Grandstream Networks GXW40XX Local SIP port, Local RTP Port, Use random port, Refer to Use Target

Local SIP port	Defines the local SIP port the GXW40XX will listen and transmit. The default value for
	Profile 1 is 5060 and 6060 for Profile 2.
Local RTP Port	Defines the local RTP-RTCP port pair the GXW40XX 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,
	port_value+3 for its RTCP and so on. The default value for Profile 1 is 5004 and 6004 for
	Profile 2.
Use random port	Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
	This is usually necessary when multiple GXW40XX/HT50X 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	Default is No. In which case if conference originator hangs up the conference will be
Conference	terminated. When option YES is chosen, originator will transfer other parties to each
Hang up	other so that B and C can choose either to continue the conversation or hang up.
Enable Ring-Transfer	Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can
	transfer the call upon receiving ring back tone.
Disable Bellcore Style	Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you
3-Way Conference	need to dial *23 + second callee number.
Remove OBP from	Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Route Header:
Support SIP Instance	Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
ID	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
message	to RFC rules. If message does not pass validation process, call will be rejected.
Check SIP user ID for	Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will
incoming INVITE	be rejected.
Allow Incoming SIP	Default is No. If incoming SIP message does not match with SIP Server, it will be
Messages from SIP	rejected.
Proxy Only
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.
Preferred DTMF	The GXW40xx supports up to 3 different DTMF methods including in-audio, via RTP
method (in listed	(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.
order)
Disable DTMF	Default is No. If set to yes, use above DTMF order without negotiation
Negotiation
Send Hook Flash	Default is No. If set to yes, flash will be sent as a DTMF event.
Event
Enable Call Features	Default is Yes. (If Yes, call features using star codes will be supported locally)
Proxy Require	SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Grandstream Networks, Inc.		GXW40XX User Manual	Page 30 of 38
		Firmware 1.0.4.2	Last Updated: 06/2011