Yealink SIP-GW3CM User manual

1.Key Feature………………………………………………… 4
1.1 Network Feature......................................4
1.2 Voice Feature......................................4
1.3 Feature Description............................4
1.4 Management Feature............................。5
1.5 Physical Feature.................................5
1.6 Packeage content.................................5
2.Quick install....................................6
2.1 Connecting to the TA......................... 6
2.2 Configuration by webpage.....................7
2.3 Network Configuration(WAN)................7
2.4 Register account(Sip Setting)...........9
2.5 The LED of a SIP-GW3CM.....................11
3.Basic Functions.................................12
3.1 Make Phone Calls....................................13
3.2 Answer Phone Calls.................................13
3.3 Phong Setting.........................................14
3.4 IVR Interface for The TA.......................21
3.5 Phone Book..............................................28
3.6 Network Items.........................................28
3.7 SIP Settings...........................................32
3.8 NAT Trans..............................................36
3.9 Save Change...........................................37
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1.Key Feature
New feature
●Make/Receive both VoIP calls and regular PSTN
calls.
●Auto-detection and auto-switch to VoIP or PSTN
calls.
●Phone rings for all incoming VoIP and PSTN calls.
Support Caller ID/Name display or block.
●Call hold,call waiting,call forwarding and call transfer.
●Support three service domain and auto-provision.
1.1 Network Feature
●Supports SIP 2.0 (RFC3261)PPPoE, STUN, DHCP.
●Support in-band and out-of-band DTMF transfer.
●Support automated NAT traversal without manual manipulation of
firewall/NAT.
●Support firmware upgrade via TFTP, FTP and HTTP.
1.2 Voice Feature
Support VAD,CNG,AEC, PLC,AJB and Volumn adjustment.
Voice Codec: G.711,G.723.1,G.726,G.729A,G.729B.
1.3 Feature Description
●Make/Receive both VoIP calls and regular PSTN calls.
●Auto-detection and auto-switch to VoIP or PSTN calls.
●Phone rings for all incoming VoIP and PSTN calls.
●VoIP or PSTN calls selected by phone key.
●Support Caller ID/Name display or block.
●Support call hold, call waiting, call forwarding, and call transfer.
●Dial plans, dial tone, busy tone, ring back tone, alert tone
can be set flexible.
1.4 Management Feature
●Configured ATA via web browser or remote.
●Firmware can be upgraded through HTTP,FTP or TFTP
1.5 Physical Feature
● Two RJ45 ports: Dual 10M/100M auto-sensing, with
router built-in, one for internet, one for PC.
● Two RJ11 Port: One FXS telephone Port and One lifeline
for PSTN line.
● LED Indication: There are three LED indicators in the TA
to show the Status, and Off-Hook indication.
● Power adaptor:Input:AC 100~240V,Output:DC12V/0.5A.
● Operating Temperature: 0 ~40℃℃.
1.6 Packeage content
●One SIP ATA main body
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●One Ethernet cable
●One user Manual
●One Universal Power Adaptor
●One PSTN cable
2.Quick install
2.1 Connecting to the TA
Please install the TA as the connection chart below:
2.2 Configuring by webpage
2.2.1 Login the webpage
Login via LAN port
a) Connect the PC to LAN port of TA.Default IP address of
LAN port is 192.168.123.1, the port number “:8000”.
b) Open web browser and input http://192.168.123.1:8000
c) Enter the account and password(default account is root and
password is test).
2.3 Network Configuration(WAN)
2.3.1 In Network you can check the Network status, configure
the WAN Settings, LAN Settings.
2.3.2 Network Status: You can check the current Network setting
in this page.
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2.3.3 WAN Settings: The WAN port is for you to connect to the
ADSL Router, Broadband Router. Also you can use
PPPoE to get the WAN IP address from your ISP.
a) The IP Phone’s default setting is NAT mode. If you don’t
need to use the NAT Mode, you can chang to Bridge
Mode.
b) The WAN port default is DHCP Client mode, You can
change the setting to Fixed IP Mode, or PPPoE Mode.
c) If you change the WAN port’s setting to Fix IP Mode, then
you have to make sure the IP address. Net Mask, Gateway,
and DNS setting is suitable in your current network
environment.
d) If you change the WAN port’s setting to PPPoE Mode, you
have to input a correct username/password to get the IP
address from your Internet Service Provider.
2.4 Register account(Sip Setting)
2.4.1You may get account information from your service
provider.
2.4.2 In Service Domain Function you need to input the account
and the related informations in this page, please refer to
your ISP provider. You can register three SIP account in
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the TA. You can dial the VoIP phone to your friends via
first enable SIP account and receive the phone from these
three SIP accounts. For the second phone you can use the
same way to register.
a) Click Active to enable the Service Domain
b) Display Name:input the name you want to display.
c) User Name: input the User Name get from your ISP.
d) Register Name: input the
Register Name get from your ISP.
e) Register Password: input the Register Password from your
ISP.
f) Domain Server: input the Domain Server get from your ISP.
g) Proxy Server: input the Proxy Server get from your ISP.
h) Outbound Proxy: input the Outbound Proxy get from your
ISP. If your ISP does not provide the information, then
you can skip this item.
i) Register Period: input the Register Period get from your
ISP. This is count in minute.
k) If you have more than one SIP account, you can following
the steps to register to the other ISP.
l) When you finished the setting, click the Submit button.
m) If there is nothing need to change, please click the Save
Change Item in the left side, then click the Save button.
The change you made will save into the system and the
system will Reboot automatically.
2.5 The LED of a SIP-GW3CM.
STATUS’s LED PSTN’s LED VOIP’s LED
Unregistered:0.5s Flash
Registered:Lighted
Register Fail:0.5s Flash
Reboot: Lighted
Offhook: Lighted
Reboot: Lighted
Offhook: Lighted
Reboot: Lighted
Reboot: Lighted
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3.Basic Functions
Note1:
1) When SIP-GW3CM is out of power, the RJ11 line jack
will act as a pass through jack. The user will be able to use the
same analog phone for PSTN calls.
2) When SIP-GW3CM doesn’t register service,Pick up the
anologe phone and press the number of “0*”, the RJ 11 line jack
will act as a pass through jack. The user will be able to use the
same analog phone for PSTN calls.
3) When SIP-GW3CM registered service, the SIP line jack
will act as a pass through jack. The user will be able to use the
same analog phone for VoIP calls.
Note2:When you finished the setting, please click the
Submit button. If there is nothing need to change, please
click the Save Change Item in the left side, then click the
Save button. The change you made will save into the system
and the system will Reboot automatically.
3.1 Make Phone Calls
3.1.1 Make VoIP Calls
Note: You can make a VoIP call only when LED “STATUS” is
lighted.
Pick up the analog phone, LED “SIP”light.
There are currently two methods to call out:
a) Dial the numbers directly and wait for 4 (general) seconds.
b) Dial the numbers directly, and press #
3.1.2 Make PSTN Calls
Note: You can make a PSTN call only when PSTN line is
connected.
Pick up the analog phone, if LED “SIP”light, Press “0*”
to switch to PSTN line.
LED “PSTN” light, and get a PSTN line dial tone, dial the
numbers directly.
3.2 Answer Phone Calls
3.2.1 Answer VoIP Calls
When somebody calls in from VoIP, LED “SIP” flashed,
the analog phone shows the caller ID.
Pick up the analog phone, LED “SIP”light, you are in
conversation.
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3.2.2 Answer PSTN Calls
When somebody call in from PSTN, LED “PSTN” flashed,
the analog phone show the caller ID.
Pick up the analog phone, LED “PSTN” light, you are in
conversation.
3.3 Phone Setting
3.3.1 In Phone Setting: contains Call Forward, SNTP Settings,
Volume Settings, Block Setting, Caller ID, Dial Plan
Setting, Flash Time Setting, Call Waiting Setting, and T.38
(FAX) Setting functions.
3.3.2 Call Forward function: you can setup the phone number
you want to forward in this page. There are three type of
Forward mode. You can choose All Forward, Busy
Forward, and No Answer Forward by click the icon.
a) All Forward: All incoming call will forward to the number
you choosed. You can input the name and the phone
number in URL field. If you select this function, then all
the incoming call will direct forward to the speed dial
number you choose.
b) Busy Forward: If you are on the phone, the new incoming
call will forward to the number you choosed. You can
input the name and the phone number in URL field.
c) No Answer Forward: : If you can not answer the phone,
the incoming call will forward to the number you choosed.
You can input the name and the phone number in URL
field.。
3.3.3 SNTP Setting function: you can setup the primary and
second SNTP Server IP Address, to get the date/time
information.
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3.3.4 Volume Setting function: you can setup the Handset
Volume and the Handset Gain.
a) Handset Volume is to set the volume for you can hear from
the handset.
b) Handset Gain is to set the volume send out to the other
side’s handset.
3.3.5 Block Setting function: you can setup the Block Setting
to keep the phone slience. You can choose Always Block
or Block a period.
a) Always Block: All incoming call will be blocked until
disable this feature.
b) Block Period: Set a time period and the phone will be
blocked during the time period.
c) When you finished the setting, please click the Submit
button.
3.3.6 Caller ID function: you can set the device to show Caller
ID in your PSTN Phone or IP Phone.
a) There are four selection of Caller ID. You need to base on
your environment to set the Caller ID function for FSK or
DTMF. When you change the setting, please also double
check the PTT seting in Others. You need to choose the
correct country code then the Caller ID will be effect.
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3.3.7 Dial Plan Setting function: This function is when you
input the phone number by the keypad but you don’t need
to press “#”. After time out the system will dial directly.
a) Symbol explan:
x or X 0.1.2.3.4.5.6.7.8.9
+ or
b) Replace rule:If replace prefix code is ON and prefix
number is matched with rule then 005 will replace prefix.
c) Auto Dial Time : Stop dialing after seconds then send dial
number out.
d) Dial Plan: When match with pattern then send dial number
out but if fisrt digit is ‘0’ then dial plan will be ignored.
Example:
*xx If matched with one of *00,*01….*99
then will send number out
#xx If matched with one of #00,#01….#99
then will send number out
10x If matched with one of 100,101….109
then will send number out
11x If matched with one of 110,111….119
then will send number out
Xxxxxxxx If dial with 8 digits then send number
out
e) Auto Prefix : Number for add before dial number.
f) Prefix Unset Plan : When first digit or dial numeb match
with pattern then ignore auto prefix.
0 Iignore auto prefix if first digit is ‘0’
1 Ignore auto prefix if first digit is ‘1’
xxxxx dial numbers are 4 digits ignore auto
prefix
xxxxxx dial numbers are 5 digits ignore auto
prefix
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