.

Set Caller/Callee Type in case of Redirecting Num

Once this feature is enabled, if the IP end carries the redirecting number in a call from IP to PSTN, you shall set separate values for the type of number and numbering scheme for the calling and called party numbers in the SETUP message, i.e. Callee Type (with Redirecting Num) and Caller Type (with Redirecting

Num). By default this configuration item is disabled.

Callee Type (with Redirecting Num)

This item is valid only when Set Caller/Callee Type in case of Redirecting Num is enabled. It sets the type of number and numbering scheme for the called party numbers in the SETUP message when the IP end carries the redirecting number in a call from IP to PSTN. The optional values are: National number, International number, Network number, Subscriber number and Unknown, with the default value of National number.

Caller Type (with Redirecting Num)

This item is valid only when Set Caller/Callee Type in case of Redirecting Num is enabled. It sets the type of number and numbering scheme for the calling party numbers in the SETUP message when the IP end carries the redirecting number in a call from IP to PSTN. The optional values are: National number, International number, Network number, Subscriber number and Unknown, with the default value of National number.

Transfer Capability

Sets the „Transfer Capability‟ filed in the signaling message. The optional values are

Voice and 3.1k Audio, with the default value of Voice.

Enter Auto Alert State upon Reception of 'CALL PROCEEDING' Message

If this item is checked, the system will go into the state of auto alert when it receives the 02 (CALL PROCEEDING) message and the progress indicator turns to be 8 or

1. By default this item is disabled.

Enter Auto Alert State upon Reception of 'PROGRESS' Message

If this item is checked, the system will go into the state of auto alert when it receives the 03 (PROGRESS) message and the progress indicator turns to be 8 or 1. By default this item is disabled.

Maximum Wait Time for Called Party’s Pick up

The maximum time waiting for the called party to pick up the call after the channel state turns to „WaitAnswer‟ during an outgoing call. The default value is 60, calculated by s.

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Minimum Length of the CalleeID of an Incoming Call

Preferential Channel Selection

Sets whether to allow the preferential channel selection. By default this item is unchecked.

Send Channel Identification Message

Sets whether the channel identification message is included in the corresponding reply message (such as CALL PROCEEDING, ALERT, etc.) after the local end receives the SETUP message from the remote PBX during an incoming call. By default this item is checked.

Wait Confirm Time (T310)

Sets the maximum time that the local end waits for the remote end to send back the acknowledgement message in an outgoing call. If no acknowledgement message is received within the specified time period, the local end will disconnect the call automatically. For ISDN User Side, the default value is 15; for ISDN Network Side, the default value is 20, calculated by s.

Send the ‘Called Party Number Completed’ Parameter

Sets whether to include or not the „Called Number Complete‟ parameter in the SETUP message during an outgoing call.

3.6.2 Number Parameter

Figure 3-72 Number Parameter Configuration Interface

Number Parameter for ISDN is almost the same as that for SS7; only the calling/called party number changes from SS7 to ISDN; “set parameter if original CalleeID available” changes to “set parameter if redirecting number available” in ISDN. See Figure 3-72 for Number Parameter for ISDN. The configuration items on this interface are the same as those on Number Parameter for SS7 (Figure 3-53, Figure 3-54).

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3.6.3 Redirecting Number

Figure 3-73 Redirecting Number Interface

Redirecting Number Pool for ISDN is almost the same as Original CalleeID Pool for SS7; only the calling/called party number changes from SS7 to ISDN. See Figure 3-73 for Redirecting Number Pool for ISDN. The configuration items on this interface are the same as those on original CalleeID pool for SS7 (Figure 3-58).

3.6.4 Add Gateway

Figure 3-74 Add Gateway Interface

See Figure 3-74 for the Add Gateway Interface. A new gateway can be added by the Add New button on the bottom right corner of the list in the above figure. The information about the added gateway will be displayed under Operation Info  PSTN Status. See Figure 3-75 for the gateway adding interface.

Figure 3-75 Add New Gateway

The table below explains the items shown in above figures.

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Item

Description

No.

The corresponding number for a new gateway, which starts from 0.

The corresponding IP address for the new gateway, which must be in the same network section of the SIP address set via VoIPSIP.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-74 to modify the gateway information. See Figure 3-76 for the gateway modification interface. The configuration items on this interface are the same as those on the Add New Gateway interface.

Figure 3-76 Modify Gateway Information

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3.7 SS1 Settings

Figure 3-77 SS1 Settings Interface

See Figure 3-77 for the SS1 settings interface. This interface appears only when the configuration item Signaling Protocol on the PCM settings interface is set to SS1. You can set general information of SS1. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the items shown in Figure 3-77.

Item

Description

Country

Sets the country to use SS1, with the default value of CHINA.

C/D Value

Sets the value of CD in the ABCD signaling codes sent by the local end to the remote PBX. The high 6 bits should be set to 0, being reserved; the low 2 bits are C/D signaling codes, Bit1 (Signaling Code C) and Bit0 (Signaling Code D), both with the default value of 1.

ABCD Duration Timeout

Sets the minimum duration of ABCD signaling codes sent out by the remote PBX, calculated by millisecond (ms), which has to be the multiple of 8, with the default value of 0. Only when the on-line ABCD signaling codes vary and the new value keeps for more than the time specified by this configuration item will the gateway confirm the change of ABCD codes, Otherwise, the driver will believe there are undesired dithering signals on the line.

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Max MFC Waiting Time

Sets the maximum waiting time, i.e. the timer T2 for the SS1 state machine, calculated by second, with the default value of 10.

CalleeID Length for Incoming Calls

Sets the way to receive the number, with the default value of 3 which means receiving all the 3 digits of the called party number of the incoming call will put the local number reception into an end.

KB Setting Timeout

Sets the maximum time to wait for the application to configure the KB signal, calculated by second, with the default value of 3.

KD Wait Time

Sets the maximum time to wait for the remote PBX to send the KD signal (i.e. the timer T3) in the SS1 channel state machine, calculated by second, with the default value of

60.

Delay Time before Ringing State

Sets the delay time, i.e. a period of waiting time before the channel goes into the „Ringing‟ state following the reception of the complete called party number in case the SS1 channel on the gateway serves as the incoming end. It is calculated by second, with the default value of 0.

ACK Wait Timeout

Sets the value of the timer T5, calculated by second, with the default value of 60.

Calling Party’s Category (KA Signal)

Sets the KA signal (calling party‟s category at the local end) sent in an outgoing call. The value range is 1~10, with the default value of 1 (ordinary/regular).

KB Wait Timeout

Sets the maximum time to wait for the KB signal from the remote PBX, calculated by second, with the default value of 60.

Originating Service Type (KD Signal)

Sets the originating service type, i.e. KD, for an outgoing call. The value range is 1~6, with the default value of 3 (local call).

3.8 Fax Settings

See Figure 3-78 for the Fax Settings interface which is used to modify the special fax configurations.

Figure 3-78 Fax Settings

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3.8.1 Fax

Figure 3-79 Fax Configuration Interface (T.38 Mode)

See Figure 3-79 for the fax configuration interface with all default settings under the T.38 fax mode. Users can configure the general fax parameters via this interface. After configuration, click Save to save your settings into the gateway or click Reset to restore the configurations. If a dialog box pops up after you save your settings asking you to restart the service, do it immediately to apply the changes. Refer to 3.12.17 Restart for detailed instructions. The table below explains the configuration items in Figure 3-79.

Item

Description

Fax Mode

The real-time IP fax mode. The optional values are T.38, Pass-through and Disable, with the default value of T.38. Setting this item to Disable means to disable both T.38 and Pass-through.

T38 Version

Version of T.38 which is defined by ITU-T. Range of value: 0~3, with the default value of 0.

T38 Negotiation

Sets the Negotiation mode of T.38, including: Unsupported, Initiate Negotiation as Fax Sender and Initiate Negotiation as Fax Receiver.

Maximum Fax Rate

Sets the maximum faxing rate for both receiving and transmitting. Range of value: 14400, 9600 and 4800, calculated by bps, with the default value of 9600.

Fax Train Mode

Sets the train mode for T.38 fax. The optional values are transferredTCF and localTCF, with the default value of transferredTCF.

Error Correction Mode

Sets the error correction mode for T.38 fax. The optional values are t38UDPRedundancy (Redundancy Error Correction) and t38UDPFEC (Forward Error Correction), with the default value of t38UDPRedundancy.

T.30 Ecm

Sets whether to enable the T.30 error correction mode. By default this feature is enabled.

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Min Duration of CNG

As stipulated in the standard FAX CNG, the minimum duration of CNG is 500ms ± 15%, calculated by ms, with the default value of 425.

Note: Usually there is no need to modify it; please contact our technicians if necessary.

Min Duration of CED

As stipulated in the standard FAX CED, the minimum duration of CED is 2600~4000ms, calculated by ms, with the default value of 2600.

Note: Usually there is no need to modify it; please contact our technicians if necessary.

If you set Fax Mode to Pass-through, you can see the interface shown as Figure 3-80.

Figure 3-80 Fax Configuration Interface (Pass-through Mode)

The table below explains the configuration item in the above figure.

Item

Description

Pass-through Payload

RTP Payload under the pass-through fax mode. Range of value: 96~127, with the default value of 102.

3.9 Route Settings

Route Settings is used to specify the routing rules for calls on two directions: IPPSTN and PSTNIP. See Figure 3-81.

Figure 3-81 Route Settings

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3.9.1 Routing Parameters

Figure 3-82 Routing Parameters Configuration Interface

See Figure 3-82 for the routing parameters configuration interface. On this interface, you can set the routing rules for calls respectively on two directions IPPSTN and PSTNIP to be routing before or after number manipulation. The default value is Route before Number Manipulate.

After configuration, click Save to save the above settings into the gateway.

3.9.2 IP to PSTN

Figure 3-83 IPPSTN Routing Rule Configuration Interface

See Figure 3-83 for the IPPSTN routing rule configuration interface. A new routing rule can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-84 for the IPPSTN routing rule adding interface.

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Figure 3-84 Add New Routing Rule (IPPSTN)

The table below explains the items shown in the above figure.

Item

Description

Index

The unique index of each routing rule, which denotes its priority. A routing rule with a smaller index value has a higher priority. If a call matches several routing rules, it will be processed according to the one with the highest priority.

Call Initiator

SIP trunk group from where the call is initiated. This item can be set to a specific SIP trunk group or SIP Trunk Group [ANY] which indicates any SIP trunk group.

CallerID Prefix, CalleeID Prefix

A string of numbers at the beginning of the calling/called party number. This item can be set to a specific string or “*” which indicates any string. These two configuration items together with Call Initiator can specify the calls which apply to a routing rule. Note: Multiple rules are supported for CallerID/CalleeID prefix. They are separated by “:”.

Call Destination

PCM trunk group to which the call will be routed.

Number Filter

Number filter rule which will be applicable to this route. It is set in Number Filter. See 3.10.4 Filtering Rule for details.

Description

More information about each routing rule.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-83 to modify a routing rule. See Figure 3-85 for the IPPSTN routing rule modification interface. The configuration items on this interface are the same as those on the Add New Routing Rule (IPPSTN) interface. Note that the item Index cannot be modified.

Figure 3-85 Modify Routing Rule (IPPSTN)

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To delete a routing rule, check the checkbox before the corresponding index in Figure 3-83 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all routing rules at a time, click the Clear All button in Figure 3-83.

3.9.3 PSTN to IP

Figure 3-86 PSTNIP Routing Rule Configuration Interface

See Figure 3-86 for the PSTNIP routing rule configuration interface. A new routing rule can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-87 for the PSTNIP routing rule adding interface.

Figure 3-87 Add New Routing Rule (PSTNIP)

The table below explains the items shown in the above figure.

Item

Description

Index

Call Initiator

PCM trunk group from which the call is initiated. This item can be set to a specific PCM trunk group or PCM Trunk Group [ANY] which indicates any PCM trunk group.

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CallerID Prefix, CalleeID Prefix

A string of numbers at the beginning of the calling/called party number. This item can be set to a specific string or “*” which indicates any string. These two configuration items together with Call Initiator can specify the calls which apply to a routing rule. Note: Multiple rules are supported in callerID/calleeID prefix. They should be separated by “:”.

Call Destination

SIP trunk group to which the call will be routed.

Number Filter

Number filter rule which will be applicable to this route. It is set in Number Filter. See 3.10.4 Filtering Rule for detailed setting.

Description

More information about each routing rule.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-86 to modify a routing rule. See Figure 3-88 for the PSTNIP routing rule modification interface. The configuration items on this interface are the same as those on the Add New Routing Rule (PSTNIP) interface. Note that the item Index cannot be modified.

Figure 3-88 Modify Routing Rule (PSTNIP)

To delete a routing rule, check the checkbox before the corresponding index in Figure 3-86 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all routing rules at a time, click the Clear All button in Figure 3-86.

3.10 Number Filter

Number Filter includes four parts: Whitelist, Blacklist, Number Pool and Filtering Rule. See Figure 3-89.

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Figure 3-89 Number Filter Interface

3.10.1 Whitelist

Figure 3-90 Whitelist Setting Interface

See Figure 3-90 for the Whitelist Setting Interface, which includes two parts: CallerID Whitelist and CalleeID Whitelist.

A new CallerID/CalleeID whitelist can be added by the Add New button. See Figure 3-91, Figure 3-92 for CallerID/CalleeID whitelist adding interface.

Figure 3-91 Add New CallerIDs in Whitelist Interface

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Figure 3-92 Add New CalleeIDs in Whitelist Interface

The table below explains the items shown in above figures.

Item

Description

Group

The corresponding Group ID for CallerIDs/CalleeIDs in the whitelist. The value range is 0~7.

No. in Group

The corresponding No. for different CallerIDs/CalleeIDs in a same group. It is allowed to set up to 100 numbers in one group.

CallerID

CallerID in the whitelist, which must be filled in with numbers or “*” (indicating any string) and can not be left empty. Example: 135*1 denotes any CallerIDs which start from 135 and end with 1 will be accepted.

CalleeID

CalleeID in the whitelist, which must be filled in with numbers or “*” (indicating any string) and can not be left empty. Example: 135*1 denotes any CalleeIDs which start from 135 and end with 1 will be accepted.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-90 to modify the CallerID or CalleeID whitelist. See Figure 3-93, Figure 3-94 for CallerIDs/CalleeIDs on the Whitelist Modification interface. The configuration items on this interface are the same as those on the Add New CallerIDs/CalleeIDs in Whitelist interface. The item Group No. cannot be modified.

Figure 3-93 Modify CallerIDs in Whitelist

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Figure 3-94 Modify CalleeIDs in Whitelist

To delete a CallerIDs/CalleeIDs in the whitelist, check the checkbox before the corresponding index in Figure 3-90 and click the „Delete‟ button. To clear all CallerIDs/CalleeIDs in the whitelist at a time, click the Clear All button in Figure 3-90.

Note: If a CallerID or CalleeID set in the whitelist is the same as one in the blacklist, it will go invalid. That is, the blacklist has a higher priority than the whitelist.

3.10.2 Blacklist

Figure 3-95 Blacklist Setting Interface

The Blacklist Setting interface is almost the same as the Whitelist Setting interface; only the whitelist changes to the blacklist. See Figure 3-95. The configuration items on this interface are the same as those on the Whitelist Setting interface (Figure 3-91, Figure 3-92).

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3.10.3 Number Pool

Figure 3-96 Number Pool Setting Interface

See Figure 3-96 for the Number Pool Setting interface. A new number pool can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-97 for the Number Pool adding interface.

Figure 3-97 Add New Number Pool

The table below explains the items shown in the above figure.

Item

Description

Group

The corresponding Group ID for numbers in the number pool. The value range is 0~15.

No. in Group

The corresponding No. for different numbers in a same group. It supports up to 100 number s in one group.

Number Range

The range of the numbers in a number Pool. It must be filled in with numbers and can not be left empty.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-96 to modify the number pool. See Figure 3-98 for the number pool modification interface. The configuration items on this interface are the same as those on the Add

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New Number Pool interface.

Figure 3-98 Modify Number Pool Interface

To delete a number pool, check the checkbox before the corresponding index in Figure 3-96 and click the „Delete‟ button. To clear all number pools at a time, click the Clear All button in Figure 3-96.

3.10.4 Filtering Rule

Figure 3-99 Filtering Rule Setting Interface

See Figure 3-99 for the Filtering Rule Setting Interface. A new filtering rule can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-100 for the Filtering Rule Adding interface.

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Figure 3-100 Add New Filtering Rule

The table below explains the items shown in the above figure.

Item

Description

No.

The corresponding number for a filtering rule. The value range is 0~99.

CallerID Whitelist

The Group No. of CallerIDs saved on the whitelist setting interface.

CalleeID Whitelist

The Group No. of CalleeIDs saved on the whitelist setting interface.

CallerID Blacklist

The Group No. of CallerIDs saved on the blacklist setting interface.

CalleeID Blacklist

The Group No. of CalleeIDs saved on the blacklist setting interface.

CallerID Pool in Whitelist

Select a Group No. which is set in the whitelist from the number pool as the CallerID pool in whitelist.

CallerID Pool in Blacklist

Select a Group No. which is set in the blacklist from the number pool as the CallerID pool in blacklist.

CalleeID Pool in Whitelist

Select a Group No. which is set in the whitelist from the number pool as the CalleeID pool in whitelist.

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CalleeID Pool in Blacklist

Select a Group No. which is set in the blacklist from the number pool as the CalleeID pool in blacklist.

Original CalleeID Pool in Whitelist

Select a Group No. which is set in the whitelist from the number pool as the original CalleeID pool in whitelist.

Original CalleeID Pool in Blacklist

Select a Group No. which is set in the blacklist from the number pool as the original CalleeID pool in blacklist.

Description

Remarks for the filtering rule. It can be any information, but can not be left empty.

After configuration, click Save to save the above settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-99 to modify the filtering rule. See Figure 3-101 for the filtering rule modification interface. The configuration items on this interface are the same as those on the Add New Filtering Rule interface.

Figure 3-101 Modify Filtering Rule Interface

To delete a filtering rule, check the checkbox before the corresponding index in Figure 3-99 and

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click the „Delete‟ button. To clear all filtering rules at a time, click the Clear All button in Figure 3-99.

3.11 Number Manipulation

Number Manipulation includes seven parts: IPPSTN CallerID, IPPSTN CalleeID, IPPSTN Original CalleeID, PSTNIP CallerID, PSTNIP CalleeID, PSTNIP Original CalleeID and CallerID Pool. See Figure 3-102.

Figure 3-102 Number Manipulation

3.11.1 IP to PSTN CallerID

Figure 3-103 IPPSTN CallerID Manipulation Interface

See Figure 3-103 for the IPPSTN CallerID manipulation interface. A new number manipulation rule can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-104 for the IPPSTN CallerID manipulation rule adding interface.

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Figure 3-104 Add IPPSTN CallerID Manipulation Rule

The table below explains the items shown in the above figure.

Item

Description

Index

The unique index of each number manipulation rule, which denotes its priority. A number manipulation rule with a smaller index value has a higher priority. If a call matches several number manipulation rules, it will be processed according to the one with the highest priority.

Call Initiator

SIP trunk group from where the call is initiated. This item can be set to a specific SIP trunk group or SIP Trunk Group[ANY] which indicates any SIP trunk group.

CallerID Prefix, CalleeID Prefix

A string of numbers at the beginning of the calling/called party number. This item can be set to a specific string or “*” which indicates any string. These two configuration items together with Call Initiator and With Original CalleeID can specify the calls which apply to a number manipulation rule.

With Original CalleeID

If this item is set to Yes, it indicates that the number manipulation rule is only applicable to the calls with original CalleeID/redirecting number. The default value is No.

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Stripped Digits from Left

The amount of digits to be deleted from the left end of the number. If the value of this item exceeds the length of the current number, the whole number will be deleted.

Stripped Digits from Right

The amount of digits to be deleted from the right end of the number. If the value of this item exceeds the length of the current number, the whole number will be deleted.

Reserved Digits from Right

The amount of digits to be reserved from the right end of the number. Only when the value of this item is less than the length of the current number will some digits be deleted from left; otherwise, the number will not be manipulated.

Prefix to Add

Designated information to be added to the left end of the current number.

Suffix to Add

Designated information to be added to the right end of the current number.

Description

More information about each number manipulation rule.

Note: The number manipulation is performed in 5 steps by the order of the following configuration items: Stripped Digits from Left, Stripped Digits from Right, Reserved Digits from Right, Prefix to Add and Suffix to Add.

After configuration, click Save to save the settings into the gateway or click Close to cancel the settings.

Click Modify in Figure 3-103 to modify a number manipulation rule. See Figure 3-105 for the IPPSTN CallerID manipulation rule modification interface. The configuration items on this interface are the same as those on the Add IPPSTN CallerID Manipulation Rule interface. Note that the item Index cannot be modified.

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Figure 3-105 Modify IPPSTN CallerID Manipulation Rule

To delete a number manipulation rule, check the checkbox before the corresponding index in Figure 3-103 and click the Delete button. Check All means to select all available items on the current page; Uncheck All means to cancel all selections on the current page; Inverse means to uncheck the selected items and check the unselected. To clear all number manipulation rules at a time, click the Clear All button in Figure 3-103.

3.11.2 IP to PSTN CalleeID

The number manipulation process for IPPSTN CalleeID is almost the same as that for IPPSTN CallerID; only the number to be manipulated changes from CallerID to CalleeID. See Figure 3-106 for IPPSTN CalleeID manipulation interface. The configuration items on this interface are the same as those on IPPSTN CallerID Manipulation Interface (Figure 3-103).

Figure 3-106 IPPSTN CalleeID Manipulation Interface

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3.11.3 IP to PSTN Original CalleeID

The number manipulation process for IPPSTN Original CalleeID is almost the same as that for IPPSTN CallerID; only the number to be manipulated changes from CallerID to Original CalleeID. See Figure 3-107 for IPPSTN Original CalleeID manipulation interface. The configuration items on this interface are the same as those on IPPSTN CallerID Manipulation Interface (Figure 3-103).

Figure 3-107 IPPSTN Original CalleeID Manipulation Interface

3.11.4 PSTN to IP CallerID

Figure 3-108 PSTNIP CallerID Manipulation Interface

See Figure 3-108 for the PSTNIP CallerID manipulation interface. A new number manipulation rule can be added by the Add New button on the bottom right corner of the list in the above figure. See Figure 3-109 for the PSTNIP CallerID manipulation rule adding interface.

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