D-Link DIV-140 User Manual

Table of Contents

Product Overview .......................................................................3

Introduction .........................................................................3

Features .............................................................................4

Hardware Overview ............................................................5

Front Panel - LEDs ........................................................5

Rear Panel - Connections .............................................6

Installation ...................................................................................7

Requirements .....................................................................7

Connection .........................................................................8

Conﬁguration ..............................................................................9

Web-based Conﬁguration .................................................10

Device Conﬁguration ...................................................12

Audio Conﬁguration .....................................................13

Call Conﬁguration ........................................................15

Network Conﬁguration .................................................17

Time Conﬁguration ......................................................18

NAT Conﬁguration .......................................................19

Media Conﬁguration ....................................................20

VoIP Conﬁguration ......................................................21

Hunt Group ..................................................................23

Destination ..................................................................24

ACR .............................................................................26

System Phone Book ....................................................28

System Speed Dial ......................................................29

Port Conﬁguration .......................................................30

Port Phone Book .........................................................32

Port Speed Dial ...........................................................33

Debug ..........................................................................35

Device Management ...................................................36

Upgrade History ..........................................................37

Call History ..................................................................38

Port Status ..................................................................40

Save Changes .............................................................. 41

Restart System............................................................. 42

Troubleshooting ......................................................................... 43

Warranty ..................................................................................... 48

Contacting Technical Support .................................................. 52

Registration ................................................................................ 53

D-Link DIV-140 User Guide 2

Product Overview

Introduction

The D-Link Analog Trunk Gateway DIV-140 is a full featured multi port SIP based VoIP gateway. It links traditional telephone network and IP network. Using the DIV-140 you can explore the wide features of VoIP technology. It enables you to make and receive calls through a low cost VoIP network using a conventional telephone network setup.

The advanced technology behind the DIV-140 presents a highly scalable and ﬂexible architecture to cater to the needs of varied market segments.

The highly ﬂexible call routing settings allow the administrator to conﬁgure different routes for calls based on the dialed number. It provides conﬁguration to route all local PSTN numbers through PSTN lines and all long distance calls through different VoIP networks. Both, proxy based calls and peer to peer calls, are supported simultaneously. It supports per port phone book and system phone book to place peer to peer calls without going through a VoIP server.

It supports per port speed dial lists and system speed dial lists to enable quick dialing to frequently accessed numbers. Also various call features can be individually conﬁgured for each port to different destinations.

With support to conﬁgure IP settings manually or use DHCP or PPPoE, the DIV-140 can work in different environments. Also, NTP is supported with standard time zone conﬁguration to keep the device time in sync with a server.

In VoIP server mode, each port can be conﬁgured with different account or the feature can be enabled to use same account for all ports. With backup server conﬁguration support, the device can still work in absence of primary server. The support for STUN and IP Sharing enables the device to work behind NAT.

It provides secure web based management interface with administrative and restricted accesses to remotely conﬁgure and monitor the system. The conﬁguration backup facility helps in maintaining multiple conﬁgurations for the device.

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Following features are supported.

Features

Protocols

• SIP RFC 3261 Compliance

• STUN and IP sharing support for NAT

• RTP and RTCP

• DTMF Tx/Rx support using SIP INFO or Out of Band as per RFC 2833.

• Fax Support using T.38 Fax Relay and Fax Passthrough

• NTP to keep device time in synchronization

• DNS/SRV RFC 3263 compliance

Features

• DTMF generation and detection

• Fax-tone detection

• Caller ID Detection and Generation

• Custom Caller ID Conﬁguration

• Detection and generation of user-speciﬁed tones

• Polarity Reversal Generation

Call Features

• Consultation Call Hold

• Attended and Unattended Call Transfer

• Call Forward (Always, on Busy, on No Answer)

• Call Waiting

• Hot line and Warm line

• ACR

• Hunt Group

• Speed Dial (System wide and per port)

• Phone Book (System wide and per port)

Audio Features

• G.729ab — G.729a with VAD and CNG support

• G.711 µ-law and A-law codec

• G.711 Annex 2. Support for VAD and CNG

• G.723.1 with 5.3 and 6.3 Kbps rates

Voice Quality

• IP layer TOS

• Dynamic/Adaptive Jitter Buffer algorithm

• Packet loss concealment (PLC)

• Echo Cancellation (G.168)

• Gain/Attenuation settings

Miscellaneous

• Forced Call disconnection from WEB/CLI/by dialing in key sequence

• Upgrade History, Call History, Port Statistics Display

• HTTP/TFTP software upgrade

• Conﬁguration Backup and Restore.

• Syslog Support

• Restore to Factory Default

D-Link DIV-140 User Guide 4

Hardware Overview

Front Panel - LEDs

Power LED

Will illuminate when the device is powered on.

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PSTN LEDs

These LEDs show the status of the FXO ports. If an LED is illuminated, the phone is off the hook, if it is not illuminated, the phone is on the hook. It will blink during a conversation or ﬁrmware upgrade.

LAN LED

Will blink to indicate activity on the LAN port.

Rear Panel - Connections

Console

Used to connect the DIV-140 to the Com Port on your computer.

Reset Button

Press to reset the DIV-140 to factory default settings.

LAN Port

Connects DIV-140 to Ethernet enabled device or switch.

PSTN Ports

Four (4) FXO ports used to connect to wall phone jacks.

Power Receptor

Receptor for Power Adapter.

D-Link DIV-140 User Guide 6

Installation

This section will explain how to connect your DIV-140 device to your network.

Requirements

The following items are required to conﬁgure the DIV-140:

1. Electrical outlet

2. PC with a serial port connection

3. Terminal emulator program (like Hyper Terminal)

4. Browser to access web interface. Following browsers are recommended: a. Internet Explorer version 6.0 and above b. Mozilla version 5.0 and above c. Netscape version 7.1 and above

5. VoIP Accounts or PSTN Lines

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Connection

A. Connect the power adaptor to power port of the DIV-140 and then connect other end of power adaptor to an

electric outlet.

B. Connect the Ethernet port of the DIV-140 to an Ethernet port on your hub or switch using a CAT5 Ethernet

cable.

C. Using the provided RJ-45 to DB9 console cable, connect the console port to your PC’s COM port.

D. Using telephone cables, connect the FXO ports to the telephone wall jacks.

E. Start a terminal emulator program on PC, such as Hyper Terminal. (115200 bps, 8 bit).

F. The power LED and Ethernet LEDs should both be illuminated.

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Conﬁguration

Once powered on, the DIV-140 will be at its default conﬁguration parameters. The default IP address of the device is 10.0.0.1. You can conﬁgure the device either through the web UI or Command Line Interface (CLI).

In order to use a web browser to conﬁgure the device, you must make sure the device has a valid Ethernet connection to a PC or LAN via its Ethernet port. We recommend using a recent version of the above mentioned browsers. The browser must have JavaScript enabled. You must make sure the PC is in the same IP subnet as this device (10.0.0.1). You can do this by changing the IP address of the PC. Type https://10.0.0.1 in the web browser address bar to access the web UI. Login with default user name and password as admin and password. Refer to the “Web Conﬁguration” section of this manual or online help for further details.

The network related parameters can also be conﬁgured through the CLI. Using a terminal emulator on your PC, you can modify these parameters as per your network settings.

For examples and details of commands used here you can refer to CLI Commands section of this manual. Type question mark ‘?’ to see the online help of command.

A. Enter user name as ‘admin’ and password as ‘password’. B. Type ‘help’ to see the list of available commands. C. Type ‘show nw’ to see the current network settings. D. Change the IP mode using command ‘set nw mode’. E. If IP mode is static IP use ‘set nw ip’ to set the new IP. F. If IP mode is PPPoE use ‘set pppoe auth’ to set the user name and password. G. Type ‘save’ to save the changes.

Now you can access the web interface of device using the new IP you have conﬁgured. Type https://<IPAddress> in your browser.

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Web-based Conﬁguration

To use the web-based conﬁguration utility you must use a computer either directly connected to the device using a cross-over Ethernet cable or connected to the same network as the DIV-140. The default IP address is 10.0.0.1. If your network is using a different IP range, you must ﬁrst statically assign your computer an IP address of 10.0.0.2 (subnet mask - 255.0.0.0) and change the IP address of the DIV-140 to an available IP address of your network.

To change your computer’s IP address, please refer to the Networking Basics section in this user guide.

Open a web browser such as Internet Explorer and enter https://10.0.0.1 and press Enter.

Note: You must enter HTTPS instead of HTTP.

If a security warning appears, click Yes to continue.

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When the login window appears, enter the user name admin and then your password. The default password is password. Click OK to continue.

The System Information page will appear. This will display information regarding your DIV-140 device such as network settings and ﬁrmware version.

The left column will contain the navigation menu. The plus (+) sign will indicate the section contains a sub-menu.

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Device Conﬁguration

Country:

Feature Code:

Speed Dial:

Auto Log Off Time:

Select your country from the drop-down menu.

Enter the feature code.

Enter the speed dial number.

Enter the time (in minutes) to automatically log off.

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Audio Conﬁguration

Codec Priority:

Jitter Buffer Delay:

Packet Loss Rate:

Here you can select the priority of audio codecs. While establishing a call, the DlV-140 will negotiate the highest priority of codec acceptable to both parties. The administrator can choose from four codecs. G.711 Mu and G.711 A provide the best voice quality but consume the most bandwidth. G.723 consumes less bandwidth with some compromise on quality. G.729 is a compromise between both. It uses signiﬁcantly less bandwidth as compared to G.711 and gives better quality as compared to G.723.

The Jitter Buffer monitors network conditions and implements a delay proﬁling algorithm that provides better tracking of adaptive jitter buffer and improves voice quality. The jitter buffer automatically determines the network jitter, maintains the desired packet loss rate, subject to the limit of the maximum allowed jitter delay. These parameters can be conﬁgured here.

The Jitter Buffer Delay speciﬁes the maximum desired jitter delay in milliseconds. Range: 0 to 500 milliseconds. Default: 150 milliseconds.

Speciﬁes the allowable packet loss rate in 0.1% units. Range: 0 to 50 (corresponding to 0.1 % to 5.0 %). Default: 1 (0.1%).

Gain Settings:

The volume settings at various levels can be conﬁgured here. Here Tx ﬁeld speciﬁes stream going from this device to IP network and Rx speciﬁes the stream coming to this device from the IP network.

VoIP: Increase this value to digitally amplify the signals. Range: -40 dB to 15 dB. Default: 0 dB.

PSTN: Increase this value to amplify the signals at analog level. Range: 0 or 1 (corresponding to 0 and 6 dB). Default: 0.

Tone: Modify this value to set the volume level of tones. Range: -20 dB to 15 dB. Default: -12 dB.

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Fax Pass Through:

In this mode of fax communication, G711 codec is used. Device will automatically switch to G711 codec on detection of FAX tone from the FAX machine.

Fax Relay:

FEC:

Packet Redundancy:

Elipsis:

Echo Tail Length:

Echo Delay

Compensation:

VAD:

DTMF Mode:

In this mode of fax communication, T.38 codec is used. Device will automatically switch to T.38 codec on detection FAX tone from the FAX machine. If both the modes of fax communication are selected then preference will be given to Fax Relay mode. Following (FEC, Packet Redundancy Level, and Ellipsis) are conﬁgurable parameters for Fax Relay features.

This feature can be selected to activate forward error correction in T.38 Fax Relay communication.

This feature can be selected to activate packet redundancy in T.38 Fax Relay communication.

This feature can be enabled to activate newer version of T.38 Fax protocol corrigendum 1. By enabling this feature, device will be able to work with newer version fax devices.

The device cancels the echo generated by the hybrid of local telephone interface and phone set so that the other party connected to the channel will not hear the echo. The parameters associated with echo cancellation can be conﬁgured here. Note: The default values should be good enough for most cases and it is recommended that you don’t change it without assistance.

Echo Tail Length: This speciﬁes the time span over which echo is expected to arrive. Tail length of 4 ~ 8 milliseconds is good enough for most cases. Range: 2 to 64 in steps of 2 (corresponding to 1 to 32 milliseconds). Default: 36 (i.e. 18 milliseconds).

This is used to properly align the delayed signal with reference signal. Range: 0 to 240 (corresponding to 0 to 30 milliseconds). Default: 0 (i.e. 0 milliseconds).

When this is enabled, the DIV-140 detects a silence interval and uses silence compression to save on bandwidth. This feature works irrespective of the codec selected. Default: Enabled.

There are four modes in which DTMF can be transmitted.

• In Band Mode

• Out Of Band Mode

• Info Mode

• Auto Mode.

In In Band Mode, DTMF packets are sent in band. In Out Of Band Mode, DTMF packets are sent as RTP Packets with different payload type as conﬁgured in DTMF Payload ﬁeld. In Info Mode, DTMF packets are sent as SIP INFO Packets and in Auto Mode the mode of DTMF transmission is selected automatically to In Band, Out Of Band or Info according to the remote ends capability. Default: Auto Mode.

DTMF Pay Load:

Enter the payload type to use for out of band DTMF packets. Range: 96 to 127. Default: 101.

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Call Conﬁguration

Timeout:

Dial Timeout:

Connection Timeout:

Alerting Timeout:

Interdigit Timeout:

Interdigit Time

while Dialing:

Various constants which are used during the call setup (like maximum timeout or number of rings) can be conﬁgured in this section.

Enter the maximum time to wait for a user to dial any key after going off hook. After this timeout is reached an error tone is played to the user. Range: 5 to 120 seconds. Default: 15 seconds.

Enter the maximum time to wait to get any response from the other end. During this period a progress tone is played and after the timeout limit is reached an error tone is played to the user. Range: 5 to 180 seconds. Default: 45 seconds.

Enter the maximum time to play a ring back tone for outgoing calls when the called party starts ringing. For incoming calls this ﬁeld deﬁnes how long the phone will ring when user doesn’t answer. Range: 5 to 120 seconds. Default: 60 seconds.

Enter the maximum time to wait in between digits while dialing the number. After this timeout, further digits are ignored and the collected digits will be dialed out. Range: 1 to 120 seconds. Default: 5 seconds.

Enter the time between digits while dialing out the received number on FXO line using DTMF tones. Range: 50 to 1000 milliseconds. Default: 300 milliseconds.

FXO Rings:

Dial Termination:

Character Termination:

Enter the number of rings to wait before going off hook on FXO port when the FXO port is ringing. Range: 1 to 9 rings. Default: 2.

Note: If this ﬁeld is conﬁgured as 1 then caller ID detection on the FXO port won’t work.

Here you can conﬁgure if any special DTMF key like * or # to use for terminating the dialed number while placing outgoing call. Note that Inter Digit timeout will be always active.

Select enable if you want to terminate the dialed number using a DTMF key, otherwise select Disable. Default: Enable. If character termination is enabled then you can also terminate the feature access keys with this.

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For example, the feature key sequence to initiate call hold feature is *111. If character termination is enabled to # then you can dial *111# for the same.

Character:

Default Route:

Route all calls to

VoIP Server:

Enter the character to use for termination. Range: 0 to 9, * or #. Default: #

Here you can conﬁgure how calls should be routed.

Enable this if all calls including calls to local FXO ports should be routed to the VoIP server. Default: Enable. Note: If this ﬁeld is enabled, then the calls to local FXO ports will fail if the VoIP server is not available.

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