Grandstream Networks DP710 Advance Configuration DNS Mode, Tel URI, SIP Registration, Contact

	If the NAT Traversal field is set to “Yes” with no specified STUN server, the DP715 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.
	Advance Configuration
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”
Tel URI	The default setting is “Disabled”. If the phone has an assigned PSTN
	Number, this field should be set to “User=Phone” then a
	“User=Phone” parameter will be attached to the “From header” in the SIP request to
	indicate the E.164 number. If server supports TEL URI format, then this option needs to
	be selected.
SIP Registration	Controls whether the DP715 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 Internet Telephone Service Provider) but is unable to receive incoming calls.
Register Expiration	This parameter allows the user to specify the time frequency (in minutes) the DP715
	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	Retry registration if the process failed. Default is 20 seconds.
Retry Wait Time
Local SIP port	Defines the local SIP port the DP715 will listen and transmit. Default is 5060 for profile 1 &
	6060 for profile 2.
Local RTP port	Defines the local RTP port the DP715 will listen and transmit. It is the base RTP port for
	channel 0. When configured,
	channel 0 uses this port _value for RTP. Default is 5004 for profile 1 & 6004 for profile 2.
Use Random Port	Default is No. This parameter forces the random generation of both the local SIP and RTP
	ports when set to Yes. This is usually necessary when multiple DP715 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.
Transfer on Conference	Default is No. In which case if the conference originator hangs up the conference will be
Hang up	terminated. When option YES is chosen, originator will transfer other parties to each other
	so that B and C can choose to either continue the conversation or hang up.
Disable Bellcore Style 3-	Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you need
Way Conference	to dial *23 + second callee number.
Remove OBP from	Default is No. When option YES is chosen, the Out Bound Proxy will be removed from
Route Header	Route 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 SIP	Default is No. If set to yes all incoming SIP messages will be strictly validated according to
message	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
incoming INVITE	call will be rejected. If this option is enabled, the device will not be able to make direct IP
	calls.

Firmware version 1.0.0.8 DP715/DP710 User Manual Page 42 of 52