Grandstream Networks 286

HandyTone -286 User Manual Grandstream Networks, Inc.

NAT Traversal This parameter defines whether the phone NAT traversal mechanism will be

activated or not. If activated (by choosing “Yes”) and a STUN server is also

specified, then the phone will behave according to the STUN client

specification. Under this mode, the embedded STUN client inside the phone

will attempt to detect if and what type of firewall/NAT it is behind

y sending

appropriate request to the specified STUN server.

If this field is set to “Yes” with no specified STUN server, then the phone will

only periodically (every 20 seconds by default) send a blank UDP packet (with

no payload data) to the SIP server to keep the mapped port open on the NAT.

keep-alive interval The HandyTone-286 sends a UDP package to the SIP server periodically in

order to keep the port open on the router. This parameter defines the interval

time that HT286 send the UDP package. The default setting is 20 second.

Use NAT IP NAT IP address is used in SIP/SDP message. Default is blank.

Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

SUBSCRIBE for

MWI Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting

Indication will be sent periodically

Offhook

Auto-Dial This parameter allows the user to configure a User ID or extension number to

e automatically dialed upon offhook. Please note that only the user part of a

SIP address needs to be entered here. The phone will automatically append the

“@” and the host portion of the corresponding SIP address.

Enable Call Feature Default is YES. If set to “Yes”, call features are supported locall y, such as call-

waiting, transfer, 3-way conference, etc.

Use Bell-style 3-wa

conference Conference mode, default option is No. If set to yes, the feature code for

coference *23 would be disabled.

Disable Call

Waiting Default is No.

Send DTMF This parameter specifies the mechanism to transmit DTMF digit. There are 3

modes supported: in audio which means DTMF is combined in audio signal

(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP

INFO.

DTMF Pa

load

Type This parameter sets the payload type for DTMF using RFC2833

Send Flash Event Default is NO. If set to yes, flash will be sen t as DTMF event.

FXS Impedance Selects the impedance of the analog telephone connected to the Phone port.