New Rock Technologies NRP1012 User Manual page 51

NRP1012 Series IP Phone
Name
Keep Alive Type
Keep Alive
Interval
User Agent
DTMF Type
DTMF SIP INFO
Mode
Ring Type
Enable Rport
Enable PRACK
Enable Long
Contact
Convert URI
Dial Without
Registered
Ban Anonymous
Call
Enable DNS SRV
Enable Missed
Call Log
Server Type
RFC Protocol
Edition
Local Port
Anonymous call
Edition
Keep
Authentication
Ans. With A
Single Codec
Auto TCP
Enable Strict
Proxy
Enable GRUU
New Rock Technologies, Inc.
Description
Sets the keep alive type. If the value is set to Option, the phone will send an option SIP
message at a set interval, and the server will send "200OK" in response. If the value is
set to UDP, the phone will send a UDP message at a set interval to the server.
Sets the interval at which the phone sends a message to check whether the server
operates normally.
Enables a user agent.
Sets the DTMF sending mode. The value can be:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Different VoIP Service providers may use different modes.
Sets the type to 10/11 or * / #.
Sets the ring tone for of each SIP line.
Enables/Disables the system to support RFC3581. Via rport is a special way to realize
SIP NAT.
Enables or disables the SIP PRACK function, which is specialized for the
(RTB). The default configuration is recommended.
Enables the Contact field to contain more parameters. This function is used together
with SEM services.
Converts # to %23 before sending URI messages.
Allows proxy-based call without registration.
Disables the anonymous call function.
Supports DNS looking up with sip.udp mode.
Enable the missed call log by it, the phone will save the missed call log into the call
history record and display the missed calls on the idle screen, or won't save the missed
call log into the call history record and display the missed calls on the idle screen.
Sets the signaling encryption mode or the type of a special server.
Sets a SIP version. If the phone needs to communicate with the gateway enabling with
SIP 1.0 (such as CISCO5300 ), you need to set the value to RFC2543. The default
value is RFC3261.
Sets a port for each SIP line.
Enables the phone to make anonymous calls safely. Both RFC3323and RFC3325 are
supported.
Enables the phone to send a registration message containing an authentication field to
the server. In this case, the phone does not need to send a separate authentication
message to the server, and the server is required to respond only to the registration
message, which simply the operation process
Enables the phone to respond to the SIP message with just one codec supported.
Enables the phone to automatically use TCP for message transport in the scenario
where the message size exceeds 1300 bytes.
Enables the phone to be compatible with special SIP servers. In composing the
message in response to the server, the phone uses the source IP address of the server,
not the IP address in the via field of the message received.
Enables the phone to support GRUU.
User Manual
Ringback tone
5-21