Dinstar DAG1000-8FXS User manual

DAG1000-4FXS Voice Gateway

User Manual V2.0

File Name

Document Version

Firmware Version

Date

Revised by

Revision Records

DAG1000-4FXS Voice Gateway User Manual

2.0

20.03.03.05

2012/03/16

Technical Support Department

Table of Contents

1.

Equipment Introduction ....................................................................... 1

1.1 Overview ....................................................................................... 1

1.2 Equipment Structure ...................................................................... 1

1.3 Installation .................................................................................... 2

1.4Network Applications ...................................................................... 3

1.5 Functions and Features .................................................................. 3

1.5.1 Protocol standard supported ................................................. 3

1.5.2 Voice and Fax parameters ..................................................... 4

1.5.3 Supplementary service ......................................................... 4

2.

Basic Operations ................................................................................. 4

2.1 Phone Call ..................................................................................... 4

2.1.1 Phone or Extension Number .................................................. 4

2.1.2 Direct IP Calls ...................................................................... 4

2.2 Call Hold ....................................................................................... 5

2.3 Call Waiting ................................................................................... 5

2.4 Call Transfer .................................................................................. 6

2.4.1 Blind Transfer ....................................................................... 6

2.4.2 Attended Transfer ................................................................. 6

2.4.3 3-way Conference ................................................................ 7

2.5 Call Features ................................................................................. 7

2.6 Sending and Receiving Fax ............................................................. 9

2.6.1 DAG（FXS）support four fax modes:..................................... 9

2.6.2 T. 38 and Pass-Through ........................................................ 9

3.

Local IVR Operation ............................................................................. 9

3.1 Inquire IP address ......................................................................... 9

3.2 Factory Reset ................................................................................ 9

3.3 Configure LAN Port’s IP Address ..................................................... 9

4.

WEB Configuration ............................................................................ 10

4.1 WEB Login .................................................................................. 10

4.1.1 Login ................................................................................. 10

4.1.2 Login WEB ......................................................................... 11

4.2 Navigation Tree ........................................................................... 12

4.3 State and Statistics ...................................................................... 13

4.3.1 System Information ............................................................ 13

4.3.2 Registration Information ..................................................... 14

4.3.3 TCP/UDP Statistics ............................................................. 14

4.3.4 RTP Session Statistics ......................................................... 14

4.4 Quick Setup Wizard ..................................................................... 15

4.5 Network Configuration ................................................................. 15

4.5.1 Local Network .................................................................... 15

4.5.2 VLAN Parameter ................................................................. 17

4.5.3MAC Clone(Routing mode will be optional) ............................ 19

4.5.4 DHCP Server (Routing mode will be optional) ....................... 20

4.5.5 DMZ Host (Routing mode will be optional) ........................... 21

4.5.6 Forward Rule(Routing mode will be optional) ....................... 21

4.5.7 Static Route Table .............................................................. 22

4.5.8 ARP ................................................................................... 23

4.6SIP Server .................................................................................... 23

4.7 Port Configuration ........................................................................ 25

4.8 Advanced .................................................................................... 26

4.8.1 FXS parameters.................................................................. 26

4.8.2 Media Parameter ................................................................ 28

4.8.3 SIP Parameter .................................................................... 29

4.8.4 Fax Parameter .................................................................... 32

4.8.5 Digit Map ........................................................................... 33

4.8.6 Feature Codec .................................................................... 35

4.8.7 System Parameter .............................................................. 37

4.9 Call & Routing ............................................................................. 38

4.9.1 Port Group ......................................................................... 38

4.9.2IP Trunk ............................................................................. 40

4.9.3 Routing Configuration ......................................................... 41

4.9.4 IP-Tel Routing .................................................................... 41

4.9.5 Tel-IP/Tel Routing ............................................................... 42

4.10 Manipulation Configuration ......................................................... 43

4.10.1 IP-Tel Callee ..................................................................... 43

4.10.2 Tel-IP Caller ..................................................................... 44

4.10.3 Tel-IP Callee ..................................................................... 45

4.11 Maintenance .............................................................................. 45

4.11.1 syslog Parameter .............................................................. 45

4.11.2 Firmware Upload .............................................................. 46

4.11.3 Data Backup .................................................................... 47

4.11.4 Data Restore .................................................................... 47

4.11.5 ping Test .......................................................................... 47

4.11.6 Tracert Test ...................................................................... 48

4.11.7 Password Modification ...................................................... 49

4.11.8 Factory Reset ................................................................... 50

4.11.9 Device Restart .................................................................. 50

5.

Glossary ........................................................................................... 51


1. Equipment Introduction

1.1 Overview

Thanks for purchasing Dinstar DAG1000-4FXS (hereinafter referred to as the DAG) analog voice gateway.DAG1000-4FXS analog gateway is access gateway based on IP network. It can provide low cost, simple operation VoIP solutions for small enterprise, the family office, remote office and branch enterprise. DAG connects to analog telephone, fax and traditional analog PBX with standard voice interfaces and provided high quality voice service.

DAG1000/2000 series VoIP access gateway adopted standard SIP protocol and compatible with leading IP PBX, soft-switch and SIP-based platform. DAG1000/2000 series FXS analog gateway includes following model:



DAG1000-4S



DAG1000-8S



DAG2000-16S



DAG2000-32S

This manual mainly to DAG1000-4S as examples, introduce the function of devices and parameter configuration.

1.2 Equipment appearance

Figure 2-3 DAG1000-4S Figure 2-4 DAG1000-8S

Dinstar Technologies Co., Ltd.

1


DAG2000-16S DAG2000-32S

1.3 Power supply

DAG1000/2000 adopts AC 110-240 V power supply, with the power adapter convert to

12VDC power.

Power parameters:

Input: 100-240V,50-60Hz

Output: 12VDC

Notes: Because power adapter interface is different in different country, please confirm the interface standard with us before shipment.


2


1.4Network Applications

Figure 4-1：Network Applications

1.5 Functions and Features

1.5.1Protocol standard supported

•

SIP V2.0 (RFC 3261,3262,3264)

•

SDP (RFC 2327)

•

REFER (RFC 3515)

•

RTP/RTCP (RFC 1889,1890)

•

STUN (RFC 3489)

•

ARP/RARP (RFC 826/903)

•

SNTP (RFC 2030)

•

DHCP/PPPoE

•

TFTP/HTTP/HTTPS

•

DNS/DNS SRV (RFC 1706/RFC 2782)

•

VLAN 802.1P/802.1Q


3


1.5.2 Voice and Fax parameters

•

G.711A/U law, G.723.1, G.729AB

•

Comfortable Noise Generation (CNG)

•

Voice Activity Detection (VAD)

•

Echo Cancellation (G.168)

•

Adaptive Dynamic Jitter Buffer

•

Voice and fax gain control

•

Modem

•

T.38/Pass-through

•

DTMF Mode: Signal/RFC2833/INBAND

1.5.3 Supplementary service

•

Call waiting

•

Call transfer（Blind transfer, Attend transfer,）

•

Quick pick

•

Call Forwarding Unconditional

•

Call Forwarding on No Reply

•

Hotline

•

Call hold

•

DND

•

3-way conference(1/2/4 port support)

•

Voice mail

•

Direct IP Call

2. Basic Operations

2.1 Phone Call

2.1.1 Phone or Extension Number

1）Dial the number directly and wait for 3 seconds (Default “

No dial timeout

”);

2）Dial the number directly and press #.

2.1.2 Direct IP Calls

DAG series device with FXS port allow two parties directly call through IP address. The user

4


DAG1000-4FXS Voice Gateway User Manual need only a simulation with the FXS port unit equipment linked together and set up calls not registered.

Elements necessary to completing a direct IP call：

1) Both DAG serial and other VoIP Device, have public IP addresses;

2) Both DAG serial and other VoIP Device are on the same LAN using private IP addresses;

3) Both DAG serial and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).

Operation Process：

1）Pick up the analog phone then dial “*47”

2）Enter the target IP address.

【

Note】：No dial tone will be played between step 1 and step 2

Examples:

If the target IP address is 192.168.0.160, the dialing convention is *47, then

192*168*0*160. Followed by pressing the “#” key or wait 3 seconds. Complete signaling interactive soon after, he was called the unit can be heard ringing.

【

Note】：You cannot make direct IP calls between FXS0 to FXS1 since they are using same

IP. It only supports the default destination port 5060.

2.2 Call Hold

Place a call on hold by pressing the “flash” button on the analog phone (if the phone has that button).Press the “flash” button again to release the previously held Caller and resume conversation. If no “flash” button is available, use “hook flash” (toggle on-off hook quickly).

You may drop a call using hook flash.

2.3 Call Waiting

Call waiting tone (3 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle between incoming call and current call by pressing the “flash” button. First

5


DAG1000-4FXS Voice Gateway User Manual call is placed on hold. Press the “flash” button to toggle between two active calls.

2.4 Call Transfer

2.4.1Blind Transfer

Blind transfer used to transfer call to the third party without inform caller. Assume that call

Caller A and B are in conversation. A wants to Blind TransferB to C:

1）Caller A presses FLASH on the analog phone to hear the dial tone;

2）Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds);

3）Caller A will hear the confirm tone. Then, A can hang up.

Note：

“

Call features enable

” must be set to “Yes” in web configuration page. Caller A can place a call on hold and wait for one of three situations:

1）A quick confirmation tone (similar to call waiting tone) followed by a dial-tone. This indicates the transfer is successful. At this point, Caller A can either hand up or make another call.

2）A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed.

3）Continuous busy tone. The phone has timed out.

2.4.2 Attended Transfer

Attended transfer allowsusers to confirm the third party response and decide whether to answer the calls and then transfer this call to the third party.

Assume that Caller A and B are in conversation. Caller A wants to

Attend Transfer

B to C:

1）Caller A presses FLASH on the analog phone for dial tone;

2）Dial Caller C’s number followed by # (or wait for 3 seconds);

3）If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to complete transfer;

6



4）If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller

B.

2.4.3 3-way Conference

3-way conference：

1）Caller A call B,

B pick up into call states

;

2）Caller A hook flash, A and B into keep states, then C call A, A through to the phone.

3）A hook flash, then A、B、C into keep states, at this time if A press 1 key, then A and B continue to call

; if A press 2 key, then A and B continue to call; if A press 3 key, then A、B、

C three parties go to call.

Figure 2.4-1：3-way Conference

2.5 Call Features

DAG (FXS) support all traditional and senior phone function.

Table 2.5-1 Feature Codec

Feature Codec

*158#

Operation Instructions

View the LAN port IP address

*159#

*114#

View the WAN port IP address

Inquire port account


7

*90*

*91#

*92*

*93#

*78#

*79#

*200#

*111#

*#

*47*

*51#

*50#

*87*

*72*

*73#

*150*

*157*

*152*

*153*


Set the way of obtain IP address

Set network method

Set IP address

Set Subnet mask

*156*

*193#

Set default gateway IP address

Obtain IP address through DHCP again

*160*1#

Open WAN port to access web

*166*000000#

Factory reset

Restart device

Call hold

IP address call

Enable call waiting

Disable call waiting

Blind transfer

Enable Unconditional Call Forward

Disable Unconditional Call Forward

Flash/Hook

Enable Busy Call Forward

Disable Busy Call Forward

Enable No Answer Call Forward

Disable No Answer Call Forward

Enable DND

Disable DND

Access Voice mail

Switch between incoming calls, If not in session, flash/hook will switch a new channel for new call.

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2.6 Sending and Receiving Fax

2.6.1 DAG（FXS）support four fax modes:

1）T.38（FoIP）

2）Pass-Through

3）Modem

4）adaptive

2.6.2 T. 38 and Pass-Through

T.38 is the preferred method because it is more reliable and works well in most network conditions. If the service provider supports T.38, please use this method by selecting T.38 as fax mode (default). If the service provider does not support T.38, pass-through mode may be used. If you have problems with sending or receiving Fax, toggle the Fax Tone

Detection Mode setting.

3. Local IVR Operation

3.1 Inquire IP address

Analog phone connected with FXS ports of device, then pick up, after dial tone, dialing

*158# to inquire LAN port IP address and dialing *159# to inquire WAN port IP address.

3.2 Factory Reset

After picking up, dial *166*000000#, then onhook and restart after “Setting successful”.

3.3 Configure LAN Port’s IP Address

9


DAG1000-4FXS Voice Gateway User Manual network;

（

3）Telephone is connecting to FXS port of device.

1) Configure dynamic IP address by DHCP：

Offhook; Dial “*150*2#”; Onhook;

If the equipment hint success, after 10 seconds, and restart the equipment.(Power-off then power-on)

2) Configure Static IP address

Offhook; Dial “*150*1#”; Onhook;

Then configure IP and mask as follow:

•

Configure IP address：

Offhook; input “*152*172*16*0*100# ”; onhook

•

Configure subnet mask：

Offhook; input “*153*255*255*0*0# ”; onhook

•

Configure gateway IP address

Offhook; input “*156*172*16*0*1# ”; onhook.

3) Query the IP address of device: Offhook, input“*158#”

4) If the DAG serial uses PPPoE method to get IP address，it need to configure by web browser.

【

Note】：the telephone will play voice prompt “Setting successfully” if the step is correct

4. WEB Configuration

4.1 WEB Login

Device is connecting to network properly, refer to chapter 3 “Operation”. Offhook and dial*158# to inquire device IP address.

4.1.1 Login

Device LAN port default IP address is 192.168.11.1, WAN port default obtain IP address by

DHCP. Advice to modify the IP address of the local computer equipment and ensure that

10


DAG1000-4FXS Voice Gateway User Manual are on the same IP segment, with Windows 7 as an example, the local computer IP address change for 192.168.11.10：

Figure 4.1-1Modify IP address

Check connection between computer and device, click “Start”-> “run”-> input “cmd”, run ping 192.168.11.10 –t order to check the connectivity between them.

4.1.2 Login WEB

Open web browser, then input IP address of device, Press“Enter”, it pop up logging on identity authentication interface.


Figure 4.1-1 DAG FXS Login Interface

11


Default username and password: admin/admin, click “OK” to entry into web interface.

Figure 4.1-2 DAG Configure Interface

4.2 Navigation Tree

DAG series voice gateway web configuration interface mainly includes navigation tree and the right configuration interface. Choose navigation tree in order to entry into the configuration interface.

Figure 4.2-1 Navigation Tree

When device is in bridge mode, navigation tree won’t display "routing configuration" items and the following "DHCP service", "DMZ host", "forward rules" and "static routing" and

"ARP" etc.

12



4.3 State and Statistics

4.3.1 System Information

System information interface shows the run information as following figure 4.3.1 below:

System information as follow:

Figure 4.3-1 System Information

Table 4.3-1 System Information Description

MAC address

Network Mode

Network

WAN Port

LAN Port

WAN port hardware address. The device ID in HEX format.

Display network mode, include bridge and rout.If it is bridge, WAN port display

Network, and the WAN port as same as the LAN port.

Display WAN and LAN port IP address, subnet mask and the way of obtain IP address.

Shows WAN IP address of DAG ,

DHCP mode: all the field values for the Static IP mode are not used (eventhough they are still saved in the Flash memory.) The DAG acquires its IPaddress from the first DHCP server it discovers from the LAN it is connected.

Using the PPPoE feature: set the PPPoE account settings. The DAG willestablish a

PPPoE session if any of the PPPoE fields is set.

Static IP mode: configure the IP address, Subnet Mask, Default Router IPaddress,

DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fieldsare set to zero by default.

Shows LAN IP address of DAG.if network Mode is bridge, LAN port won’t display.

DNS Server Display DNS server IP address and default gateway information

System Uptime

Network

Statics

Version

Traffic

Time elapsed from device power on to now.

Total bytes of message received and sent by network port.

Includes: product mode, software version, hardware version and built time etc.

13



4.3.2 Registration Information

Figure 4.3-2 Port and Port group registration information

4.3.3 TCP/UDP Statistics

Figure 4.3-3 TCP/UDP Statistics Information

Figure 4.3-3 shows TCP sending and receiving, UDP sending and receiving packets of statistical information since the device launched.

4.3.4 RTP Session Statistics

Figure 4.3-4 RTP Session Statistics

Figure 4.3-4 display real-time RTP conversation flow data information, includes:

Port, voice codec, packet period, local port, peer IP, peer port, sent packets, receive packets, lost packets, jitter and duration.

14



4.4 Quick Setup Wizard

Quick configuration guide will guide users to configure the device step by step. Users only need to configure network, SIP server and sip port in quick setup wizard. Basically, after these three steps, users are able to make voice call through device.

4.5 Network Configuration

4.5.1 Local Network

DAG has two kinds of work mode: route and bridge. When DAG is set rout mode, the DAG will work as small router and NAT function has enabled. In this situation, WAN port is normally connect to uplink router/switch or ADSL MODEM, LAN port used to connect local computer or other network device(such as Ethernet switches, Hubs etc); When DAG is set bridge mode, WAN and LAN port are the same. The DAG just work as two ports or four ports Ethernet switch.

When it set to bridge mode, only need to configure WAN port IP address and DNS.If set to route mode, default LAN port IP will display and it can be change by users.Network configure interface as below:

15



Figure 4.5-1Route Mode

Figure 4.5-2 Bridge Mode

•

“Link Speed &Duplex”used to select Ethernet port work mode, include 5 kinds of choice,“Auto Detect”

、

“10Mbps half-duplex”

、

“10Mbps

16


DAG1000-4FXS Voice Gateway User Manual full-duplex”,“100Mbpshalf-duplex”,“100Mbps full-duplex”,default is “Auto Detec”.

•

When select“Obtain IP address automatically”, DAG will obtain IP address by DHCP.

•

When select “Use the following IP address”, that configure DAG to fixed IP address mode.

•

When select “PPPoE”, please fill in account and password offered by ISP in internet account and password.

【

Notes】：

1） If select DHCP to obtain IP address, please ensure DHCP server in network and work normally.

2） Under route mode, please configure LAN port and WAN port in different segment, otherwise DAG can’t work normally.

3） Under route mode, login DAG configuration interface only used LAN port.

4） After configuration, restart device configuration validation.

4.5.2 VLAN Parameter

Generally, Internet provides only Best Effort Service. Since ethernet is the most spread LAN access technology, importance of providing it a quality of service mechanism ought not to be neglected.

Ethernet technology also used as WAN technology, not only as LAN technology. Due to rapidly increasing use Internet through Public Switched Telecommunication Network

(PSTN), Telephone Companies are forced to implement IP-based networks as their PSTN backbones. A network like this without any Quality of Service mechanisms would be disastrous. Just imagine yourself trying to get an emergency call through while others just surf the Internet.

1) 802.1Q

The IEEE 802.1Q standard defines architecture for Virtual Bridged LANs, the services provided in Virtual Bridged LANs and the protocols and algorithms involved in the provision of those services.

No Quality of Service mechanisms are defined in this standard, but an important

17


DAG1000-4FXS Voice Gateway User Manual requirement for providing QoS is included in this standard, e.g. abitity to regenerate user priority of received frames using priority information contained in the frame and the User

Priority Regeneration Table for the reception Port.

2) 802.1p

IEEE 802.1p standard, Traffic class expediting and dynamic multicast filtering. It describes important methods for providing QoS at MAC level. IEEE 802.1p is in fact quite good.

Lower priority level packets are not sent, if there is packets in queued in higher level queues.IEEE 802.1p describes no admission control protocols. It would be possible to give

Network Control priority to all packets and the network would be easily congested.

There are three VLAN: data VLAN, voice LAN and management VLAN. VLAN configuration interface as following figure 4-4-3:


Figure 4.5-3 VLAN parameter configuration

Table 4.5-1VLAN parameter configuration

18

Data VLAN

Voice VALN


Data 802.1Q VLAN ID(0-4095)

Data 802.1p Priority（0-7）

Voice 802.1Q VLAN ID(0-4095)

Voice 802.1p Priority（0-7）

IP address

Fill out an ID to describe a data VLAN group,ID 0 used to management VLAN, can’t used to service configure.

802.1 protocol to control network traffic priority,

Priority from 0-7.

Fill out an ID to describe a voice VLAN group, ID 0 used to management VLAN, can’t used to service configure.


Priority from 0-7.

Can use dynamic or static IP address

Management

VLAN

Voice VLAN DNS Server Can use dynamic or static DNS server address

Management 802.1Q VLAN

ID(0-4095)

Management 802.1p Priority

（

0-7）

Fill out an ID to describe a data VLAN group, ID 0 used to management VLAN, can’t used to service configure.


Priority from 0-7.

IP address Can use dynamic or static IP address

Management VLAN DNS server Can use dynamic or static DNS server address

【

Note】：restart the device to take configuration effect.

4.5.3 MAC Clone(Routing mode)

Figure 4.5-4 MAC Clone Interface

More client in LAN have already can’t share internet used the traditional “gateway set law”.

Because IP address binding in only a legitimate MAC address by ISP. If the ISP’s switch discover illegal MAC address, it will refuse service.

The best way is MAC clone for MAC binding. Most ADSL MODEM, broadband router,

19


DAG1000-4FXS Voice Gateway User Manual wireless router have this feature. The principle of MAC address clone is deliberately exposed MAC address of bound computer to the ISP server and let the ISP server think that used only a single piece of computer, in fact many computers in sharing the Internet.

This function used to prevent ISP limiting to share the Internet.

【

Note】：restart device to take configuration effect.

4.5.4 DHCP Server (Routing mode)

Under route mode, DAG network part as a small router to configure DHCP service, that

DAG as a DHCP server in network.

Start and end address of address pool determine the range of IP address automatically assigned to other devices;

•

IP Expire Time means use time of assigned IP address. More than the lease time, if the

IP address is not used by network equipment, IP address will be recovered;

•

Subnet mask, gateway, DNS and other information configured by DHCP protocol.

Configuration interface as figure 4.5-5：

Figure 4.5-5 DHCP Configuration Interface

【

Note】：When configure start and end IP address, subnet mask and gateway, please set the same segment with LAN port. Otherwise, device will not work normally. After configuration, restart device configuration validation.

20



4.5.5 DMZ Host (Routing mode)

DMZ(Demilitarized Zone) connect web, e-mail etc. server allowed external to access to this area. Make the internal network located the back of the zone of confidence and not allow any access, separation of inside and outside the network, protect user information.DMZ can be understood that a special areas of the network and different from the external network or intranet. Public server that does not contain confidential information usually placed in DMZ, such as web, Mail, FTP etc. Accuser from intranet can visit the service of

DMZ, but can’t come into contact with confidential or private information stored in the network. Even if DMZ server is damaged, it will not be confidential information in the internal network .

Figure 4.5-6 DMZ Configuration Interface

【

Note】：After configuration, restart device configuration validation.

4.5.6 Forward Rule(Routing mode)

In some cases, LAN network equipment need to provide some communication in WAN network (such as port for 21 FTP service), This time can be configured forwarding rules for the network equipment.

Service ports namely the need to provide service network mouth WAN ports, IP address that LAN network provide services to the mouth of the network equipment IP address, the protocol is TCP or UDP.

The different between forward rule and DMZ host is that DMZ Host offers continuous multiple

Port (0-1024) and all the foreign communication agreement; while the forward rule offers a

21


DAG1000-4FXS Voice Gateway User Manual single or a few port foreign communication on some protocol. When the conflicts exist between forward rule and DMZ host,the configuration of forwarding rules is preferred.

Forward rule configuration interface as follows:

Figure 4.5-7 Forward rule configuration interface

4.5.7 Static Route Table

Static Route Table is IP communication direction in network, generally do not need to configure static route. When there are many segments in LAN network and need to complete some specific application among these segments, the static route need to be configured.

Static Route configuration interface as follows:

Figure 4.5-8 Static route configuration interface


22


4.5.8 ARP

ARPbrief introduction:

ARP is address resolution protocol. After configuring ARP, users can get physical address through device IP address. Under TCP/IP network environment, each host is assigned a

32-bit IP address. But the message transmission needs to know the purpose the physical address of the party. ARP is a tool that converts IP address into MAC address.

ARP configuration interface as follows:

Figure 4.5-9 ARP Parameters

4.6 SIP Server

SIP server introduction:

1）SIP server is the main component of VoIP network and responsible for establishing all the SIP phone calls. SIP server also called SIP proxy server or regist ered server.

IPPBX and the soft-switch can act as SIP server role.

2）Usually, SIP server does not participate in the media process.

In SIP network, the media always using end-to-end to hand the consultation. In some particular situation or business processing, such as “Music On Old”, SIP server will actively participate in the media negotiation. Simple SIP server is responsible only for establishment, maintenance and cleaning conversation, don't interfere in call. While relatively complex SIP server also called SIP PBX. It not only provides the basic call, and basic conversational support, also offer plenty of business, such as: Presence,

Find-me, Music On Hold.

3）SIP server based on Linux platform, such as: OpenSER、sipXecx，VoS，Mera etc.

4）SIP server based on windows platform, such as :miniSipServer、Brekeke，VoIPswitch etc.

23



5）Carrier grade soft-switch platform, such as Cisco, Huawei, Zteetc.

SIP server configuration interface as follows:

Figure 4.6-1 SIP Server Configuration Interface

SIP parameter description:

Primary SIP Server IP SIP Server IP address or Domain name provided by VoIP service provider.

Primary SIP Server port

Register interval

Service port, default is 5060 protects registrar against excessively frequent registration refreshes while limiting the state.Every once in a while send request for registration to the terminal server, default is 1800s.

Heartbeat Heartbeat message detect the connection status between device and SIP server.

Secondary SIP Server IP address

Backup SIP Server’s IP address or Domain name provided by VoIP service provider.

Secondary SIP Server port

Secondary SIP

Register interval

Service port, default is 5060 server protects registrar against excessively frequent registration refreshes while limiting the state.Every once in a while send request for registration to the terminal server, default is 1800s.

Secondary SIP heartbeat Heartbeat message detect the connection status between device and SIP server.

Use Random Port

Set Local SIP port

Random SIP service ports for DAG

Default SIP service port is 5060.

24



4.7 Port Configuration

Port parameters include: Send gain, receive gain, primary display name etc.

Figure 4.7-1 Port configuration interface

Port parameters introduce as follows:

Tx Gain

Rx Gain

Primary /Secondary SIP

Display Name

It is use to control the volume of conversation, Adjust "TX gain" will affect the end users voice size, the default value is 0.

Its value range from-10 – 10 dB

It is use to control the volume of conversation, Adjust "RX gain" will affect the end users voice size, the default value is 0.

Its value range from -10 – 10 dB

Primary /Secondary SIP account description,Its purpose is so you can identify the SIP account with a meaningful name

Primary /Secondary User account information, provided by VoIP service provider (ITSP). Usually in

25


SIPUser ID

DAG1000-4FXS Voice Gateway User Manual the form of digit similar to phone number or actually a phone number.

Primary/Secondary SIP

Authenticate ID

Primary/Secondary

Authenticate password

SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID.

SIP password which registers to soft switch/SIP server

Offhook Auto-dial

Auto-dial Delay Time

Pre-assign an extension or phone number so that automatically dial a number as soon as you pick up the phone set

Delay 0-3 seconds to automatically dial a number, 0 means dial number immediately

DND Do not disturb, the phone set won’t receive any calls in case it enabled

Caller ID Enable or disable caller ID for corresponding port

Number for CFU

Number for CFB

Number for CFNRy

Call Waiting call forward unconditional, all incoming calls willforward to pre-assigned number automatically

Call forward on busy, if the line is busy, the call will forward to pre-assigned number automatically

Call forward no reply, if the line is not answer the call, the call will forward to pre-assigned number automatically

If call waiting enabled, it will send a special tone if another caller tries to reach you when you are using your telephone

Play Call Waiting Tone Enable call waiting tone, caller will hear special tone.

4.8 Advanced

4.8.1 FXS parameters

FXS characteristic parameters include: Call progress Tone, Timeout for Dialing, Send

Polarity Reversal etc. Configuration interface as follow:

26



Figure 4.8-1 FXS Parameters Configuration Interface

FXS parameters description:

Call Process Tone

Timeout for dialing

Timeout for answer(Outgoing call)

Timeout for answer(Incoming call)

Send Polarity Reversal

Hear the dial tone when pick up the phone. Choose the national standards from the drop-down box. Default is the United States.

With the help of dialing timeout, you can limit the time while users typing the digits from an extension. If the timeout expire while the user is typing in the extension then DAG will consider the extension as complete and it will try to send to SIP server. Default value is 4 seconds

This timer set how long the caller party waiting whenmakes outgoing call on extension.

This timer set how long the phone sets ringing when get incoming call

Detect Hook flash

CID Type

Enable polarity reversal to billing.

A protruding button where putting the receiver boards, called Flash. Always press is hang up, pick up the receiver, the fork lift machine from reed called, by hand clap called "Hook flash". Hook flash is a process that put the flash fast by pressing and let go.In essence is to cut off the dc access about 80 to

200 ms. Then switches don't think it's hang on, but keep the call, taking some other operating. The typical application of hook flash is the telephone switchboard. When need to transfer the call to other extension, then telephone hook flash to transfer the call.

There are DTMF and FSK, General for the default.

Message Type The call display formats SDMF and MDMF, General for the default

27



Send CID before Ringing

Delay of sending CID after

Ringing

After enable this configuration, The DAG send caller to phone set before ringing, otherwise the caller ID will display after ringing.

Definite delay timer of caller ID while it set to send caller ID after ringing.

Its Default value 500ms

SLIC Setting Set the unit impedance

4.8.2 Media Parameter

Media parameter mainly include: RTP start port, DTMF parameter, PreferedVocoder.

Configuration Interface as follow：

Figure 4.8-2 Media Parameter Configuration Interface

Media parameter description:

RTP Start Port Default RTP port 8000

DTMF Method

RFC2833 Payload Type Optimization

RFC2833 Payload Type

SINGAL、INBAND、RFC2833

It is configurable When RFC2833 is selected, payload negotiation parameter with remote side, it includes two options: Local and remote

Payloadvalue, default is 101

DTMF Gain

DTMF Send Interval

Coder Name

Default is 0 DB

DTMF send signal interval, default is 200ms.

DAG supports G729、G711U、G711A、G723. while it make outgoing call, G.729 will used as figure

4.8.2 displayed

28


Payload Type

Packetization Time

Rate

Slience Suppression


Each kind of coding has a unique type load value, refer toRFC3551

Voice package time

Voice data flow rate, system default

Default is disable, if enable, according to the current noise environment dynamically adjust mute inhibit threshold,thus in the user in silent state stop transmission background noise bag and save about VoIP bandwidth.In the low bandwidth environment, can reduce the network congestion, greatly improving VoIP call effect.

4.8.3 SIP Parameter

Figure 4.8-3 SIP Parameter Configuration Interface


29


SIP parameter description：

SUBSCRIBE for MWI

Voicemail User ID

RTP Mode in SDP when Call Holding

IP-to-IP Call

URI Includes user=phone

Only Accept Call from Server

Anonymous Call

Reject Anonymous Call

Send Flash Event

# as ending Dial Key

PRACK

Value of “Refer To“ refers to ”Contact”

Domain Query Type

Domain Re-resolution Interval

Voicemail message indicator, it is to be realized in the way of NOTIFY

Access code to voicemail box

When call come into holding, if select to receive and not send packet, then the local can hear call waiting tone. If select to not receive and not send packet, then doesn’t play call waiting tone.

Enable this function, users may use the * business call IP address on the phone.

SIP carries the information, the system defaults not open.

Default is no, it indicates the DAG accept incoming call from

SIP server only

Enable anonymous call,“anonymous” will include in SIP message

Enable this function, reject all anonymous call. Disable by default

After hook flash, flash event will report flash message to server and server deal with this information.

Dial-up, use # as a end descriptor.

RFC3262 defined an optional extension methods—PRACK

（ provisional ack）, Used to support the reliability of the temporary response.

Its function is to require the receiving partycontact with the third partythrough the use of supplied in the request in the address information. “Refer to” field of SIP message fill in

“contact header”.

There are two modes option: A QUERY and SRV QUERY.

Default is A QUERY.

Default 0: forbidden

T1

T2

T4

Max Timeout

Heartbeat Interval

T1 timer of SIP protocol, default is 500ms



The max timeout of sending or receiving, default is 32s

Default is 10s.

30



Voice mail instructions

:

Here DAG work with Elastixas the example, introduces how voicemail work in DAG.

1）DAG register to Elastix server. Corresponding extension number enable voice mail function in

Elastix and set password. As below：

Figure 4.8-4 Elastix Voicemail Configuration Interface

2）check feature code in Elastix and change it as necessary. Its default feature codes setting as below:

Figure 4.8-5 Elastix Voicemail Setting

Figure4.8-6 Voice Mail Setting In SIP Parameter

3）Enable voice mail in DAG andElastix will ask you to leave a message after ringing 15 seconds, thenElastixwill record and display your message.

31



Figure 4.8-7 Voicemail Setting

4）DAG dial *200#, then dial voicemail account and thenask password for Validation. After that the user will hear voice message.

4.8.4 Fax Parameter

Fax introduction：

DAG fax parameter includes: fax mode, Fax sound detection party, ECM, Rate.

Figure 4.8-8 Fax Parameter Configure Interface

Fax parameter description:

Mode Fax mode support T.38, T.30(Pass-through),Modem, Adaptive.

Tone Detection by

ECM

Rate

Fax sound detection mode: Caller, Callee, Automatic.

Fax error correction information

The rate of sending and receiving.

32


4.8.5 Digit Map


Figure 4.8-9Digit Map

Gateway is collect digits dialed by user, if received a number to be immediately report, the efficiency is too low and a large number of take up network resources. A reasonable method is concentration sending a message after receiving all number. How to judge the gateway receiving all number is the difficulties of this method. The solution is the call agent loading a

“Digit Map” to gateway.

Digit Map includes a series figure characters, when the dial-up sequence and one received a character string matching, it means the number has received neat. Digital string contains characters allowed：data0~9, letterA~D,“#”,“*”, letter T, letter x and “.”. "|" parts of each string is a choice of dial-up solutions; “[]”means choose anyone;“*”means one reports; letter T means detected timer overtime; x means any data; “.”means multiple characters can be behind， include 0; “#”means report immediately.

Digit Map Syntax:

1. Supported objects

Digit: A digit from "0" to "9".

Timer: The symbol "T" matching a timer expiry.

DTMF: A digit, a timer, or one of the symbols "A", "B", "C", "D", "#", or "*".

33



2. Range []

One or more DTMF symbols enclosed between square brackets ("[" and "]"), but only one can be selected.

3. Range ()

One or more expressions enclosed between round brackets ("(" and ")"), but only one can be selected.

4. Separator

|: Separated expressions or DTMF symbols.

5. Subrange

-: Two digits separated by hyphen ("-") which matches any digit between and including the two. The subrange construct can only be used inside a range construct, i.e., between "[" and "]".

6. Wildcard x: matches any digit ("0" to "9").

7. Modifiers

.: Match 0 or more times.

8. Modifiers

+: Match 1 or more times.

9. Modifiers

?: Match 0 or 1 times.

Example:

Assume we have the following digit maps:

1. xxxxxxx | x11 and a current dial string of "41". Given the input "1" the current dial string becomes "411". We have a partial match with "xxxxxxx", but a complete match with "x11", and hence we send "411" to the Call Agent.

2. [2-8] xxxxxx | 13xxxxxxxxx

Means that first is "2","3","4","5","6","7" or "8", followed by 6 digits;

34


DAG1000-4FXS Voice Gateway User Manual or first is 13, followed by 9 digits.

3. (13 | 15 | 18)xxxxxxxxx

Means that first is "13","15" or "18", followed by 8 digits.

4. [1-357-9]xx

Means that first is "1","2","3" or "5" or "7","8","9", followed by 2 digits.

4.8.6 Feature Codec

Feature codec includes device function and call function. Feature codec as follow:

Figure 4.8-10 Feature Code Configuration Interface


35


Inquire LAN port IP address

Inquire WAN port IP address

Inquire Phone Number

Setting IP Mode

Network Work Mode

Configure IP Address

Network subnet mask configure

Network Gateway Configure

Renew DHCP

Access Web by Wan in Rout Mode

Reset Factory

Restart Device

Call onhold/offhold

Dial*158# to obtain device WAN port IP address

Dial*159# to obtain device WAN port IP address

Dial*114# to obtain port account

*150*0#, means pppmodem, *150*1#, means static IP,

*150*2#, means obtain IP address by DHCP, *150*3#, means pppoe.

*157*0#, set network work mode to routing mode; *157*1#, set network work mode to bridge mode

*152*+IP, set gateway IP address

*153*+subnet mask, set gateway subnet mask

*156*+gateway IP, set gateway

*193#, set dynamic IP again

Allow access web through WAN port：*160*1#; don’t allow access web through WAN port：*160*0#

*166*000000#, reset factory

*111#, restart device

When call process, dial*# into call hold.（Recovery the call through hook flash or *#）

Directly dial the end user IP to call

*51#, enable call waiting function

Call by IP

Call Waiting Activate

Call Waiting Deactivate

Blind Transfer

*50#, forbid call waiting function

If the call transfer to 801, first hook flash and then dial the *

87 * 801#

Call Forward Unconditional Activate *72*+ phone number#, transfer the call from the phone number

Call Forward

Deactivate

Unconditional

Call Forward Busy Activate

*73#, forbid call forward unconditional

*90*+ forward busy number#

Call Forward Busy Deactivate *91#, forbid call forward busy

Call Forward No Reply Activate

Call Forward No Reply Deactivate

Do Not Disturb Activate

Do Not Disturb Deactivate

Dial Voicemail

*92*+ forward no reply number#

*93#, close this function

*78#, enable DND function

*79#, close DND function

*200#, visit voice mail box

36



Note：* private services are open by default

4.8.7 System Parameter

System parameters include: STUN、NTP、Provision、WEB parameter、Telnet.

1）STUN：STUN（Simple Traversal of UDP over NATs）is a network protocol. It allows users back of NAT find their own public network address, NAT type and internet end port have been bound by NAT for a local port. Two back of NAT router devices established UDP communication through this information.

STUN doesn’t support TCP connection and H.323.

2）NTP：Network Time Protocol（NTP）is a computer time synchronization protocol.

3）Provision：Auto Provisioning can be used to provide general and specific configuration parameters ("Settings") to the DAGsand to manage firmware actualization.

System parameter configuration interface as follow:


Figure 4.8-11System Configuration Interface

37


STUN Server Address

STUN Server Port

Primary NTP server address

STUN server IP address

STUN server port

Primary NTP server IP address, system default is us.pool.ntp.org

Primary NTP server port Default is 123

Secondary NTP server address Default is 18.145.0.30

Secondary NTP server port

SYN Interval

Time Zone

Primary Provision server IP

Secondary Provision server IP

Check Interval

WEB Port

Access Web by WAN

Telnet Port

Default is 123

Every certain time synchronization gateway time, the system default every 3600 s synchronous once.

Time zone can be chosen. System default the United States central time, Chicago.

Server IP address or domain provided by Provision server.

Server IP address or domain provided by Provision server.

Every once in a while check whether a program or configuration files need to be updated. System default 24 hours

Gateway web port, default is 80

Enable or disable accessing web by WAN

Telnet service port, default is 23.

4.9 Call & Routing

4.9.1 Port Group

Port group parameter include: Index, description etc. Port group configure interface as follow:

38



Index

Description

Figure 4.9-1 port group configuration interface

Port groupNumber, It uniquely identifies a route,range from

0-7

Primary/Secondary Display Name

Primary/Secondary SIP User ID

Primary/Secondary Authenticate ID

Port group description,its purpose is so you can identify the port group with a meaningful name

Port group display, which will be used in SIP message, example:

INVITE sip:[email protected] SIP/2.0

Via:SIP/2.0/UDPpc33.atlanta.com;branch=z9hG4bK776asdhds

Max-Forwards: 70

To: Bob <sip:[email protected]>

From: Alice <sip:[email protected]>;tag=1928301774

Here Bob and Alice is the display

User account information, provided by VoIP service provider

(ITSP). Usually in the form of digit similar to phone number or actually a phone number.

SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User

ID.

Primary/Secondary

Password

Authenticate

Password of SIP user ID

Port Select

 It specifies the policy for selecting port in a port group

 Ascending: the system always selects a port from the minimum number. The preferential selection of the port can be realized through this mode

39


Port


 Cyclic ascending: when system selects ports’ Priority, it always begin from the number next to the number selected last time, if the maximum priority number is selected last time, then the next number is the minimum priority number, and move in cycles like this

 Descending: when system selects ports’ priority, it always begin to select from the maximum priority number

 Cyclic descending: when system selects ports’ Priority, it always begin from the number before to the number selected last time, if the minimum priority number is selected last time, then the next number is the maximum priority number, and move in cycles like this

 Group ring: all ports ringing at the same time

Add some ports to the same group

4.9.2IP Trunk

A peer-to-peer VoIP call occurs when two VoIP phones communicate directly over IP without IP

PBXs between them. A peer-to-peer call can be initiated directly by dialing destination phone number in DAGs and also receiving incoming calls from other peer to peer gateway. IP trunk is help to DAGs establish peer-to-peer call between DAGs and other VoIP phones. IP trunk will be used in routing configuration.


Figure 4.9-2 IP Trunk Configuration Interface

40

Index

Description

Remote Address

Remote Port

Heartbeat


IP trunk number, it is range from 0 to 63

The description of IP trunk,its purpose is so you can identify the IP trunk with a meaningful name

Peer IP addressor domain name

Peer SIP port

Default is disable, if enable, DAG will send “OPTION” to peer device

4.9.3 Routing Configuration

Figure 4.9-3 Routing Parameter Configuration Interface

This option determines the following routing of call take effect before or after manipulation.

4.9.4IP-Tel Routing


Figure 4.9-4 IP-Tel Routing Parameter

41

Index

Description

Calls from

Caller Prefix

Callee Prefix


Routing priority: 0-31, 0 is the highest priority. its purpose is so you can identify theIP0->Tel routing with a meaningful name

IP Trunk/SIP Server, any means any IP

Caller number Prefix, its length normally less or equal to caller number, which helps to matching routing exactly. if caller number is 2001, the caller prefix can be 200 or 2. “any” means match any caller number like

“bob1”,”29801”

Called number Prefix, its length normally less or equal to callednumber, which helps to matching routing exactly. if called number is

008675526456659, the called prefix can be 0086755 or 00.,“any” means match any called number

This call routing is routing to port or port group Calls to

4.9.5 Tel-IP/Tel Routing

Index

Description

Calls From

Figure 4.9-5 Tel-IP/Tel Parameters Configuration

Routing priority :0-31, 0 is the highest priority. its purpose is so you can identify the routing with a meaningful name

Tel-IP call select port or port group

42


Caller Prefix

Callee Prefix


Caller number Prefix, its length normally less or equal to caller number, which helps to matching routing exactly. if caller number is 2001, the caller prefix can be 200 or 2. “any” means match any caller number like “bob1”,”29801”

Called number Prefix, its length normally less or equal to called number, which helps to matching routing exactly. if called number is

008675526456659, the called prefix can be 0086755 or 00., “any” means match any called number

This call routing is routing to port, port group, IP trunk and SIP server. Calls to

4.10 Manipulation Configuration

4.10.1 IP-Tel Callee

Description

Calls From

Caller Prefix

Figure 4.10-1 IP-Tel Callee number configuration

IP-Tel manipulation name

This call come from IP trunk or SIP server.

Caller number Prefix, its length normally less or equal to caller number, which helps to matching routing exactly. if caller number is 2001, the caller prefix can be 200 or 2. “any” means match any caller number like “bob1”,”29801”

43


Callee Prefix

Calls to

Stripped Digits from Left


Called number Prefix, its length normally less or equal to called number, which helps to matching routing exactly. if called number is 008675526456659, the called prefix can be 0086755 or 00.,

“any” means match any called number

This call routing is routing to port, port group

Remove the called number digits from the left

Stripped Digits from Right

Prefix to Add

Suffix to Add

Number of Digits to Leave from

Right

Remove the called number digits from the right

Add a number prefix

Add a number suffix

Starting from the right to retain the called number digits

4.10.2 Tel-IP Caller


Figure 4．10-2 Tel-IP Caller

44


Configuration parameters are the same with “IP->Tel Callee”.

4.10.3 Tel-IP Callee

Figure 4.10-3 Tel-IPCallee

Configuration parameters are the same with “Tel->IP Caller”.

4.11 Maintenance

4.11.1 syslog Parameter

Syslogis a protocol used in (TCP/IP) network transmission of record of the standard file information.

Syslog agreement belongs to a kind of master slave agreement: Syslog sender will sent a small

45


DAG1000-4FXS Voice Gateway User Manual text information (less than 1024 bytes) to syslog the receiver. The receiver are: “syslogd”、

“syslog daemon” or syslog server. Syslog message can be transferred by

TCP/UDP.

Syslog level：

• none Used to misarrange

• debug Not including function conditions or the question of other information

• notice importance common conditions

• warning Early warning information

• error Stop error conditions of tools or some part of the realization of the function subsystem

Figure 4.11-1 Syslog Parameter Configuration

Enable send CDR, and then send communication information to syslog server.

4.11.2 Firmware Upload

The process of firmware upload：

1）Click “Firmware Upload”

2) Browse files and choose the loading program (Name the file extension. ldf)

3) Click “Upload”, the upload process will last about 60s and device can automatically restart after uploading.（The firmware update process don't shut off the power）

Figure 4.11-2 Firmware upload Configuration


46


4.11.3 Data Backup

The process data backup:

1) Click “Data Backup”

2) Click “Backup” to backup data to PC.

Figure 4.11-3 Data Backup Interface

4.11.4 Data Restore

The processes of data restore:

1) Click “Data Restore”

2) Browse file, select data file.

3) Click “Restore” and then import successfully, the device will restart automatically.

Figure 4.11-4 Data Restore Interface

4.11.5 Ping Test

Send test data packets to IP, check each other whether have response and statistical response time. It is ping. Used to test internet and analyzed network fault.

Application format

：

Ping IP address. It is used to check the network connectivity or network connection speed command.

Pinginstructions:

1) Click “ping test”

47



2) Fill IP address or domain connected, click start.

3) Received a message indicates that network connection normal, or network connected to a fault.

Figure 4.11-5 Ping Parameter Interface

4.11.6 Tracert Test

Tracert is trace router and used to tracking routing.

Tracert sends a sequence of Internet Control Message Protocol (ICMP) echo request packets addressed to a destination host. Determining the intermediate routers traversed involves adjusting the time-to-live (TTL), aka hop limit, Internet Protocol parameter. Frequently starting with a value like 128 (Windows) or 64 (Linux), routers decrement this and discard a packet when the TTL value has reached zero, returning the ICMP error message ICMP Time Exceeded.

Tracert works by increasing the TTL value of each successive set of packets sent. The first set of packets sent have a hop limit value of 1, expecting that they are not forwarded by the first router. The next set have a hop limit value of 2, so that the second router will send the error reply. This continues until the destination host receives the packets and returns an ICMP Echo

Reply message.

Trace route uses the returned ICMP messages to produce a list of hops (which usually consists

48


DAG1000-4FXS Voice Gateway User Manual of routers and layer 3 switches) that the packets have traversed. The timestamp values returned for each router along the path are the delay (aka latency) values, typically measured in milliseconds for each packet.

Tracert introduce：

1) Click tracert test.

2) Fill IP address or domain connected, click start.

Figure 4.11-6 Tracert Test Interface

4.11.7 Password Modification

Includes WEB username and password, Telenet username and password modify.

Note：Default web and telnet username and password is: admin, admin.

49



Figure 4.11-7 Password Modification Interface

4.11.8 Factory Reset

Click “Apply” to restore the factory settings.

Figure 4．11-8 Factory Reset Interface

4.11.9 Device Restart

Click the “Save” button in the Configuration page to save the changes to the equipmentconfiguration. The following screen confirms that the changes are saved. If the changes need restart, reboot or power cycle the equipment to make the changes take effect.


Figure 4.11-9 Device Restart

50


5. Glossary

•

DNS：Domain Name System

•

SIP：Session Initiation Protocol

•

TCP：Transmission Control Protocol

•

UDP：User Datagram Protocol

•

RTP：Real Time Protocol

•

PPPOE：point-to-point protocol over Ethernet

•

VLAN：Virtual Local Area Network

•

ARP：AddressResolution Protocol

•

CID：Caller Identity

•

DND：Do NOT Disturb

•

DTMF：Dual Tone Multi Frequency

•

NTP：Network Time Protocol

•

DMZ：Demilitarized Zone

•

STUN：Simple Traversal of UDP over NAT

•

PSTN：Public Switched Telephone Network

51


Dinstar DAG1000-8FXS User manual

DAG1000-4FXS Voice Gateway

User Manual V2.0

Revision Records

Table of Contents

1. Equipment Introduction

1.1 Overview

1.2 Equipment appearance

1.3 Power supply

1.4Network Applications

1.5 Functions and Features

2. Basic Operations

2.1 Phone Call

No dial timeout

2.2 Call Hold

2.3 Call Waiting

2.4 Call Transfer

Call features enable

Attend Transfer

2.5 Call Features

2.6 Sending and Receiving Fax

3. Local IVR Operation

3.1 Inquire IP address

3.2 Factory Reset

3.3 Configure LAN Port’s IP Address

4. WEB Configuration

4.1 WEB Login

4.2 Navigation Tree

4.3 State and Statistics

4.4 Quick Setup Wizard

4.5 Network Configuration

4.6 SIP Server

4.7 Port Configuration

4.8 Advanced

4.9 Call & Routing

4.10 Manipulation Configuration

4.11 Maintenance

5. Glossary

Related manuals

Table of contents

Dinstar DAG1000-8FXS User manual

DAG1000-4FXS Voice Gateway

User Manual V2.0

Revision Records

Table of Contents

1. Equipment Introduction

1.1 Overview

1.2 Equipment appearance

1.3 Power supply

1.4Network Applications

1.5 Functions and Features

2. Basic Operations

2.1 Phone Call

No dial timeout

2.2 Call Hold

2.3 Call Waiting

2.4 Call Transfer

Call features enable

Attend Transfer

2.5 Call Features

2.6 Sending and Receiving Fax

3. Local IVR Operation

3.1 Inquire IP address

3.2 Factory Reset

3.3 Configure LAN Port’s IP Address

4. WEB Configuration

4.1 WEB Login

4.2 Navigation Tree

4.3 State and Statistics

4.4 Quick Setup Wizard

4.5 Network Configuration

4.6 SIP Server

4.7 Port Configuration

4.8 Advanced

4.9 Call & Routing

4.10 Manipulation Configuration

4.11 Maintenance

5. Glossary

Related manuals

Dinstar

DAG2000-16S

Dinstar

DAG1000-8S

Dinstar

DAG1000-1S/2S

Dinstar

DAG2000 Series

Dinstar

DAG1000-4S

Dinstar

FXSFXO

Dinstar

DAG2000 Series

Table of contents