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SIP NAT Traversal Method
NAT Traversal Method: STUN Client / Symmetric RTP
Dialing Plan to SIP protocol
The “Dialing plan” needs setting when the user uses the method of Peer-to-Peer SIP VoIP call or
registering SIP Proxy Server Mode. The SIP Dialing Plan has two kinds of directions: Outgoing (call out)
and Incoming (call in).
Outgoing Dial Plan:
Peer-to-Peer Call Mode: Effective
Registering to SIP Proxy Server Mode: Effective
Incoming Dial Plan
Peer-to-Peer Call Mode: Effective
Registering to SIP Proxy Server Mode:
The leading number would register to SIP Proxy Server
PSTN Route Dial Plan
Peer-to-Peer Call Mode: The same as the Incoming Dial
Plan
Registering to SIP Proxy Server Mode: The leading
number would NOT register to SIP Proxy Server
In the “Outgoing Dial Plan Configurations” settings: Maximum Entries: 50
“Outbound number” is the leading digits of the call out dialing number.
“Length of Number” has two text fields need filled: “Min Length” and “Max Length” is the min/max
allowed length you can dial.
“Delete Length” is the number of digits that will be stripped from beginning of the dialed number.
“Add Digit Number” is the digits that will be added to the beginning of the dialed number.
“Destination IP Address / Domain Name” is the IP address / Domain Name of the destination
Gateway that owns this phone number.