Grandstream Networks HandyTone-386 User Manual page 29

HandyTone-386 User Manual
SIP User ID
Authenticate ID
Authentication
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
User account information, provided by VoIP service provider (ITSP), usually
has the form of digit similar to phone number or actually a phone number
ID used for authentication, usually same as SIP user ID, but could be different
and decided by ITSP.
Password for ATA to register to (SIP) servers of ITSP. Purposely blank out
once saved for security. Maximum length is 25.
User name, not user ID, for information only.
Default is No. If set to Yes the client will use DNS SRV to lookup for the
server
If the HandyTone. If set to yes, a "user=phone" parameter will be attached to
the "From" header in SIP request
This parameter controls whether the HT386 needs to send REGISTER
messages to the proxy server. The default setting is "Yes".
Default is No. If set to Yes, the device will first send registration request to
indicate SIP registra to remove previous bindings.
HT386 will refresh 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 HT386 will listen and transmit.
The default value is for FXS1 is 5060, FXS2 is 5062
This parameter defines the local RTP-RTCP port pair the HT386 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
FXS1 is 5004, FXS2 is 5008.
Default No. If set to Yes, the device will pick randomly-generated SIP and RTP
ports. This is usually necessary when multiple SIP devices are behind the same
NAT. For Direct IP to IP call, this should be set to No.
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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Grandstream Networks, Inc.