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SIP Operation Manual
Enable Out-of-Band DTMF: Send DTMF keys (0~9, *, #), follow the RFC2833 rules or via
SIP Info.
Enable Hook Flash Event: According to RFC2833 or SIP info the gateway will deliver
Hook Flash signal to the remote party.
Payload Type
：
Payload type of RFC2833.
SIP Info: This is an alternative for DTMF event over IP. When enabled, DTMF is relayed
over SIP signaling path using SIP NOTIFY messages.
Uses Second CPT after SIP registered: This function is usually applied when the user
select VoIP as the primary path for outgoing calls. The gateway will generate a different
set of tones to inform the user that VoIP is in service. When VoIP call is failed, the user
will hear the first set CPT instead of the second one. (for CPT settings, refer to
CPT/Cadence Settings
)
Enable Non-SIP Inbox Call: Untick on the check box to disable Non-SIP inbox call if all
calls need to go through ITSP. Non
Line Settings
Listening Volume: Adjusts the hearing volume.
Speaking Volume: Adjusts the speaking volume.
Tone Volume: Adds a new option to make tone volume adjustable. This setting will be
applied to all tones generated by the gateway including Dial Tone, Busy Tone, and so on.
Flash Time:
FXS: Adjust the detecting period of flash signal from the phone set connected to the FXS
port. For example, if pressing the HOLD key will disconnect a call, increase the “Flash
Time” should fix this issue.
FXO: Set the time frame that FXO generates a FLASH signal.
Enable Polarity Reversal:
FXS: As the remote site answer this call or hook on the FXS port will reverse the
polarity.
FXO: This option forces the gateway to detect the reversal of polarity on FXO port as
the primary signal to drop a call. Some telephone switches or PBX reverse the line
polarity to inform the remote site to drop an ongoing call. Please consult with the
telephone service provider for availability of this feature.
PSTN Answer Detection: This is only used with ITSP. When someone makes a call to this
FXO port from Internet, it could identify if the remote party of PSTN port answer this call.
After it dials to PSTN, it will send “183” to another UAC/UAS. After the remote party of
PSTN port answers this call, it will send “200ok” to another UAC/UAS.
