Grandstream Networks HT503 Authenticate Password, Outgoing Call Without, SIP registration failure

	according to the STUN client specification. Using this mode, the embedded STUN
	client will detect if and what type of firewall/NAT is being used.
	If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the
	HT503 will use its mapped public IP address and port in all of its SIP and SDP
	messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the
	HT503 will periodically (every 20 seconds or so) send a blank UDP packet (with no
	payload data) to the SIP server to keep the “hole” on the NAT open.
SIP User ID	User account information, provided by VoIP service provider (ITSP). Usually in the form
	of digit similar to phone number or actually a phone number.
Authenticate ID	The SIP service subscriber’s ID used for authentication. Can be identical to or different
	from SIP User ID.
Authenticate Password	SIP service subscriber’s account password.
Name	SIP service subscriber’s name for Caller ID display.
DNS mode	One from the 3 modes 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 HT503 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	Controls whether the HT503 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 Without	Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if
Registration	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 HT503
	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).
SIP registration failure	This parameters allows the user to specify the time frame (in seconds) the HT503 will
retry wait time	wait before sending another SIP registration INVITE in case the first INVITE fails.
Local SIP Port	Defines the local SIP port the HT503 will listen and transmit. The default value for FXS
	port is 5062.
Local RTP Port	This parameter defines the local RTP-RTCP port pair used by the HandyTone ATA. It
	is the base RTP port for FXO channel.
	When configured, the FXO port will use this port _value for RTP and the port_value+1
	for its RTCP.
	The default value for FXO port 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 HT503 units are behind the same
	NAT.
Refer to Use Target	Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
Contact	transferred target’s contact header information.
Remove OBP from Route	Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Header
Support SIP instance ID	Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
	Instance ID as defined in IETF SIP Outbound draft.
Validate incoming	Default is No. If set to yes all incoming SIP messages will be strictly validated
message	according to RFC rules. If message will not pass validation process, call will be
	rejected.
Check SIP User ID for	Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the
Grandstream Networks, Inc.	HT503 User Manual	Page 30 of 38
	Firmware 1.0.4.2	Last Updated: 06/2011