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Grandstream Networks, Inc. HT-502 User Manual Page 24 of 32
Firmware 1.0.0.77 Last Updated: 1/2008
DNS Mode
One from the 3 modes are available for “DNS Mode” configuration:
-A Record (for resolving IP Address of target according to domain name)
-SRV (DNS SRV resource records indicates how to find services for various protocols)
-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)
One mode can be chosen for the client to look up server.
The default value is “A Record”
User ID is Phone
Number
If the HT502 has an assigned PSTN telephone number, this field should be set to
“Yes”. Otherwise, set to “No”. If set to “Yes”, a “user=phone” parameter will be
appended to the “From” header in SIP request.
SIP Registration Controls whether the HT502 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.
Outgoing Call w/o
Registration
Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if
allowed by ITSP) but is unable to receive incoming calls.
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the HT502
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 Defines the local SIP port the HT502 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 Defines the local RTP-RTCP port pair the HT502 will listen and transmit. It is the base
RTP port for channel 0. When configured,
channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1
uses 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 5012.
Use Random Port This parameter forces the random generation of both the local SIP and RTP ports when
set to Yes. This is usually necessary when multiple HT502 are behind the same NAT.
Refer to Use Target
Contact
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s Contact header information.
Transfer on Conference
Hang up
Default is No. In which case if the conference originator hangs up the conference call
will be terminated. When option YES is chosen, originator will transfer other parties
to each other so that B and C can choose either to continue the conversation
or hang up.
Validate incoming
message
Default is No. If set to yes all incoming SIP messages will be strictly validated
according to RFC rules. If message will not pass validation process, call will be
rejected.
SIP T1 Timeout
T1 is an estimate of the round-trip time between the client and server transactions.
If the network latency is high, select larger value for more reliable usage.
SIP T2 Interval
Maximum retransmission interval for non-INVITE requests and INVITE responses.
DTMF Payload Type
Sets the payload type for DTMF using RFC2833.
DTMF in-audio
Send DTMF as inband (in-audio).
DTMF via RFC2833
Send DTMF via RTP (According to RFC 2833).
DTMF via SIP INFO
Send DTMF via SIP INFO message.
Send Flash Event
Default is No. If set to yes, flash will be sent as DTMF event.
Enable Call Features Default is Yes. (If Yes, call features using star codes will be supported locally)
Offhook Auto-Dial This parameter allows users to configure a User ID or extension number that is
automatically dialed when off-hook. Only the user part of a SIP address needs is
entered here. The HT502 will automatically append the “@” and the host portion of the