ArtDio IPS 1000 Series Manuale utente
Voice Internet Phone Gateway
IPS 1000 Series
User Manual
Version:3.1 Update:2004/5/7
ARTDio Company Inc.
IPS 1000 Series VoIP Gateway
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Contents
1. Safety Instructions ............................................................................................................................4
2. Preface..............................................................................................................................................4
2.1. What is SIP........................................................................................................................4
2.1.1. Components of SIP ...................................................................................................5
3. Package Contents.............................................................................................................................7
4. Panel Descriptions............................................................................................................................7
4.1. Front Panel........................................................................................................................7
4.2. Rear Panel ........................................................................................................................8
5. LED Indicators ................................................................................................................................10
6. Connectors......................................................................................................................................11
7. IDC Connectors (Only for 1008/1016) ............................................................................................11
8. Information required before Installation...........................................................................................12
8.1. IPAddress.......................................................................................................................12
8.2. SIP Information................................................................................................................13
8.3. Prepare a password for Web Management.....................................................................13
9. Installation and Configuration .........................................................................................................14
9.1. Confirming the Region ID................................................................................................14
9.1.1. Phone Setting..........................................................................................................14
9.1.2. System console settings (Only 3704/3708/3716)....................................................15
9.2. IPAddress Settings.........................................................................................................15
9.2.1. Static IP Mode .........................................................................................................16
9.2.2. DHCP Mode ............................................................................................................17
9.2.3. PPPoE Mode...........................................................................................................17
10. SIP Configuration....................................................................................................................23
10.1. Channels and SIP entity..................................................................................................24
10.2. SIP Proxy and Register Parameters ...............................................................................25
10.3. SIP Entity.........................................................................................................................25
10.4. SIP Outbound Authentication ..........................................................................................26
10.5. Configure STUN..............................................................................................................27
10.6. Check SIP entity Status...................................................................................................28
10.7. Phone Book.....................................................................................................................28
10.8. Make SIP Calls................................................................................................................29
10.9. Contact Address..............................................................................................................30
11. Other SIP Parameters.............................................................................................................31
11.1. Dialing Plan.....................................................................................................................31
11.2. Call Forward....................................................................................................................32
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11.3. Inbound Authentication....................................................................................................32
11.4. FAX..................................................................................................................................33
11.4.1. The devices at two sides are all IPS 1000 series gateway......................................33
11.4.2. The devices at two sides are IPS 1000 and the other brands.................................33
12. WEB MANAGEMENT INTERFACE........................................................................................34
12.1. BASIC / GENERAL .........................................................................................................35
12.2. IP SETTING ....................................................................................................................37
12.3. ADVANCED / GENERAL ................................................................................................39
12.4. SIP COMMON.................................................................................................................41
12.5. SIP OUTBOUND AUTHENTICATION.............................................................................44
12.6. SIP INBOUND ANTHENTICATION.................................................................................45
12.7. Dialing Plan.....................................................................................................................46
12.8. STUN...............................................................................................................................48
12.9. CHANNEL.......................................................................................................................49
12.10. PHONE BOOK................................................................................................................51
13. Use Private IP (Behind NAT)...................................................................................................52
14. Appendix.................................................................................................................................53
14.1. Appendix A: Phone-Set Command..................................................................................53
14.2. Appendix B: Console Command.....................................................................................55
14.3. Specifications ..................................................................................................................56
14.4. Mapping table of characters used in PPPoE...................................................................57
14.5. Region ID ........................................................................................................................58
15. Contact Information.................................................................................................................59
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1. Safety Instructions
1. Do not attempt to service the product yourself. Any servicing of this product should be referred to
qualified service personal.
2. To avoid electric shock, do not put your finger, pin, wire, or any other metal objects into vents and
gaps.
3. To avoid accidental fire or electric shock, do not twist power cord or place it under heavy objects.
4. The product should be connected to a power supply of the type described in the operating
instructions or as marked on the product.
5. To avoid hazard to children, dispose of the product’s plastic packaging carefully.
6. The phone line should always be connected to the LINE connector. It should not be connected to the
PHONE connector as it may cause damage to the product.
7. Please read all the instructions before using this product.
2. Preface
The IPS 1000 unit is a personal SIP VoIP gateway developed using the latest in VoIP technology. It is
also very simple to install and easy to operate.
2.1. What is SIP
Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for
multimedia conferencing over IP. SIP is anASCII-based, application-layer control protocol (defined in
RFC 2543& RFC 3621) that can be used to establish, maintain, and terminate calls between two or more
end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session
management within a packet telephony network. Signaling allows call information to be carried across
network boundaries. Session management provides the ability to control the attributes of an end-to-end
call.
SIP provides the following capabilities:
Determine the location of the target end point—Supports address resolution, name mapping, and call
redirection.
Determine the media capabilities of the target end point—By using Session Description Protocol (SDP),
WARNING
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SIP determines the highest level of common services between the end points. Conferences are
established using only the media capabilities that can be supported by all end points.
Determine the availability of the target end point—If a call cannot be completed because the target end
point is unavailable, SIP determines whether the called party is already on the phone or did not answer
in the allotted number of rings. It then returns a message indicating why the target end point is
unavailable.
Establish a session between the originating and target end point—If the call can be completed, SIP
establishes a session between the end points. SIP also supports mid-call changes, such as the addition
of another end point to the conference or the changing of a media characteristic or Codec.
Handle the transfer and termination of calls—SIP supports the transfer of calls from one end point to
another. During a call transfer, SIP simply establishes a session between the transferee and a new end
point (specified by the transferring party) and terminates the session between the transferee and the
transferring party. At the end of a call, SIP terminates the sessions between all parties.
2.1.1. Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs).A user agent can
function in one of the following roles:
User agent client (UAC)—A client application that initiates the SIP request.
User agent server (UAS)—A server application that contacts the user when a SIP request is received
and that returns a response on behalf of the user.
Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UAthat
initiated the request.
From an architecture standpoint, the physical components of a SIP network can be grouped into two
categories: clients and servers.
IPS 1000 Series VoIP Gateway
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Architecture
SIP Clients
SIP clients include the following:
Phones—Can act as either a UAS or UAC. Soft phones (PCs that have phone capabilities installed) and
Cisco SIP IP phones can initiate SIP requests and respond to requests.
Gateways—Provide call control. Gateways provide much functionality. The most common one is a
translation function between SIP conferencing endpoints and other terminal types. This function includes
translation between transmission formats and between communications procedures. In addition, the
gateway also translates between audio and video Codec and performs call setup and clearing on both
the LAN side and the switched-circuit network side.
SIP Servers
SIP servers include the following:
Proxy server—The proxy server is an intermediate device that receives SIP requests from a client and
then forwards the requests on the client's behalf. Basically, proxy servers receive SIP messages and
forward them to the next SIP server in the network. Proxy servers can provide functions such as
authentication, authorization, network access control, routing, reliable request retransmission, and
security.
Redirect server—Provides the client with information about the next hop or hops that a message should
take, then the client contacts the next hop server or UAS directly.
Registrar server—Processes requests from UACs for registration of their current location. Registrar
servers are often co-located with a redirect or proxy server.
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3. Package Contents
The IPS 1000 Gateway x 1
Power Core X 1
Accessories for fixing support X 1 (For 1008/1016)
System CD-ROM X 1
5 IDC Connector X 4 (For 1008/1016)
Rubber footer
RJ-45 Ethernet Cable X 1
RJ-11 Telephone Cable X 1
4. Panel Descriptions
4.1. Front Panel
REGISTERED STUN
IPS 1016 Front Panel (16 ports)
REGISTERED STUN
IPS 1008 Front Panel (8 ports)
IPS 1004 Front Panel (4 ports)
IPS 1000 Series VoIP Gateway
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IPS 1002 Front Panel (2 ports)
4.2. Rear Panel
There is a button on the rear panel of gateway for special maintenance. Please don’t touch this button
under normal operation.
IPS 1016 Rear Panel (16 ports)
IPS 1008 Rear Panel (8 ports)
IPS 1202 Rear Panel (4 ports)
IPS 1004 Rear Panel (4 ports)
IPS 1400 Rear Panel (4 ports)
IPS 1000 Series VoIP Gateway
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IPS 1103 Rear Panel (4 ports)
IPS 1101 Rear Panel (2 ports)
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5. LED Indicators
LED Label Description
On Link up
Off Link down
LNK/ACT
Flash Sending/Receiving
data packets
On (LNK is on) 100Mbps
10/100
Ethernet
100Mbps
Off (LNK is on) 10Mbps
On Off hook
Off On hook
FXS
Flash Ringing out
On Line is active
Off Line is inactive
LOOP/RING
FXO
Flash Ringing in
Alarm The red light “On” indicates that system has
some problem; please contact your vender.
Power “On” indicates that the power supply is
working normally.
CPU/ACT “On” indicates that the CPU is working
normally.
Registered “On” indicates that all SIP entities are
registered successful.
“Off” indicates that all SIP entities are
registered fail.
“Flash” indicates that one of these SIP
entities is registered fail.
Device
STUN “On” indicates communicate with STUN
Server once.
“Off” indicates never communicate with
STUN Server.
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