Basic Phone Functions; Place Or Answer A Call - ATCOM AT800 Series User Manual

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AT800 AT800 AT800 AT800 Series Series User Series Series User User User Manual
3. Configuring the SIP parameters:
�
Click admin on the right corner of the webpage to login administrator interface
� Click Account tab
� Under SIP parameters, make the necessary configuration changes to the fields shown in the
following table and click submit:
Enable: yes/no
User ID:
Password:
SIP Server:
SIP Port:
Transport Type:
After that, click submit button to save the configuration. The phone will be restart automatically to make
the changes effect.
W W W W arning: arning: arning: If the parameter remember arning: remember remember remember account
automatically, user don't have to make a manual login after a restart; if it is set to no no no no, user will have to
enter the User User ID ID ID ID and password User User password password password manually every time restating the phone.
Basic Phone Functions
Basic Basic Basic Phone Phone Phone Functions
Place Place or or or or Answer Answer Answer a a a a call
Place Place Answer
�
Press headset button to place or answer a call
� If the auto answer parameter is set to yes, the incoming call will be answered automatically.
� Enter the Redial list menu, use navigation button to scroll down to highlight an history number, and
press Dial to complete the call.
� Use the programmable key to place a call to the pre-configured number.
For AT800/AT800P, user can place a call as followings:
1. Press headset button, enter number and # (send key) to dial
2. Press headset button, enter a speed dial number, the pre-configured number will be dialed
3. To hang up the active call, press headset button again.
Note: If setting * as the send key, use * instead of #, If send key is set to none, wait for about 10 seconds,
the number will be dialed automatically.
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Manual Manual Manual
Enable/disable the SIP registration
The username, provided by ISP.
The password, provided by ISP.
The registrar and proxy server (we don't distinguish
between the registrar server and the proxy server right
now).
The default SIP port is: 5060, user can modify it to fit the SIP
server's port.
Optional transport type, can be set as UDP/TCP/TLS
account account account is set to yes
Functions Functions
call
call call
yes yes yes, the account information will be saved
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