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S06
User manual
1. Welcome
Congratulations on becoming an owner of S06 telephone adapter! You made an excellent
choice and we hope you will enjoy all its capabilities.
S06 offers a rich set of functionality and superb sound quality at ultra affordable price. It
is fully compatible with SIP industry standard and can interoperate with many other SIP
compliant devices and software on the market.
2. Product brief
2.1. Key Features
• Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, HTTPS, ICMP, ARP, DHCP,
NTP/SNTP, TFTP protocols.
• Support STUN and symmetric RTP for NAT.
• Interoperable with various 3
and gateway products.
• Advanced 32bit embed MCU + DSP technology to ensure superior audio quality.
• Advanced adaptive jitter buffer control, packet delay and loss concealment
technology.
• Support popular vocoders including G.723, G.729A, G.711 (a-law and u-law),
G.726(40K/32K/24K/16K).
• Support standard voice feature such as Caller ID Display, Call Waiting, Hold,
Transfer, in-band and out-of-band DTMF, three way conference.
• Support Silence Suppression, VAD(Voice Activity Detection), CNG(Comfort
Noise Generation), Line Echo Cancellation(G.168), and AGC(Automatic Gain
Control)
2.2. Hardware Overview
IPP 200analog phone adapter is a small CPE. It is easy to install. See picture 2.1.
rd
party SIP end user device, Proxy/Registrar Server,
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IPP 200has following interfaces for networking, visual status indication, and power:
1. One phone (FXS) RJ-11 jack interface.
2. Two Ethernet 10baseT RJ-45 jack interface.
3. One unit status LED.
4. One data link and activity LED.
5. One 5V power adapter interface. It is used to connect to the power adapter going
with the unit. If you connect the unit to some other power adapter, we are not
guarantee that the unit works properly.
2.2.1. LED Indicator
Unit status indicator (STATUS): It is blinking when unit is booting. It will be off when
booting is finished. It will be on when you pick up the phone. It will be off when you
hang up the phone. It is blinking when an inbound call from VoIP. It will be on after
phone is picked up. It will be off after phone is hanged up.
Data link and activity indicator (ACT): It will blink when there is data transferring.
3. Installation
Please make sure you have the following items in the product package you received,
1. Analog Telephone Adapter Box
2. 5V power supply
In addition, you need,
1. An analog telephone set or fax machine
2. One RJ-11 telephone wire
3. Two Cat-5 Ethernet cables
4. An Ethernet port connecting to IP networks
Please follow the instruction below for S06 setup:
1. Insert one end of Cat-5 Ethernet cable to one “LAN” port on the back of ATA and
the other end to Ethernet port connecting to IP networks.
2. Insert one end of Cat-5 Ethernet cable to another “LAN” port on the back of ATA
and the other end to Ethernet port connecting to PC.
3. Insert one end of RJ-11 telephone wire to the “Phone” port on the back of ATA, and
the other end to either the phone set or fax machine.
4. Insert 5V power supply cable into the power plug of ATA.
5. Check and confirm all connections are correct.
Now you can power on the 5V power supply.
4. Setup Guide
It is possible that this product has been pre-setup by either a service provider or network
manager. Then user’s own configurable settings can be very limited. However, if end-
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user is allowed to make changes to the system and service settings, the following are the
instructions for your device setup.
End-user needs to get the following information from his/her service provider or network
manager,
• IP address if it is static, subnet and network gateway IP address, or user DHCP.
• SIP proxy server IP address and any registration information.
• User information such as user ID and password.
4.1 Setup with Keypad
Connecting telephone to I-06’s “Phone” port, press “**00”to enter into S06 setup menu.
You can use keypad referencing following menu to do basic network setup. Please use
web browser to do more advance setup.
4.1.1 Keypad Setup Menu
Option Code User Input Notes
===============================================================
Enter configuration menu **00 None After “Setup menu”,
enter option and end
by “#”
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Check DHCP 10 None Announce
DHCP Status
on or off
------------------------------------------------------------------------------------------------------------
Set DHCP 11 1 to enable Press “#”
0 to disable to confirm the
input
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Check IP Address 20 None
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Set static IP Address 21 Enter value using End input
numbers on the by ‘#’
telephone pad.
Use‘*’when enter
a decimal point.
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Check Subnet Mask 30 None
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Set Subnet Mask 31 Enter value using End input
numbers on the by ‘#’
phone pad. Use ‘*’
for decimal point.
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Check Gateway 40 None
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Set Gateway 41 Enter value using End input
number on the ， by ‘#’.
phone pad. Using
‘*’ for decimal
point
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Check primary DNS 50 None
Server setting
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Set Primary DNS Server 51 Enter value using End input
number on the by ‘#’.
phone pad. Using
‘*’ for decimal
point
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Manufacture Reboot 79898 Enter “1” to confirm ATA will reboot and
Enter “*” to cancel all the datas will
return to the original
form
===============================================================
4.2 Setup with Web Browser
Connecting S06’s “LAN” port to a router. The procedure in 4.2.1 shows you how to setup
your ATA and the menu in 4.2.2 for your reference.
4.2.1 Use Web Browser for system setup
1. After you have got your ATA’s IP address, you can use web browser for system
setup within the same subnet.
2. Enter http://<ATA IP address>/, where <ATA IP Address> is the said IP address of
your ATA.
3. Web browser will show the webpage interface of the system setup menu. Choose
“Admin Login” to enter administrator setup,
4. We use Free-World-Dialup(FWD) IP phone service as an example to setup network
and service.
If your ATA is in a LAN and your LAN connect to Internet through a firewall (with
NAT), your ATA’s IP address will be private. Your system settings should include the
follows,
1. Choose outbound proxy server for NAT traversal
Under User Account Setup
• Set “User ID” to the number provided by your service provider. If you use FWD,
here you can put your FWD number e.g. 180000. As for how to get free FWD
number, please check details of http://www.fwd.pulver.com
• Set “Password” to the password provided by your service provider.
.
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Under Proxy Server Setup
• Set “Proxy” to the SIP proxy server IP address or domain name. In FWD case, it
is “fwd.pulver.com”.
• Set “Use Outbound Proxy” to “yes”.
• Set “Outbound Proxy” to be the outbound proxy server IP address or domain
name. In FWD case, it is http://fwdnat.pulver.com:5082.
At bottom of the pager:
• Click “Save Changes” to apply the changes.
2. Choose STUN server for NAT traversal
Under User Account Setup:
• Set“User ID”to the number provided by your service provider. If you use FWD,
here you can put your FWD number e.g. 180000.
• Set “Password” to the password provided by your service provider.
Under Proxy Server Setup:
• Set “Proxy” to the SIP proxy server IP address or domain name. In FWD case, it
is “fwd.pulver.com”.
• Set “Use Outbound Proxy” to “no”.
Under NAT Support:
• Set“NAT Mapping Enable”to “yes”.
• Set“NAT Keep Alive Enable”to “yes”.
• Set “Substitute VIA address” to “yes”.
• Set “Stun Enable” to “yes”.
• Set “STUN server” to be one of the public STUN sever IP address or domain
name. For an example, http://stun01.sipphone.com
.
At bottom of the pager:
• Click “Save Changes” button to apply all the changes.
If your ATA is connected to the Internet directly, your ATA’s IP address will most likely
be public IP address. Your setting should be,
Under User Account Setup:
• Set“User ID”to the number provided by your service provider. If you use FWD,
here you can put your FWD number e.g. 180000.
• Set “Password” to the password provided by your service provider.
Under Proxy Server Setup:
• Set “Proxy” to the SIP proxy server IP address or domain name. In FWD case, it
is “fwd.pulver.com”.
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At bottom of the pager:
• Click “Save Changes” button to apply all the changes.
4.2.2 Setup Menu
Network Information
Under this tab, the current network information of ATA is presented.
Product Information
Under this tab, the current product information of ATA is presented.
System Status
Under this tab, The states and statistics information, which are collected by ATA during
normal operation, are presented. Line maintains up to 2 calls: Call 1 and 2.
Hook State State of the hook switch
Registration State Registration state of the line
State State of the call
Tone Tone playing for this call
RTP Packets Sent Total number of RTP packets sent
RTP Bytes Sent Total number of RTP bytes sent
RTP Packets Recv Total number of RTP packets received
RTP Bytes Recv Total number of RTP bytes received
SIP Messages Sent Total number of SIP messages sent
SIP Bytes Sent Total number of SIP bytes sent
SIP Messages Recv Total number of SIP messages received
SIP Bytes Recv Total number of SIP bytes received
Password Setup
Parameter Name Function Default
Admen Password The password for administrator
User Password The password for User
Note: the above password is only for getting in “Setup Menu”
Network Setup
Parameter Name Function Default
DHCP Enable or Disable DHCP yes
Static IP Static IP address of ATA, which will take 0.0.0.0
effect if DHCP is disabled
HostName Host Name of ATA
Domain The network domain of ATA
NetMask The NetMask used by ATA when DHCP is 255.255.255.0
disabled
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Gateway The default gateway used by ATA when 0.0.0.0
DHCP is disable
Primary DNS DNS server used by ATA in addition to
DHCP supplied DNS servers if DHCP is
enable; when DHCP is disable, this will be
the primary DNS server
Secondary DNS DNS server used by ATA in addition to
DHCP supplied DNS servers if DHCP is
enable; when DHCP is disable, this will be
the secondary DNS server
SIP and RTP
Parameter Name Function Default
===============================================================
SIP port SIP message listening port and transmission 5060
port
Ext SIP Port External port to substitute for the actual
SIP port of the unite in all outgoing SIP
Messages. If “0” is specified, no SIP port
substitution is performed.
RTP Port Min Minimum port number for RTP 16580
transmission and reception
RTP Port Max Maximum port number for RTP 16680
transmission and reception
RTP Packet Size Packet size in sec. Value must be multiple 0.020
of 0.01s(0.01 to 0.16)
NAT Support
Parameter Name Function Default
===============================================================
NAT Mapping Enable Enable the use of externally mapped of IP yes
address and SIP/RTP ports in SIP message.
NAT Keep Alive Enable If it is sent to “yes”, “NAT Keep Alive Msg” yes
is sent periodically.
Handle_VIA_reveived If set to “yes”, the ATA will process the no
“received” parameter in the VIA header
inserted by the server in a response to any
one of its request. Else the parameter is
ignored
Handle_VIA_rport If set to “yes”, the ATA will process the no
“rport” parameter in the VIA header
inserted by the server in a response to
any one of its request. Else the parameter
is ignored.
Insert VIA reveived Insert received parameter in VIA header no
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in SIP responses if received from IP and
VIA sent-by IP differ
Insert VIA rport Insert rport parameter in VIA header in no
SIP responses if received-from port and
VIA sent-by port differ
Substitute VIA addr Use nat-mapped IP: port value in VIA no
header
Send Resp To Src Port Send response to the request source port no
instead of the VIA sent-by port
STUN Enable Enable the use of STUN to discover NAT no
mapping
STUN Sever STUN server to contact for NAT mapping
discovery
Ext IP External IP address to substitute for the actual
IP address of the unit in all outgoing SIP
Messages. If “0.0.0.0” is specified, no IP
Address substitution is performed.
Ext RTP Port Min External port mapping of “RTP Port Min”.
If value is non-zero, the RTP port number
in all outgoing SIP messages is substituted
by the corresponding port value in the
external RTP port range
Proxy Server Setup
Parameter Name Function Default
===============================================================
Proxy SIP Proxy Server for all outbound requests
Use Outbound Proxy Enable the use of “Outbound Proxy”
Outbound Proxy SIP Outbound Proxy Sever
Register Enable periodic registration with the yes
“Proxy”
Make Call Without Reg Allow making outbound calls without no
successful registration by the unit. If it set
to “no”, dial tone will not play unless
registration is successful
Register Expires Expires value in second in a “REGISTER” 3600
request.
Ans Call Without Reg Allow answering inbound calls without no
successful registration by the unit
User Account Setup
Parameter Name Function Default
Display Name Subscriber’s display name to appear in
caller-id
User ID Subscriber’s user-id
Password Subscriber’s password
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Auth ID Subscriber’s authentication ID
Use Auth ID If it is set to “yes”, the pair “Auth ID> and no
“Password” are used for SIP authentication.
Else the pair “User ID” and “Password”
are used
Audio Setup
Parameter Name Function Default
Preferred Codec Preferred codec for all calls. However, the G729a
Actual codec used in a call still depends on
The outcome of the codec negotiation
Protocol.
DTMF Tx Method Method to transmit DTMF signals to the auto
Far end
Use Pref Codec Only Only use the preferred codec for all calls. no
Silence Supp Enable Enable silence suppression, if it set to no
“yes”, silent audio frames are not
transmitted
Echo Canc Enable If it set to “yes”, the use of echo canceller yes
is enabled.
Echo Canc Adap Enable If it set to “yes”, echo canceller to adapter yes
is enabled
4.3. Call features
Name Code Description
----------------------------------------------------------------------------------------------------------Call return *21 call the last caller code
-----------------------------------------------------------------------------------------------------------Call forward *31 forward all calls to the target specified after
the activation code.
Example:
A with number 1000, B with number 2000,
pick up A, key in *312000, then hang up.
when number 1000 is called, B rings instead
of A. This will stay until someone picks up
A, then key in *32.
-----------------------------------------------------------------------------------------------------------Cancel call forward *32 cancel call forward code
-----------------------------------------------------------------------------------------------------------Call transfer None example: A call B, B transfer A’s call to C.
1)A dial B’s number.
2)B answer, then click hang up hook. After
head a dial tone, key in C’s number.
3)After C answers the phone, B hang up,
A’s call will transfer to C.
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-----------------------------------------------------------------------------------------------------------Three way conference None example: A call B and C for conference.
1)A dial B’s number.
2)After B answered, A(or B)click hang up
hook.
3)After A(or B)heard a dial tone, A(or B)
dial C’s number.
4)After C answered, A(or B)click hang up
hook again, Three way conference is on.
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5. Product Use Environment
S06 is recommended to work as in-door equipment.
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