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
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.