Grandstream Networks HandyTone-496 User Manual page 28

HandyTone-496 User Manual
Authenticate ID
Authenticate
Password
Name
Use DNS SRV:
User ID is Phone
Number
SIP Registration
Unregister on
Reboot
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the
Local SIP port
Local RTP port
Use Random Port
DTMF Payload
Type
Send DTMF
Send Flash Event
SIP service subscriber's Authenticate ID. Can be identical to or different from
SIP User ID.
SIP service subscriber's account password
SIP service subscriber's name which will be used for Caller ID display
Default is No. If set to Yes the client will use DNS SRV for server lookup
If the HandyTone ATA has an assigned PSTN telephone number, this field
should be set to "Yes". Otherwise, set it to "No". If "Yes" is set, a
"user=phone" parameter will be attached to the "From" header in SIP request
This parameter controls whether the HandyTone ATA needs to send
REGISTER messages to the proxy server. The default setting is "Yes".
Default is No. If set to yes, the SIP user's registration information will be
cleared on reboot.
HandyTone ATA refreshes its registration with the specified registrar. The
default interval is 60 minutes (or 1 hour). The maximum interval is 65535
minutes (about 45 days).
This parameter defines the local SIP port the HandyTone ATA will listen and
transmit. The default value for FXS port 1 is 5060. The default value for FXS
port 2 is 5062.
This parameter defines the local RTP-RTCP port pair the HandyTone ATA will
listen and transmit. It is the base RTP port for channel 0. When configured,
channel 0 will use this port _value for RTP and the port_value+1 for its RTCP;
channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The
default value for FXS port 1 is 5004. The default value for FXS port 2 is 5008.
This parameter, when set to Yes, will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple HandyTone ATAs
are behind the same NAT.
This parameter sets the payload type for DTMF using RFC2833
This parameter specifies the mechanism to transmit DTMF digit. There are 3
modes supported: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP
INFO.
Default is NO. If set to yes, flash will be sent as DTMF event.
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