Alling, Sip-to-sip calling – LevelOne VOI-7100 User Manual

Page 68

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5.2. SIP-to-SIP Calling

Applications:

The SIP-to-SIP calling works when both calling and answering parties are

registered to SIP server with given registered phone numbers. The ADSL

connections can be as in either Diagrams A or B. Both parties are registered

to SIP server under NAT router. For Diagram A without NAT router, you may

select NAT mode to enable the embedded NAT router. For Diagram B with

external NAT router, you may select Bridge mode to disable the embedded

NAT.

Configurations:

1.

Select either “NAT“ or “Bridge” in accord with your network in “WAN

settings” page,

2.

Select “DHCP Client” to automatically get an IP address from NAT router.

3.

Remember to click the “Submit” button,

4.

Select Active “ON” in the “SIP settings / Service Domain” page,

5.

Enter the Register Name, Register Password, Proxy Server, and

Outbound Proxy,

6.

Select “ON” in the “STUN setting”, if Outbound Proxy is NOT available.

7.

Upon successful SIP registration, the

REG

LED indicator will be

ON

and

the LCD will show registered <phone number>.

Callings:

8.

Pick up the phone, and you should hear a dial tone for VoIP mode.

9.

Press 1688# or 1688 to call the party with the registered SIP phone

number 1688. Note that # key will dial out the number immediately.

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