Grandstream Networks GXW40XX User Manual download pdf (Page 30)

Grandstream Networks, Inc. GXW40XX User Manual Page 30 of 38

Firmware 1.0.4.2 Last Updated: 06/2011

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 wi

ll use port_value+2 for RTP,

port_value+3 for its RTCP and so on. The default value for Profile 1 is 5004 and 6

004 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

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 conference originator hangs up the conference 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.

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

3-Way Conference

Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you

need to dial *23 + second callee number.

Remove OBP from

Route Header:

Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.

Support SIP Instance

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

message

Default is No. If set to yes all incoming SIP messages will be strictly validated according

to RFC rules. If message does not pass validation process, call will be rejected.

Check SIP user ID for

incoming INVITE

Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will

be rejected.

Allow Incoming SIP

Messages from SIP

Proxy Only

Default is No. If incoming SIP message does not match with SIP Server, it 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.

Preferred DTMF

method (in listed

order)

The GXW40xx supports up to 3 different DTMF methods including in-audio, via RTP

(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.

Disable DTMF

Negotiation

Default is No. If set to yes, use above DTMF order without negotiation

Send Hook Flash

Event

Default is No. If set to yes, flash will be sent as a DTMF 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.

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Grandstream Networks GXW40XX User Manual Page 30

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