DAG Serial User Manual

Dinstar Technologies Co., Ltd.

Address: Floor 6, Guoxing Building, Changxing Road, Nanshan District, Shenzhen, China 518057
Telephone: +86 755 6130 2265
Fax:
Emails:

+86 755 2645 6659

sales@dinstar.com, support@dinstar.com

Website: www.dinstar.com

TABLE OF CONTENTS
WELCOME ...................................................................................................................................1
CONFIGURE YOUR DAG1000 ...................................................................................................4
EQUIPMENT PACKAGING ........................................................................................................4
CONNECT YOUR DAG ...............................................................................................................4
CONFIGURE YOUR DAG serial ................................................................................................5
CONFIGURE YOUR DAG serial ................................................................................................6

DAG serial APPLICATION DESCRIPTION .................................................................................7

A. IP PBX / SIP Server with DAG serial ....................................................................................7
B. FXS Gateway with DAG1000-XO [No SIP Server required] ...............................................8

PRODUCT OVERVIEW ..............................................................................................................9

BASIC OPERATIONS ...............................................................................................................11
PLACING A PHONE CALL ...................................................................................................... 11
PHONE OR EXTENSION NUMBERS .............................................................................. 11
DIRECT IP CALLS............................................................................................................ 11
CALL HOLD ............................................................................................................................. 12
CALL WAITING ........................................................................................................................ 12
CALL TRANSFER .................................................................................................................... 12

Call Features ............................................................................................................................. 13

SENDING AND RECEIVING FAX ............................................................................................ 13

CONFIGURATION GUIDE.........................................................................................................14
CONFIGURE LAN PORTS IP ADDRESS .............................................................................. 14
CONNECT THE DAG serial USING THE WEB BROWSER .................................................. 14
ACCESS THE WEB CONFIGURATION MENU .............................................................. 14
IMPORTANT SETTINGS ......................................................................................................... 15
CONFIGURE SIP .............................................................................................................. 15
DTMF METHODS ............................................................................................................. 15
PREFERRED VOCODER (CODEC) ................................................................................ 15
CONFIGURE THE DAG serial USING THE WEB BROWSER ............................................. 15
SYSREM INFO .................................................................................................................. 16
NETWORK PARAMETER ................................................................................................ 17
SYSTEM PARAMETER .................................................................................................... 21
TOOLS .............................................................................................................................. 29
SAVING THE CONFIGURATION CHANGES .................................................................. 31

ii

WELCOME
Thank you for purchasing DAG Serial. The affordable, features rich Analog Telephone Adaptor. It
features the rich audio quality, a broad range of voice codecs.
This manual will help you learn how to operate and manage your DAG Serial Analog Telephone
Adaptor and make the best use of its many upgraded features including simple and quick installation, and
direct IP-IP Calling. This DAG Serial is very easy to manage and configure, and is specifically designed
to be an easy to use and affordable VoIP solution for both the residential user and the residential user and
the tele-worker.

GLOSSARY OF TERMS
ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit
from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream,
depending on line distance.
AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to
control the gain of a system in order to maintain some measure of performance over a changing
range of real world conditions.
ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically
IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The
protocol operates below the network layer as a part of the interface between the OSI network and
OSI link layer. It is used when IPv4 is used over Ethernet
ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like
allywll DAG1000 series products.
CODEC Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM A data packet carrying its own address information so it can be independently routed from
its source to the destination computer
DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while sub-sampling.
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain
names into IP addresses
DSP Digital Signal Processor. A specialized CPU used for digital signal processing. Allywll products all
have DSP chips built inside.
DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing using
in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals
support only 12 of them (0-9, * and #).
FXO Foreign eXchange Office. An FXS device can be an analog phone, answering machine, fax, or

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anything that handles a call from the telephone company like AT&T. They should also operate the
same way when connected to an FXS interface.
An FXS interface will accept calls from FXS or PSTN interfaces. All countries and regions have their
own standards.
FXS is complimentary to FXS (and the PSTN).
FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXS
extension (usually an analog phone).
An FXS device will allow any FXS device to operate as if it were connected to the phone company.
This makes your PBX the POTS+PSTN for the phone.
The FXS Interface connects to FXS devices (by an FXS interface, of course).
DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses,
to deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to
provide other configuration information such as the addresses for printer, time and news servers.
ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing
echo from a voice communication in order to improve voice quality on a telephone call. In addition to
improving quality, this process improves bandwidth savings achieved through silence suppression by
preventing echo from traveling across a network. There are two types of echo of relevance in
telephony: acoustic echo and hybrid echo. Speech compression techniques and digital processing
delay often contribute to echo generation in telephone networks.
H.323 A suite of standards for multimedia conferences on traditional packet-switched networks.
HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve
functions of a server
IP Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX IP-based Private Branch Exchange
IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for
the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax,
and other forms of information that have traditionally been carried over the dedicated circuit-switched
connections of the public switched telephone network (PSTN). The basic steps involved in originating
an

IP Telephony call are conversion of the analog voice signal to digital format and

compression/translation of the signal into Internet protocol (IP) packets for transmission over the
Internet or other packet-switched networks; the process is reversed at the receiving end. The terms
IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100
per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users
who have free or fixed-price Internet access, IP Telephony software essentially provides free
telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the
quality of service expected by subscribers. Session border controllers resolve this issue by providing
quality assurance comparable to legacy telephone systems.
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and
perhaps other media.

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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets
eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The
maximum for Ethernet is 1500 byte.
NAT Network Address Translation
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks, the port used is
UDP 123, allywll products using NTP to get time from Internet.
OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called Caller. The
OBP/SBC acts as if it was the called VoIP phone and places a second call to the called Caller. The
effect of this behavior is that not only the signaling traffic, but also the media traffic (voice, video etc)
crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP
phones. Private OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected
enterprise network. Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols
from private networks with internet connections using NAT.
PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or
called POT (Plain Old Telephone), or circuit switched network.
RTCP Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does
not transport any data itself. It is used periodically to transmit control packets to participants in a
streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of
service being provided by RTP.
RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video
over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first
published in 1996 as RFC 1889
SDP Session Description Protocol is a format for describing streaming media initialization parameters. It
has been published by the IETF as RFC 2327.
SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice
transmission and uses fewer resources and is considerably less complex than H.323. All Dinstar
products are SIP based
STUN Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple
NATs) to find out its public address, the type of NAT it is behind and the internet side port associated
by the NAT with a particular local port. This information is used to set up UDP communication
between two hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will
usually work well with non-symmetric NAT routers.
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP,
applications on networked hosts can create connections to one another, over which they can
exchange data or packets. The protocol guarantees reliable and in-order delivery of sender to
receiver data.

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TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very
basic form of FTP; It uses UDP (port 69) as its transport protocol.
UDP User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP,
programs on networked computers can send short messages known as datagrams to one another.
UDP does not provide the reliability and ordering guarantees that TCP does; datagrams may arrive
out of order or go missing without notice. However, as a result, UDP is faster and more efficient for
many lightweight

or time-sensitive purposes.

VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing
wherein, the presence or absence of human speech is detected from the audio samples.
VLAN A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-exist
on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323,
etc.

CONFIGURE YOUR DAG1000

EQUIPMENT PACKAGING
Unpack and check all accessories. Equipment included in the package:
one 12V universal power adapter
one Ethernet cable
one device unit
one DAG stand

CONNECT YOUR DAG

First, familiarize yourself with the features of the DAG .
FIGURE 1: CONNECTING

DAG1000-4FXS/4 lifeline

DAG1000-8FXS

DAG2000-16FXS

TABLE 1: DEFINITIONS OF THE DAG CONNECTORS

Power Cable

Power adapter connection

WAN Port (RJ-45)

Connect to the internal LAN network or router.

LAN Port (RJ-45)

Connect the LAN port with an Ethernet cable to your PC.

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PHONE (RJ-11)
FXO

FXS port to be connected to analog phones / fax machines.

FXO ports to be connected to physical PSTN lines from a traditional
PSTN PBX or PSTN Central Office.

TABLE 2: DEFINITIONS OF THE DAG LEDS

Power LED

Indicates Power. Remains ON when Power is connected.

WAN LED

Indicates LAN (or WAN) port activity.

LAN LED

Indicates PC (or LAN) port activity.

FXS LED

Indicate status of the respective FXS Ports on the back

panel
Busy
ON (Solid Green)
Available OFF

FXO LED

Indicate status of the respective FXO Ports-PHONE0 to PHONE7 on the back

panel
Busy
ON (Solid yellow)
Available OFF

Run LED

Slow blinking When stand not registered.

Quick blinking When stand registered.

CONFIGURE YOUR DAG serial

The DAG serial is easy to configure using the embedded GUI pages and the following five (5) steps.
FIVE EASY STEPS TO CONFIGURE THE DAG serial
1. Connect a standard touch-tone analog telephone (or fax machine) to first FXS port.
2. Connect another standard touch-tone analog telephone (or fax machine) to second FXS port.
3. Insert the Ethernet cable into the WAN port of DAG serial and connect the other end of the Ethernet
cable to an uplink port (a router or a modem, etc.)
4. Connect a PC to the LAN port of DAG serial .
5. Insert the power adapter into the DAG serial and connect it to a wall outlet.

Ex:DAG serial

Page 5

FIGURE 2: CONFIGURING THE DAG serial

CONFIGURE YOUR DAG serial

The DAG serial is easy to configure using the embedded GUI pages and the following five (5) steps.
FIVE EASY STEPS TO CONFIGURE THE DAG serial
1. Connect a standard touch-tone analog telephone (or fax machine) to FXS port.
2. Connect PSTN line to LINE port.
3. Insert the Ethernet cable into the WAN port of DAG serial and connect the other end of the Ethernet
cable to an uplink port (a router or a modem, etc.)
4. Connect a PC to the LAN port of DAG serial.
5. Insert the power adapter into the DAG serial and connect it to a wall outlet.

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FIGURE 3: CONFIGURING THE DAG serial

DAG serial APPLICATION DESCRIPTION

A. IP PBX / SIP Server with DAG serial
A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the DAG Serial.
In this environment, the SIP server handles SIP registration and call control and the DAG serial
processes media conversion between IP and PSTN calls.
There are 2 ways to configure DAG serial when using with a SIP Server:1. With SIP accounts configured on Channels page. In this case, the GXW acts like an
endpoint requesting registration from the SIP Server. Under the Channels webpage you will
need to fill in the information like SIP User ID, Password, etc. Now, when you try to make
calls from IP, the call will be routed to the SIP Server which will forward it to one of the SIP
accounts on the DAG serial, which will then forward it to the PSTN line.
2. Without SIP accounts. In this case, you simply have to configure the SIP Server to perform
forwarding of the SIP INVITE message with the FXO destination number to the gateways IP
Address. The DAG serial will receive the digits and immediately forward them on the FXO
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lines to the destination PSTN. Most of the configuration on the Gateway for this case will
remain default, except Stage Method needs to be set to 1, and SIP Server IP Address/DNS
name has to be filled.

FIGURE 4: FUNCTIONAL DIAGRAM OF SIP-Server & DAG serial

For incoming calls from the PSTN analog endpoints to the DAG1000-8O, the device will auto
forward each call to a configured IP extension or DAG serials FXS ports. The SIP Server can then
route the call based on its own configuration or IVR system.

B. FXS Gateway with DAG1000-XO [No SIP Server required]

Alternatively, the DAG1000-XO (Ex. DAG1000-8S) can be used without a SIP Server. You can use
it in conjunction with a FXS Gateway (Ex. DAG1000-8S) and still be able to originate and terminate
calls from IP to PSTN and vice versa. All you need to make sure is that the 2 gateways are able to
locate each other (they should be on the same LAN or on Public IPs).

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FIGURE 5: DAG1000-8S& DAG1000-8O SCENARIO/TOLL- FREE CALLING BETWEEN LOCATIONS

In this diagram, configure the SIP Server field to be the IP Address of the other gateway (i.e.
configure IP address of FXS gateway to be SIP Server of DAG1000-8O and vice versa). Please be
sure you set SIP Registration to No.
Expected call flow: Analog Phone (DAG1000-8S) picks up and dials destination PSTN number.
The call gets routed to the DAG1000-8O which dials out the digit string onto the FXO Lines, thus
reaching the destination PSTN endpoint. On the reverse, incoming calls from PSTN endpoints will
be routed automatically to the FXS Gateway through the DAG1000-8O.

PRODUCT OVERVIEW
The DAG serial is a full feature voice and fax-over IP device that offers a high-level of integration
including dual 10M/100Mbps network ports with integrated router, NAT, DHCP server, dual port FXS
telephone gateway, market-leading sound quality, rich functionalities, and a compact and lightweight
design. The DAG serial fully compatible with SIP industry standard and can interoperate with many other
SIP compliant devices and software on the market. Moreover, it supports comprehensive voice codecs
including G.711 (a/-law), G.723.1 and G.729AB.

KEY FEATURES
Ethernet
Ports

DHCP

PSTN Pass
through

Voice Codec

Remote
Configuration

4 RJ-45 (LAN)

Server/Client

No

T.38,
PCMU,PCMA,
G.723, G.729AB

TFTP/HTTP

TABLE3: DAG serial TECHNICAL SPECIFICATIONS

DAG100 has one FXS portDAG2002QA has two FXS ports, DAG2004QA has
Telephone Interfaces four FXS ports, DAG1000-8S has eight FXS ports, DAG2004QB has four FXO
ports, DAG1000-8O has eight FXO ports

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Network Interface
LED Indicators

Voice over Packet

Capabilities

DHCP Server/Client
Telnet Server
Fax over IP
QoS
IP Transport
DTMF Method

10M/100 Mbps, RJ-45.

Power, WAN, LAN, Run,PHONE ( FXS and FXO )
Voice Activity Detection (VAD) with CNG (comfort noise generation) and PLC
(packet loss concealment), Dynamic Jitter Buffer, Modem detection &
auto-switch
to G.711, Packetized Voice Protocol Unit (supports RTP/RTCP and AAL2
protocol), G.168 compliant Echo Cancellation, LEC (line echo cancellation) with
NLP
Yes, NAT Router or Switched Mode
Yes
T.38 Fax up to 14.4kpbs and Fax Pass-through, Fax Datapump V.17, V.19,
V.27ter, V.29 for T.38
Diffserve, TOS, 802.1 P/Q VLAN tagging
RTP/RTCP
Flexible DTMF transmission method, user interface of In-audio, RFC2833, or
SIP Info

IP Signaling

SIP (RFC 3261)

Provisioning

TFTP, HTTP

Management

Syslog support, HTTP and telnet, remote management using Web

browser, Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (Tos,
DiffSery, MPLS), Auto/manual provisioning system

Power
Environmental
Short and long haul
Call Handling
Features
Caller ID
Polarity Reversal /
Wink
EMC
Safety

Output: 12VDC / Input: 100240 VAC/50 Hz

Operational: 32o104oF or 0o40oC
Storage: -20~70C / Humidity: 1090% Non-condensing
REN3: Up to150 ft on 24 AWG line
Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer,
hold, forward, mute, Do-Not-Disturb(DND)
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
Yes
EN55022/EN55024 and FCC part15 Class B
UL

TABLE 4: DAG serial HARDWARE SPECIFICATIONS

Page 10

LAN Interface

Universal Switching
Power Adaptor
Operation Temperature
Storage Temperature
Humidity
Compliance
FXS polarity reversal
signal to be sent
Protection of telephone
interface against
lightning strike

DAG100/DAG2004QA/DAG2004QB have Two (2) 10M/100 Mbps, RJ-45,

and DAG2002QA/DAG2004QA/DAG2004QB have Tour (4) 10M/100 Mbps,
RJ-45. (integrated router)
Input: 100-240V AC, 50Hz, 0.5A Max
Output: 12V DC, 1.0A
UL certified
32o104oF / 0~40C
-20~70C
10% - 90% (non-condensing)
FCC, CE
Yes
10/700,4000V

BASIC OPERATIONS
PLACING A PHONE CALL
PHONE OR EXTENSION NUMBERS
1. Dial the number directly and wait for 3 seconds (Default No dial timeout);
2. Dial the number directly and press #.
Examples:
1. Dial an extension directly on the same proxy, (e.g. 8080), and then press the # or wait for 3 seconds.
2. Dial an outside number (e.g. (626) 666-8080), first enter the prefix number (usually 1+ or international
code) followed by the phone number. Press # or wait for 3 seconds. Check with your VoIP service
provider for further details on prefix numbers.
DIRECT IP CALLS
Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to
talk to each other in an ad hoc fashion without a SIP proxy.
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).
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 2.
Examples:

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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. Destination port is 5060.
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.
Disable direct IP-IP calling must be set to No in web configuration page.
Call features enable must be set to Yes in web configuration page.

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.

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 call is placed on hold. Press the
flash button to toggle between two active calls.

CALL TRANSFER
Blind Transfer
Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:
1. Caller A presses FLASH on the analog phone to hear the dial tone.
2. Caller A dials *87 then dials caller Cs 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 (transferee has received a 200 OK from transfer target). 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.
Attended Transfer
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. Caller A then dials Caller Cs 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.
4. If Caller C does not answer the call, Caller A can press flash to resume call with Caller B.

Page 12

Call Features
The DAG serial supports all the traditional and advanced telephony features.
TABLE 5: DAG serial CALL FEATURE DEFINITIONS

*47

Direct IP Calling. Dial *47 + IP address. No dial tone is played in the middle. Detail see
Direct IP Calling section on page 9.

*50

Disable Call Waiting (for all subsequent calls)

*51

Enable Call Waiting (for all subsequent calls)

*72

Unconditional Call Forward: Dial *72 and then the forwarding number followed by #.
Wait for dial tone and hang up. (dial tone indicates successful forward)

*73

Cancel Unconditional Call Forward. To cancel Unconditional Call Forward, dial *73,
wait for dial tone, and then hang up.

*78

Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.

*79

Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.

*87

Blind Transfer

*90

Busy Call Forward: Dial *90 and then the forwarding number followed by #. Wait
for dial tone then hang up.

*91

Cancel Busy Call Forward. To cancel Busy Call Forward, dial *91, wait for dial tone,
and then hang up.

*92

No Answer Call Forward. Dial *92 and then the forwarding number followed by #. Wait
for dial tone then hang up.

*93

Cancel no Answer Call Forward. To cancel no Answer Call Forward, dial *93, wait for
dial tone, and then hang up.

*99

Cancel All Call Forward

*114#

Report Phone No.

*158#

Report IP Address

*111#

Reset

Flash/Hook

Toggles between active call and incoming call (call waiting tone). If not in conversation,
flash/hook will switch to a new channel for a new call.

SENDING AND RECEIVING FAX

DAG serial supports fax in two modes: 1) T.38 (Fax over IP) and 2) fax 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.

Page 13

CONFIGURATION GUIDE
CONFIGURE LAN PORTS IP ADDRESS
Telephone set must connect to the FXS port:
1.Dynamic IP address by DHCP
Offhook;
Input *150*2#;
Onhook;
If hint success, after 10 seconds, restart the DAG1000 . (Power-off then power-on)
2.Static IP address
Offhook; input *150*1#; onhook; then
1 Config IP address
Offhook; input *152*172*16*0*100# ; onhook.
2 Config netmask
Offhook; input *153*255*255*0*0# ; onhook.
3 Config gateway IP address (next hop)
Offhook; input *156*172*16*0*1# ; onhook.
4 if success, After 10 seconds, restart the DAG1000 ; (Power-off then power-on)
3. Query the IP address of DAG1000 : Offhook, input *158#
4. If the DAG serial uses PPPoE method to get IP addressit need to configure by web browser.
NOTE: the telephone will play voice hint Setting successfully if the setting step is correct.

CONNECT THE DAG serial USING THE WEB BROWSER

DAG serial has an embedded Web server that will respond to HTTP GET/POST requests. It also has
embedded HTML pages that allow users to configure the DAG serial through a web browser such as
Microsofts IE and AOLs Netscape.
ACCESS THE WEB CONFIGURATION MENU
The DAG serial GUI configuration menu can be accessed via LAN or WAN port:
From the LAN port:
1. Offhook, input *158# and get an IP address. The default gateway IP address: 192.168.1.111.
2. Directly connect a computer to the LAN port.
3. Open a command window on the computer.
4. Type in ipconfig /release, the IP address etc. becomes 0.
5. Type in ipconfig /renew, the computer gets an IP address in 192.168.1.x segment by default
6. Open a web browser, input the default gateway IP address. http://192.168.1.111. You will see the login
page of the device.
7. The default username is admin while the default password is admin too.
From the WAN port:
The WAN port HTML configuration option is disabled by default from factory. To access the HTML
configuration menu from the WAN port:
1. Access the Web from LAN port.

Page 14

2. Set Work Mode to Route Mode, restart DAG serial.

3. Enable the Remote Manage.
4. Find the WAN IP address of the DAG serial .example:172.16.0.177
5. Access the DAG serial Web Configuration page by the following URI via WAN port: http://
172.16.0.177

IMPORTANT SETTINGS
You must configure the following settings according to the local environment.
NOTE: Most settings on the web configuration pages are set to the default values.
CONFIGURE SIP
SIP server settings are in System Parameter->SIP Config pages, and SIP user & password setting are in
System Parameter->Port Config, you must configure:
SIP server address,
SIP server port,
SIP user ID,
Authenticate ID,
Authenticate Password,
If you need outbound proxy, input:
Outbound proxy address,
Outbound proxy port
DTMF METHODS
DTMF Settings are in System Parameter->Service Config pages.
DTMF INBAND
DTMF RFC2833
DTMF SIGNAL
Enable DTMF method based on your PBX system.
PREFERRED VOCODER (CODEC)
The DAG serial supports a broad range of voice codecs. Under System Parameter->Service Config
web pages, choose your preferred order of different codecs:
PCMU
PCMA
G729 AB

G723

CONFIGURE THE DAG serial USING THE WEB BROWSER

The web page configuration includes 1) system info, 2) Network Parameter, 3) System
Parameter, 4) Tools.

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FIGURE 6: Web trees

SYSREM INFO

The system information shows the current status of the system, including IP address, running
time, network traffic, version and FXS ports status.
FIGURE 7: System Info

TABLE 8: System Info Definitions

MAC Address
Network Mode

The device ID in HEX format. This is needed for ISP troubleshooting. Note
there are separate MAC addresses for the WAN side and the LAN side.
Route mode or bridge mode, if it is bridge, WAN Port display Network,

and the wan port as same as the LAN port

WAN port

Shows WAN IP address of DAG serial ,

DHCP mode: all the field values for the Static IP mode are not used (even
though they are still saved in the Flash memory.) The DAG serial acquires
its IP address from the first DHCP server it discovers from the LAN it is
connected.
Using the PPPoE feature: set the PPPoE account settings. The DAG serial
will establish a PPPoE session if any of the PPPoE fields is set.
Static IP mode: configure the IP address, Subnet Mask, Default Router IP
address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These
fields are set to zero by default.

LAN port

Shows LAN IP address of DAG1000 .if network Mode is bridge, LAN port
will no display.

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DNS Server

Short for Domain Name System (or Service or Server), an Internet service
that translates domain names into IP addresses

System Up Duration

Shows system up time since the last reboot.

Network Connection
Occupancy

Show network occupancy

Version Information

Program: This is the main software release. Boot and Loader are seldom
changed.

Port Information

Port No.
DAG serial has X ports,No.0 and No.X-1,X is DAG200Xs X.
Type
FXS or FXO
Registered Yes or No
Number
User Extension number

NETWORK PARAMETER
Network parameter includes 1)Local Network 2) QoS Parameter 3)MAC Clone 4)DHCP Server 5) DMZ
Host 6) Forward Rule 7) Static Route, and last four options will display if local network is route mode.
FIGURE 9:Network Parameter

Local Network
FIGURE 10: Local Network

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Table 7 Local Network

Work Mode

This parameter controls whether the device is working in NAT router mode
or Bridge mode.

WAN Port Config

This option specifies the WAN ports ip address, and its Ethernet network
work mode.
Link speed & duplex
This option specifies the Ethernet networks work mode. It have:
Auto Detect,10Mbps/Half Duplex,10Mpbs/Full Duplex,100Mbps/Half
Duplex,100Mpbs/Full Duplex,
There are three modes to operate the DAG serial WAN IP address, default
option is Dynamically assigned via DHCP:
Dynamically assigned via DHCP:
All the field values for the Static IP mode are not used (even though they are
still saved in the Flash memory.) The DAG serial acquires its IP address
from the first DHCP server it discovers from the LAN it is connected.
Static IP mode:
Configure the IP address, Subnet Mask, Default Router IP address, DNS
Server 1 (primary), DNS Server 2 (secondary) fields. These fields are set to
zero by default.
Dynamically assigned via PPPoE :
Set the PPPoE account settings. The DAG serial will establish a PPPoE
session if any of the PPPoE fields is set.

PPPoE account

PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to

Point Protocol over Ethernet) connection.

PPPoE password

PPPoE account password.

LAN Port Config

This option specifies the LAN ports ip address, and its Ethernet network
work mode.
Link speed & duplex

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This option specifies the Ethernet networks work mode. It have:

Auto Detect,10Mbps/Half Duplex,10Mpbs/Full Duplex,100Mbps/Half
Duplex,100Mpbs/Full Duplex,
IP address

Set LAN IP address, default value is 192.168.1.111

Subnet mask
Sets the LAN subnet mask. Default value is 255.255.255.0
This option specifies get the DNS server ways: Dynamically assigned via
DNS Server

DNS(WAN IP address is DHCP IP) and set static DNS (WAN IP

address is static IP)

Qos Config
FIGURE 11: Qos Config

DSCP code point is used for diffserv setting. It utilizes the first 6 bits of IP ToS. You can use different
DSCP for voice or data according to the network provider.

MAC Clone
FIGURE 12: MAC Clone

This page providences the setting of WAN MAC Address. In route mode, DAG serial requires to bind PC
MAC address, as a Certification basis for network provider.

DHCP Server
FIGURE 12: DHCP Server

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This option sets the DHCP server, if you want to establish a sub network, you can use the DAG serial
DHCP Server function, and no need ex-equipment.
Table 8 DHCP Server
IP address pool start
IP address pool end

DHCP address pool initiative address

DHCP address pool end address

IP lease time

Default value is 72hrs (3 Days.) The time IP

address is assigned to the LAN clients

Subnet Mask

optional

Default gateway

optional

Primary DNS server

optional

Secondary DNS server

optional

DMZ Host
Forward all WAN IP traffic to a specific IP address if no matching port is used by DAG serial or defined in
port forwarding.
FIGURE 13:DMZ Host

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Foward Rule
Forwards a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port.
FIGURE 14:Forward Rule

Static Route
FIGURE 15:Static Rule

SYSTEM PARAMETER
System parameter includes 1) System Config 2) Service Config 3) SIP Config 4) port Config 5) Fax
Config.
FIGURE 16:System parameter

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System Config
The page of system setting is mainly used to set some system parameters, including:
1) Hint Language 2) Provision NTP 4) Time Zone.
FIGURE 17:System Config

Table 9 System Config

Hint Language

Languages supported with voice prompt

Provision

A service provider or an enterprise with large deployment of DAG serial can

easily manage the configuration and software upgrading of individual
devices remotely from a central server.

NTP

URI or IP address of the NTP (Network Time Protocol) server. Used by the
phone to synchronize the date and time.

Time Zone

Controls how the date/time displayed according to the specified time zone

Service Config (DAG1000-XS & DAG100)

In order to adapt to different environments, it is necessary to set some parameters in the service config
page.

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FIGURE 18:Service Config (DAG1000-XS & DAG100)

Table 10 Service Config (DAG1000-XS & DAG100)

RTP Start Port

Defines the local RTP-RTCP port pair the DAG1000 will listen and transmit.

Silence Suppression

Endpoints sending audio as an RTP stream are not required to send packets
during silent periods. The capability to stop sending RTP packets during

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silent periods is known as "Silence Suppression" or VAD (Voice Activity

Detection).
Call Progress Tone
SLIC Setting

Configure ring or tone frequencies according to preference. By default tones

are set to USA frequencies. Frequencies should be configured with known
values to avoid uncomfortable high pitch sounds.
Dependent on standard phone type (and location).

Hook Flash Detect

Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To
prevent unwanted activation of the Flash/Hold and automatic phone
ring-back, adjust this time value.

Preferred Vocoder

The DAG serial supports up to 4 different Vocoder types including PCMU,

PCMA, G.723.1, G.729AB.The user can configure Vocoders in a preference
list that will be included with the same preference order in SDP message.
The first Vocoder is entered by choosing the appropriate option in Choice
0. The last Vocoder is entered by choosing the appropriate option in
Choice 3.

DTMF

Flexible DTMF transmission method, user interface of In-audio, RFC2833 or

SIP Info

INBAND

Send DTMF as in-audio (inband).

SIGNAL

Send DTMF via SIP INFO message.

RFC 2833

Send DTMF via RTP (According to RFC 2833).

STUN

IP address or Domain name of the STUN server

Polarity Reversal

Default is No. If set to Yes, polarity will be reversed upon call establishment
and termination.

Send Flash Event

Default is NO. If set to yes, flash will be sent as DTMF event

Call Features

Default is Yes. (If Yes, call features using star codes will be supported
locally)

Direct IP-IP Calling

Default is No.

Send Anonymous

Default is No .If this parameter is set to Yes, users ID will be sent as

anonymous; essentially block the Caller ID from displaying.

Reject Anonymous Call

No Dial Timeout

Default is No. If set to Yes, incoming calls with anonymous Caller ID will be
rejected with 486 Busy message.
Default is 3 seconds.

Service ConfigDAG1000-XO
In order to adapt to different environments, it is necessary to set some parameters in the service config
page.

Page 24

FIGURE 19:Service Config (DAG1000-XO)

Page 25

Table 9 Service Config (DAG1000-XO)

RTP Start Port

Defines the local RTP-RTCP port pair the DAG1000-8O will listen and
transmit.

Silence Suppression

Endpoints sending audio as an RTP stream are not required to send packets
during silent periods. The capability to stop sending RTP packets during
silent periods is known as "Silence Suppression" or VAD (Voice Activity
Detection).

Call Progress Tone

Configure ring or tone frequencies according to preference. By default tones

are set to USA frequencies. Frequencies should be configured with known
values to avoid uncomfortable high pitch sounds.

Preferred Vocoder

The DAG1000-8O supports up to 4 different Vocoder types including

PCMU, PCMA, G.723.1, G.729AB.The user can configure Vocoders in a
preference list that will be included with the same preference order in SDP
message. The first Vocoder is entered by choosing the appropriate option in
Choice 0. The last Vocoder is entered by choosing the appropriate option
in Choice 3.

FXO Config
FXO Round Robin

Default is Yes.

Default is No. If set Yes, round robin within the 0~7 ports.

FXO Port 1 stage

Calling

Default is No.

Play Hint to FXO

Default is No.

Send Real Caller ID

Default is Yes.

Tone Disconnect

Default is Yes.

Current Disconnect

Default is No.

FXO Silence Timeout

DTMF

Default is 60 second.
Flexible DTMF transmission method, user interface of In-audio, RFC2833 or
SIP Info

INBAND

Send DTMF as in-audio (inband).

SIGNAL

Send DTMF via SIP INFO message.

RFC 2833

Send DTMF via RTP (According to RFC 2833).

STUN

IP address or Domain name of the STUN server

Send Anonymous
Reject Anonymous Call
Use as Dial Key

Default is No .If this parameter is set to Yes, users ID will be sent as

anonymous; essentially block the Caller ID from displaying.
Default is No. If set to Yes, incoming calls with anonymous Caller ID will be
rejected with 486 Busy message.
Default is Yes.

Note: The device will restart automatically when preferred vocoder is changed between G.723.1 and
G.729AB.

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SIP Config
It is used to set the local SIP port, the SIP server address and port. If an outbound proxy is required,
please fill in the IP address and port of the outbound proxy; otherwise, just keep the default settings.
FIGURE 20:SIP Config

Table 11 SIP Config

SIP Server Address

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

SIP Server Port

SIP Server listening port, i.e. SIP Servers registered port.

Outbound Proxy

IP address or Domain name of Outbound Proxy, or Media Gateway, or

Session Border Controller. Used by DAG serial for firewall or NAT
penetration in different network environments. If symmetric NAT is detected,
STUN will not work and ONLY outbound proxy can correct the problem.

Local SIP Port

Is register
Reg interval
Keepalive interval
100rel enable

Defines the local SIP port the DAG serial will listen and transmit. The
default value for is 2060.
Default is Yes, if it set No, the stand will not register the SIP server, and
the RUN LED will quick blinking (resisted status is Yes)
This parameter allows the user to specify the time frequency (in seconds)
the DAG serial refreshes its registration with the specified registrar.
This parameter specifies how often the DAG serial sends a blank UDP
packet to the SIP server in order to keep the hole on the NAT open. Default
is 10 seconds.
Default is No .

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Port Config
This page is used to config the account and password of DAG serial, call transfer, call waiting and
do-not-disturb service. Port 0 maps to FXS0, while Port n maps to FXS n.
FIGURE 21:Port Config

Table 12 Port Config

SIP User ID
Authenticate ID
Authenticate Password

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 subscribers Authenticate ID used for authentication. Can be
identical to or different from SIP User ID.
SIP service subscribers account password.

TX/RX Gain

Handset volume adjustment.

RX is for receiving volume,
TX is for transmission volume.
Default values are -6dB for both parameters. Loudest volume: +6dB Lowest
volume: -6dB.

Offhook Auto-Dial

This parameter allows users to configure a User ID or extension number to

be automatically dialed upon off-hook. Only the user part of a SIP address
needs is entered here. The DAG serial will automatically append the @
and the host portion of the corresponding SIP address.

Unconditional call
forwarding

All incoming calls are transferred appointed phone.

Busy call forwarding

Call will transfer appointed phone when Busy.

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No answer call
forwarding
call waiting
call waiting tone
DND

Call will transfer appointed phone when nobody answer the call.
Default is No. User can use * code to use this feature per call basis.
indicates an incoming calldefault is 2 short beeps.
When enabled all incoming calls are rejected.

Fax Config
FIGURE 22:Fax Config

Table 13 Fax Config

Fax mode
Fax tone detection mode
ECM
Fax rate

T.38 (Auto Detect) FoIP by default, or Pass-Through

Default is Callee. This decides whether Caller or Callee sends out the
re-INVITE for T.38 or Fax Pass Through.
Error Correction Mode
The DAG serial support most 14400bps least 2400bps fax rate. The default
value is 14400bps.

TOOLS
Tools includes 1) Software Update 2) Login Password 3) Remote Manage 4) Factory Reset 5) Restart.
FIGURE 23:Tools

Software Update
Software upgrade can be done via TFTP. The corresponding configuration settings are in the Software
update configuration page.
NOTES:
If the version is incompatible , the load will fail.

Page 29

Please dont reset DAG serial during software updating, it will restart automatically after updating.
Firmware upgrade may take as long as 1 to 20 minutes over Internet, or just 20+ seconds if it is
performed on a LAN.
Alternatively, user can download a free TFTP server and conduct local firmware upgrade.
FIGURE 24:Software Update

Table 14 Update Config

File Name
TFTP server address

Default name is DAGsipappv7.ldf, you need not change it.

TFTP server address is IP address.
e.g. 172.16.119.119

User & Password

Modify login user name and password to avoid unauthorized access to your DAG serial . The default
username is admin while the default password is admin too.
FIGURE 25:Username & Password

Page 30

Remote Manage
In route mode, if you want to access your DAG serial through WAN port, you need to enable remote
Management.
Note
This option will be displayed only in route mode.
This function can be enabled only by into the web interface through the LAN port.
FIGURE 26:Remote Manage

Factory Reset
WARNING! Reset the Factory Default Setting will DELETE all configuration information of the device.
Please BACKUP or PRINT out all the settings before you approach to following steps. We will not take
any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service
provider.
FIGURE 27:Factory Reset

Restart
When modified, many settings can be effective only after restart, which have prompts in the bottom of the
setting page.
Remotely restart the DAG serial by clicking the Restart button at the bottom of the configuration page.
When finished, re-login to the DAG serial after waiting for about 30 seconds.
FIGURE 28:Restart

SAVING THE CONFIGURATION CHANGES

Click the Save button in the Configuration page to save the changes to the DAG serial configuration.

Page 31

The following screen confirms that the changes are saved. If the changes need restart, reboot or power
cycle the DAG serial to make the changes take effect.
FIGURE 29:Rave Config

Page 32