User manual
SIP Settings
8.61. You can setup the Service Domain, Port Settngs, Codec Settings, RTP Setting, RPort Setting
and Other Settings for SIP Proxy Server registrations in this page.
Service Domain Settings:
8.62. You may register up to three SIP accounts in the VS211. You can call your friends via firstly
enabled SIP account and receive the phone calls from all the three SIP accounts.
8.63. Click “Active” ON to enable the Service Domain, then enter the following items:
8.64. Display Name: enter the name you want to display.
8.65. User Name: enter the User Name given by your ITSP.
8.66. Register Name: enter the Register Name given by your ITSP.
8.67. Register Password: enter the Register Password given by your ITSP.
8.68. Domain Server: enter the Domain Server given by your ITSP.
8.69. Proxy Server: enter the Proxy Server given by your ITSP.
8.70. Outbound Proxy: enter the Outbound Proxy of ITSP. If not provided, you may skip this.
8.71. Register Period: enter the Register Period in minute given by your ITSP.
8.72. When it shows “Registered” in the Register Status, it indicates a successful registration to
the ITSP, and the “PHONE” LED will start flashing. The VS211 is then ready for VoIP call.
8.73. If you have more than one SIP account, please follow the steps to register to other ITSPs.
8.74. After you finished the setting, please click the “Submit” button.
DTMF Settings:
8.75. You can setup the options for DTMF function in this page. The options include RFC2833
(Outband DTMF), Inband DTMF, and Send DTMF SIP info. The default is set at Inband
DTMF. If you are making two-stage callings for extension to PSTN, you might need to select
Outband DTMF option.
Port Settings:
8.76. You can setup the SIP and RTP port number in this page. Each ITSP provider might have
different SIP/RTP port setting, please refer to the ITSP to setup the port number correctly.
When you finished the setting, please click the “Submit” button. The defaults for SIP port
and RTP port are 5060 and 60000, respectively.
STUN Settings:
8.77. The STUN function must be enabled to work properly behind NAT when registered in SIP
server. You may enter the STUN server IP address and the STUN port number as shown in
the following example.
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