Grandstream Networks HandyTone-496 - page 28

HandyTone-496 User Manual Grandstream Networks, Inc.
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Authenticate ID SIP service subscriber’s Authenticate ID. Can be identical to or different from
SIP User ID.
Authenticate
Password SIP service subscriber’s account password
Name SIP service subscriber’s name which will be used for Caller ID display
Use DNS SRV: Default is No. If set to Yes the client will use DNS SRV for server lookup
User ID is Phone
Number 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
SIP Registration This parameter controls whether the HandyTone ATA needs to send
REGISTER messages to the proxy server. The default setting is “Yes”.
Unregister on
Reboot Default is No. If set to yes, the SIP user’s registration information will be
cleared on reboot.
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the
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).
Local SIP port 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.
Local RTP port 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.
Use Random Port 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.
DTMF Payload
Type This parameter sets the payload type for DTMF using RFC2833
Send DTMF 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.
Send Flash Event Default is NO. If set to yes, flash will be sent as DTMF event.